Re: [asterisk-users] PJSIP Losing knowledge of external_media_address

[George Joseph]

On Fri, Aug 18, 2023 at 10:09 AM Mark Murawski 
wrote:

> I've seen this happen three times in the wild now.  I've been trying to
> isolate the source of the issue, but so far it seems like there's not
> enough debug output to know why this occurs.
>
> Long story short:
> - Start Asterisk
> - PJSIP Handles receiving INVITE from ITSP via WAN (Asterisk is behind
> NAT).  SIP is handled correctly, Asterisk responds OK with RTP media
> address of external_media_address
> - After 30 minutes to an hour or sometimes months later after startup,
> upon receiving INVITE from ITSP via WAN, Asterisk responds OK with
> INTERNAL LAN IP instead of external_media_address
> - I've observed this occur after 30 minutes from startup with no
> configuration changes that were made or any pjsip reloads done during
> this period
>

Any network change activity going on at the time?  VPN coming up or down?
Network interface flapping? DNS availability issue (could still be an issue
even though you're using ip addresses for local_net)?

One way to get more information would be to modify
that ast_sip_transport_is_local block in
res_pjsip_session:session_outgoing_nat_hook to print more debug info if the
ast_sip_transport_is_local check fails and the destination is your itsp.
If you want really detailed info, you could compile with DONT_OPTIMIZE and
put an abort() in there then run ast_coredumper when asterisk crashes to
get the backtrace.  That's a service disruption of course.



>
> pjsip
> -
> [global]
> endpoint_identifier_order = username,ip,anonymous
>
> [system]
> type=system
> threadpool_initial_size=30
> threadpool_auto_increment=5
> threadpool_idle_timeout=0
> threadpool_max_size=100
>
> [transport-udp]
> type   = transport
> symmetric_transport= yes
> protocol   = udp
> bind   = 0.0.0.0:5060
> external_media_address = 152.X.Y.Z
> external_signaling_address = 152.X.Y.Z
> external_signaling_port= 5060
> allow_reload   = no
> tos= cs3
> cos= 3
> local_net  = 127.0.0.1/24
> local_net  = 192.168.50.0/24
> local_net  = 192.168.1.0/24
> local_net  = 10.3.2.0/24
> local_net  = 10.20.1.0/24
> local_net  = 10.10.41.0/24
> local_net  = 10.5.1.0/24
>
> pjsip_wizard
> -
>
> [isoft-sr-in-1]
> type = wizard
> transport = transport-udp
> remote_hosts = 192.81.237.20
> aor/max_contacts = 1
> aor/remove_existing = yes
> aor/qualify_frequency = 60
> aor/qualify_timeout = 2000
> endpoint/ice_support = no
> endpoint/disallow = g723,slin,ilbc,lpc10,g729,speex,g726aal2,g722
> endpoint/allow = ulaw,alaw,adpcm,gsm
> endpoint/direct_media = no
> endpoint/force_rport = yes
> endpoint/rewrite_contact = yes
> endpoint/rtp_keepalive = 30
> endpoint/rtp_symmetric = yes
> endpoint/rtp_timeout = 60
> endpoint/rtp_timeout_hold = 60
> endpoint/send_pai = yes
> endpoint/send_rpid = yes
> endpoint/trust_id_inbound = yes
> endpoint/trust_id_outbound = yes
> endpoint/trust_connected_line = no
> endpoint/send_connected_line = no
> endpoint/context = trunkhandler_pbx-sip-t1
>
>
> Attached sip sessions and debug log... the only thing I found
> interesting was finding a lack of a log item
> We SHOULD be seeing:
> DEBUG[X] res_pjsip_session.c: (null session): Setting external media
> address to 152.X.Y.Z
> This message is clearly lacking from the debug session where the
> incorrect media address is sent.  But there's not enough detail in the
> debugs to see why this decision was not made to use external_media_address
>
> By default we use nat settings for all our endpoints, but obviously it's
> not required here for an ITSP that has trustworthy media ports in the
> SDP.  Maybe a bandaid is turning off rewrite_contact for this endpoint?
> Going to try that as soon as possible.
>
> Why is external_media_address not being used all of a sudden?  Has
> anyone else seen this... is this a bug?--
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> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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[asterisk-users] Asterisk Infrastructure Move to GitHub

[George Joseph]

In order to reduce the amount of system maintenance and administration that
needs to be done by the Asterisk team at Sangoma, we've decided to move
capabilities such as issue tracking, code management/review and
documentation/wiki to hosted solutions. Last year, we compared GitHub and
GitLab and while the evaluation of documentation/wiki alternatives is still
ongoing, we've decided that GitHub offers the best alternative for issues
and code management/review.

The [Asterisk Community Forums](https://community.asterisk.org/) are
already hosted by Discourse and are not moving but you can now also use
your GitHub account to log into the forums. Make sure the email you use for
the forums is also listed under your account Settings/Emails in GitHub.

So...

Over the weekend of April 29-30 2023, GitHub will become the official and
sole platform for issue tracking and code management.  IT IS NOT POSSIBLE
FOR US TO MIGRATE EITHER ISSUES OR CODE REVIEWS TO THE NEW PLATFORMS but
the existing Jira issue tracker and Gerrit code management systems will be
placed in read-only mode for historical reference.  At some point in the
future, the historical issues in Jira will be exported to a searchable
format and the system deactivated.  The Gerrit system will be deactivated
at the same time but since the most important historical data is already
captured as part of the commit history, there's no need to create a
searchable archive.

More detailed information, especially concerning release tarballs,
changelogs, etc are at
https://wiki.asterisk.org/wiki/display/AST/Release+Management

NOTE:  If you're an Asterisk contributor, stay tuned.  There will be more
info about the code management/review process in the next day or so.

-- 
*George Joseph*
*Asterisk Software Developer*
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Re: [asterisk-users] I think there may be a bug in 18.14.0 ${GEOLOC_PROFILE(profile_precedence)}, seems to always return prefer_incoming

[George Joseph]

On Wed, Aug 24, 2022 at 7:51 AM George Joseph  wrote:

> Yeah, that's weird.  I opened an issue for this...
> https://issues.asterisk.org/jira/browse/ASTERISK-30190
>
> OK, It's actually not weird :)
Let's say the configured profile is set to discard_config or
prefer_incoming and there actually is an incoming profile.  In this
situation, by the time you reach the dialplan, we've already discarded the
configured profile in favor of the incoming one so profile_precedence is
going to be what's on the incoming one which will always be
prefer_incoming.  Is that going to be an issue?

BTW, there still is a bug where effective_location will be blank in this
same situation and there are patches up on Gerrit that fix that and a few
other bugs.


> On Tue, Aug 23, 2022 at 2:47 PM Dan Cropp  wrote:
>
>> Running into a problem when retrieving the profile_precedence in the
>> extensions.conf
>>
>>
>>
>> Creating a very basic geolocation.conf to allow passing through
>> geolocation values for outbound.
>>
>>
>>
>> [discard_config]
>>
>> type = profile
>>
>> profile_precedence = discard_config
>>
>>
>>
>> [discard_incoming]
>>
>> type = profile
>>
>> profile_precedence = discard_incoming
>>
>>
>>
>> [prefer_config]
>>
>> type = profile
>>
>> profile_precedence = prefer_config
>>
>>
>>
>> [prefer_incoming]
>>
>> type = profile
>>
>> profile_precedence = prefer_incoming
>>
>>
>>
>>
>>
>> I have tried setting the pjsip.conf geoloc_incoming_call_profile to all
>> four of these profiles for inbound call testing.  The discard_incoming
>> correctly blocks the geo location information.  Other 3 pass the geo
>> location values through
>>
>>
>>
>> [192.168.33.31]
>>
>> type = endpoint
>>
>> context = IS
>>
>> transport = transport1
>>
>> aors = 192.168.33.31
>>
>> accountcode = 20
>>
>> dtmf_mode = inband
>>
>> device_state_busy_at = 1600
>>
>> moh_passthrough = no
>>
>> identify_by = username,ip,header
>>
>> disallow = all
>>
>> allow = ulaw
>>
>> acl = acl1
>>
>> geoloc_incoming_call_profile = prefer_config
>>
>> geoloc_outgoing_call_profile = prefer_config
>>
>>
>>
>> When I have the following line in the extensions.conf, it’s retrieving
>> the GEOLOC_PROFILE(profile_precedence) to the variable, but it’s being set
>> to prefer_incoming even when it should be discard_config or prefer_config.
>>
>>
>>
>> same =>
>> n,Set(MY__GEO_PROFILE_PRECEDENCE=${GEOLOC_PROFILE(profile_precedence)})
>>
>>
>>
>> Dan
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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Re: [asterisk-users] [External] [External] Geo location 18.14.0-rc1 question

[George Joseph]

On Tue, Aug 16, 2022 at 8:58 AM Dan Cropp  wrote:

> Thank you George.
>
>
>
> As you pointed out, my mistake of the double equal sign caused the problem.
>
>
>
> Using the passthrough profile and in the AMI Originate setting the
> Variable: GEOLOC_PROFILE(name) is exactly what we need.
>
> My software will receive the GEO settings from third party software.
>
> If third party passed a field/value that doesn’t match the Asterisk
> defaults, our software will add the GEOLOC_PROFILE(name) to the Originate
> Variable field.
>
> Then I send the Originate packet to Asterisk via AMI.
>
>
>
Good news!



> Thank you for all your work on this!!!
>

Thanks for testing and your suggestions!!!


>
>
> Dan
>
>
>
> *From:* asterisk-users  *On
> Behalf Of *George Joseph
> *Sent:* Tuesday, August 16, 2022 7:49 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject:* Re: [External] [asterisk-users] [External] Geo location
> 18.14.0-rc1 question
>
>
>
>
>
>
>
> On Mon, Aug 15, 2022 at 1:59 PM Dan Cropp  wrote:
>
> Thank you George.
>
>
>
> Good idea on the passthrough profile.
>
>
>
> Is there a way to set the GEOLOC_PROFILE values from the AMI Originate
> command?
>
>
>
> I tried the following, but it doesn’t like the GEOLOC_PROFILE values in
> the variable parameter.  If there is a way to do this, the passthrough
> would solve the problem of Geo Location information settings needing to be
> provided by a third-party application.
>
>
>
> Action: Originate
>
> Channel: PJSIP/1234@192.168.33.31
>
> Exten: createcall
>
> Context: mycontext
>
> Priority: 1
>
> Timeout: 6
>
> CallerID: John Smith <8005551234>
>
> Variable:
> GEOLOC_PROFILE(format)=civicAddress,GEOLOC_PROFILE(method)=Manual,GEOLOC_PROFILE(location_info)
> *==*
> "country=US,A1=Florida,A3=Orlando,HNO=100,RD=Main,STS=Street",CALLERID(num-pres)=allowed_passed_screen
>
> Async: true
>
>
>
>
>
> You've got 2 equals signs when you set location_info :).
>
> I just tried
>
> GEOLOC_PROFILE(format)=civicAddress,GEOLOC_PROFILE(location_info)="country=US,A3=\"New
> York\"",GEOLOC_PROFILE(pidf_element)=device
>
> and it worked.
>
>
>
> I believe this portion believe indicates Asterisk treats the
> GEOLOC_PROFILE(xxx) as GEOLOCPROFILESTATUS variable name.
>
>
>
> GEOLOCPROFILESTATUS is the variable GEOLOC_PROFILE sets to indicate
>
> success or failure.  The value of "0" indicates success.  What was the
> actual result in the channel?
>
>
>
> [08/15 13:41:41.609] DEBUG[42424] manager.c: Examining AMI event:
>
> Event: VarSet^M
>
> Privilege: dialplan,all^M
>
> Channel: PJSIP/192.168.33.31-^M
>
> ChannelState: 0^M
>
> ChannelStateDesc: Down^M
>
> CallerIDNum: ^M
>
> CallerIDName: ^M
>
> ConnectedLineNum: ^M
>
> ConnectedLineName: ^M
>
> Language: en^M
>
> AccountCode: 20^M
>
> Context: IS^M
>
> Exten: s^M
>
> Priority: 1^M
>
> Uniqueid: 1660588901.0^M
>
> Linkedid: 1660588901.0^M
>
> Variable: GEOLOCPROFILESTATUS^M
>
> Value: 0^M
>
> ^M
>
>
>
> Dan
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] [External] Geo location 18.14.0-rc1 question

[George Joseph]

On Mon, Aug 15, 2022 at 3:07 PM Dan Cropp  wrote:

> George,
>
>
>
> Is it possible to set the GEOLOC_PROFILE fields similar to the way
> PJSIP_HEADER(add, …) works?
>

I can try but unlike the HEADER functions, the profile has multiple levels
of name/value pairs so it might take a bit of work and testing.   Let me
finish the other things first then I'll evaluate this.


>
>
> Asking, because we already use the PJSIP_HEADER(add, xxx) from AMI
> Originate to add PJSIP headers to the outbound originate.
>
>
>
>
>
> Action: Originate
>
> ActionID: S62
>
> Channel: PJSIP/1234@192.168.12.34
>
> Exten: createcall
>
> Context: mycontext
>
> Priority: 1
>
> Timeout: 6
>
> CallerID: John Smith <8005551212>
>
> Variable:
> PJSIP_HEADER(add,abc)=123,CALLERID(num-pres)=allowed_passed_screen
>
> Async: true
>
> Codecs: ulaw
>
>
>
> Dan
>
>
>
>
>
>
>
> *From:* Dan Cropp
> *Sent:* Monday, August 15, 2022 2:00 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject:* RE: [External] [asterisk-users] Geo location 18.14.0-rc1
> question
>
>
>
> Thank you George.
>
>
>
> Good idea on the passthrough profile.
>
>
>
> Is there a way to set the GEOLOC_PROFILE values from the AMI Originate
> command?
>
>
>
> I tried the following, but it doesn’t like the GEOLOC_PROFILE values in
> the variable parameter.  If there is a way to do this, the passthrough
> would solve the problem of Geo Location information settings needing to be
> provided by a third-party application.
>
>
>
> Action: Originate
>
> Channel: PJSIP/1234@192.168.33.31
>
> Exten: createcall
>
> Context: mycontext
>
> Priority: 1
>
> Timeout: 6
>
> CallerID: John Smith <8005551234>
>
> Variable:
> GEOLOC_PROFILE(format)=civicAddress,GEOLOC_PROFILE(method)=Manual,GEOLOC_PROFILE(location_info)=="country=US,A1=Florida,A3=Orlando,HNO=100,RD=Main,STS=Street",CALLERID(num-pres)=allowed_passed_screen
>
> Async: true
>
>
>
> I believe this portion believe indicates Asterisk treats the
> GEOLOC_PROFILE(xxx) as GEOLOCPROFILESTATUS variable name.
>
>
>
> [08/15 13:41:41.609] DEBUG[42424] manager.c: Examining AMI event:
>
> Event: VarSet^M
>
> Privilege: dialplan,all^M
>
> Channel: PJSIP/192.168.33.31-^M
>
> ChannelState: 0^M
>
> ChannelStateDesc: Down^M
>
> CallerIDNum: ^M
>
> CallerIDName: ^M
>
> ConnectedLineNum: ^M
>
> ConnectedLineName: ^M
>
> Language: en^M
>
> AccountCode: 20^M
>
> Context: IS^M
>
> Exten: s^M
>
> Priority: 1^M
>
> Uniqueid: 1660588901.0^M
>
> Linkedid: 1660588901.0^M
>
> Variable: GEOLOCPROFILESTATUS^M
>
> Value: 0^M
>
> ^M
>
>
>
> Dan
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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Re: [asterisk-users] [External] Geo location 18.14.0-rc1 question

[George Joseph]

On Mon, Aug 15, 2022 at 1:59 PM Dan Cropp  wrote:

> Thank you George.
>
>
>
> Good idea on the passthrough profile.
>
>
>
> Is there a way to set the GEOLOC_PROFILE values from the AMI Originate
> command?
>
>
>
> I tried the following, but it doesn’t like the GEOLOC_PROFILE values in
> the variable parameter.  If there is a way to do this, the passthrough
> would solve the problem of Geo Location information settings needing to be
> provided by a third-party application.
>
>
>
> Action: Originate
>
> Channel: PJSIP/1234@192.168.33.31
>
> Exten: createcall
>
> Context: mycontext
>
> Priority: 1
>
> Timeout: 6
>
> CallerID: John Smith <8005551234>
>
> Variable:
> GEOLOC_PROFILE(format)=civicAddress,GEOLOC_PROFILE(method)=Manual,GEOLOC_PROFILE(location_info)
> *==*
> "country=US,A1=Florida,A3=Orlando,HNO=100,RD=Main,STS=Street",CALLERID(num-pres)=allowed_passed_screen
>
> Async: true
>
>
>

You've got 2 equals signs when you set location_info :).
I just tried
GEOLOC_PROFILE(format)=civicAddress,GEOLOC_PROFILE(location_info)="country=US,A3=\"New
York\"",GEOLOC_PROFILE(pidf_element)=device
and it worked.

I believe this portion believe indicates Asterisk treats the
> GEOLOC_PROFILE(xxx) as GEOLOCPROFILESTATUS variable name.
>

GEOLOCPROFILESTATUS is the variable GEOLOC_PROFILE sets to indicate
success or failure.  The value of "0" indicates success.  What was the
actual result in the channel?

>
>
> [08/15 13:41:41.609] DEBUG[42424] manager.c: Examining AMI event:
>
> Event: VarSet^M
>
> Privilege: dialplan,all^M
>
> Channel: PJSIP/192.168.33.31-^M
>
> ChannelState: 0^M
>
> ChannelStateDesc: Down^M
>
> CallerIDNum: ^M
>
> CallerIDName: ^M
>
> ConnectedLineNum: ^M
>
> ConnectedLineName: ^M
>
> Language: en^M
>
> AccountCode: 20^M
>
> Context: IS^M
>
> Exten: s^M
>
> Priority: 1^M
>
> Uniqueid: 1660588901.0^M
>
> Linkedid: 1660588901.0^M
>
> Variable: GEOLOCPROFILESTATUS^M
>
> Value: 0^M
>
> ^M
>
>
>
> Dan
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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Re: [asterisk-users] Geo location 18.14.0-rc1 question

[George Joseph]

Oh, for now, you can also use something like this to conditionally set an
element:
${IF(${EXISTS(${MyA3})}?A3=${MyA3})}



On Mon, Aug 15, 2022 at 9:57 AM George Joseph  wrote:

> The email was getting too long for the mailing list so I've truncated it.
> Anyway...
>
> Dan, Have you tried using the GEOLOC_PROFILE dialplan function to create a
> profile on the fly?  I'm not sure how this would fit into your use of
> Originate but...
>
> Create a dummy geoloc profile.
> []
> type=profile
> profile_precedence=discard_config
>
> In pjsip.conf, set
> geoloc_outgoing_call_profile = 
> on the outgoing channel.
>
> Either in the dialplan or via the Originate (I forget how) call
> GEOLOC_PROFILE.  Here's what it'd look like in the dialplan...
>
> [geoloc_callee]
> exten = s,1,NoOp(); CID: ${CALLERID(all)}  CL: ${CONNECTEDLINE(all)})
> same  = n,Set(GEOLOC_PROFILE(format)=civicAddress)
> same  = n,Set(GEOLOC_PROFILE(location_info)=country=US,A1=Colorado)
> same  = n,Set(GEOLOC_PROFILE(pidf_element)=device)
> same  = n,Return()
>
> [default]
> exten  = 911,Dial(PJSIP/upstream-carrier,20,b(geoloc_callee^s^1))
>
> Not sure if this helps.  I am working on the other suggestions though.
>
>
> On Mon, Aug 15, 2022 at 3:46 AM George Joseph  wrote:
>
>>
>>
>> On Sat, Aug 13, 2022 at 3:55 PM Dan Cropp  wrote:
>>
>>> Thank you George.
>>>
>>>
>>>
>>> rc2 did fix the issue.
>>>
>>
>> Whew.
>>
>>
>>>
>>>
>>> I am now able to program the variables in the location_info and pass
>>> values via the AMI Originate variables.
>>>
>>>
>>>
>>> Is there a way to make the location_info optional?
>>>
>>
>> I _believe_ I can allow you to specify "location" parameters directly on
>> a profile which would be mutually exclusive with the location_reference
>> parameter.  Let me look at it.
>>
>>
>>>
>>>
>>> On the same PJSIP Endpoint, we may need to originate calls with
>>> different requirements for the fields passed.
>>>
>>> For example
>>>
>>> Dialing a number for customer A, need to pass country, A1, A3, HNO, RD,
>>> STS, PC, FLR and ROOM.
>>>
>>> Dialing a number for customer B, need to pass country, A1, A2, and A3.
>>>
>>>
>>>
>>> Is it possible to program the Profile/Location to support values for all
>>> the settings, but have Asterisk ignore any settings where the value is
>>> blank?
>>>
>>
>> A good idea.  I can't think of any element that would need to be sent
>> empty but I'm wondering if I'll need to add a config option like
>> "suppress_empty_elements".  Let me investigate.
>>
>>
>>>
>>>
>>> Right now, if I have HNO=${MYGEO_FLR} but the variable MYGEO_FLR is not
>>> set, it still passes the FLR in the sip as 
>>>
>>>
>>>
>>> I believe it’s perfectly fine to send the 
>>>
>>> My fear is some system may interpret a blank FLR differently than the
>>> FLR not being present at all.
>>>
>>>
>>>
>>> Dan
>>>
>>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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[asterisk-users] Geo location 18.14.0-rc1 question

[George Joseph]

The email was getting too long for the mailing list so I've truncated it.
Anyway...

Dan, Have you tried using the GEOLOC_PROFILE dialplan function to create a
profile on the fly?  I'm not sure how this would fit into your use of
Originate but...

Create a dummy geoloc profile.
[]
type=profile
profile_precedence=discard_config

In pjsip.conf, set
geoloc_outgoing_call_profile = 
on the outgoing channel.

Either in the dialplan or via the Originate (I forget how) call
GEOLOC_PROFILE.  Here's what it'd look like in the dialplan...

[geoloc_callee]
exten = s,1,NoOp(); CID: ${CALLERID(all)}  CL: ${CONNECTEDLINE(all)})
same  = n,Set(GEOLOC_PROFILE(format)=civicAddress)
same  = n,Set(GEOLOC_PROFILE(location_info)=country=US,A1=Colorado)
same  = n,Set(GEOLOC_PROFILE(pidf_element)=device)
same  = n,Return()

[default]
exten  = 911,Dial(PJSIP/upstream-carrier,20,b(geoloc_callee^s^1))

Not sure if this helps.  I am working on the other suggestions though.


On Mon, Aug 15, 2022 at 3:46 AM George Joseph  wrote:

>
>
> On Sat, Aug 13, 2022 at 3:55 PM Dan Cropp  wrote:
>
>> Thank you George.
>>
>>
>>
>> rc2 did fix the issue.
>>
>
> Whew.
>
>
>>
>>
>> I am now able to program the variables in the location_info and pass
>> values via the AMI Originate variables.
>>
>>
>>
>> Is there a way to make the location_info optional?
>>
>
> I _believe_ I can allow you to specify "location" parameters directly on a
> profile which would be mutually exclusive with the location_reference
> parameter.  Let me look at it.
>
>
>>
>>
>> On the same PJSIP Endpoint, we may need to originate calls with different
>> requirements for the fields passed.
>>
>> For example
>>
>> Dialing a number for customer A, need to pass country, A1, A3, HNO, RD,
>> STS, PC, FLR and ROOM.
>>
>> Dialing a number for customer B, need to pass country, A1, A2, and A3.
>>
>>
>>
>> Is it possible to program the Profile/Location to support values for all
>> the settings, but have Asterisk ignore any settings where the value is
>> blank?
>>
>
> A good idea.  I can't think of any element that would need to be sent
> empty but I'm wondering if I'll need to add a config option like
> "suppress_empty_elements".  Let me investigate.
>
>
>>
>>
>> Right now, if I have HNO=${MYGEO_FLR} but the variable MYGEO_FLR is not
>> set, it still passes the FLR in the sip as 
>>
>>
>>
>> I believe it’s perfectly fine to send the 
>>
>> My fear is some system may interpret a blank FLR differently than the FLR
>> not being present at all.
>>
>>
>>
>> Dan
>>
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] [External] [External] [External] [External] [External] Geo location 18.14.0-rc1 question

[George Joseph]

On Sat, Aug 13, 2022 at 3:55 PM Dan Cropp  wrote:

> Thank you George.
>
>
>
> rc2 did fix the issue.
>

Whew.


>
>
> I am now able to program the variables in the location_info and pass
> values via the AMI Originate variables.
>
>
>
> Is there a way to make the location_info optional?
>

I _believe_ I can allow you to specify "location" parameters directly on a
profile which would be mutually exclusive with the location_reference
parameter.  Let me look at it.


>
>
> On the same PJSIP Endpoint, we may need to originate calls with different
> requirements for the fields passed.
>
> For example
>
> Dialing a number for customer A, need to pass country, A1, A3, HNO, RD,
> STS, PC, FLR and ROOM.
>
> Dialing a number for customer B, need to pass country, A1, A2, and A3.
>
>
>
> Is it possible to program the Profile/Location to support values for all
> the settings, but have Asterisk ignore any settings where the value is
> blank?
>

A good idea.  I can't think of any element that would need to be sent empty
but I'm wondering if I'll need to add a config option like
"suppress_empty_elements".  Let me investigate.


>
>
> Right now, if I have HNO=${MYGEO_FLR} but the variable MYGEO_FLR is not
> set, it still passes the FLR in the sip as 
>
>
>
> I believe it’s perfectly fine to send the 
>
> My fear is some system may interpret a blank FLR differently than the FLR
> not being present at all.
>
>
>
> Dan
>
>
>
> *From:* asterisk-users  *On
> Behalf Of *George Joseph
> *Sent:* Thursday, August 11, 2022 1:22 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject:* Re: [External] [asterisk-users] [External] [External]
> [External] [External] Geo location 18.14.0-rc1 question
>
>
>
>
>
>
>
> On Thu, Aug 11, 2022 at 8:43 AM Dan Cropp  wrote:
>
> Thank you George.
>
>
>
> I am still running on asterisk 18.14.0-rc1 and have not retrieved the
> patches yet.
>
> Did this version have a bug with the variables?
>
>
>
> It's quite possible.  RC2 was just released so you should try that.
>
>
>
> I’m trying the location_info and variables in the AMI Originate you
> recommended at the end of the previous e-mail.
>
>
>
> In case it’s not coming through correctly via e-mail, the variable names
> are preceeded with a single underscore in the AMI and in the location_info
> values.
>
>
>
>
>
> [IS_loc_5]
>
> type = location
>
> format = civicAddress
>
> location_info = country=${_MY_GEO_COUNTRY}
>
> location_info = A1=${_MY_GEO_NATIONAL_SUBDIVISION}
>
> location_info = A2=${_MY_GEO_NATSUB}
>
> location_info = A3=${_MY_GEO_CITY}
>
> location_info = HNO=${_MY_GEO_HNO}
>
> location_info = RD=${_MY_GEO_RD}
>
> location_info = STS=${_MY_GEO_STS}
>
> location_info = PC=${_MY_GEO_PC}
>
>
>
> [IS_prof_9]
>
> type = profile
>
> location_reference = IS_loc_5
>
> pidf_element = device
>
> profile_action = discard_incoming
>
> usage_rules = retransmission_allowed=yes
>
>
>
>
>
> [192.168.33.31]
>
> type = endpoint
>
> context = IS
>
> transport = transport1
>
> auth = auth14
>
> aors = 192.168.33.31
>
> accountcode = 20
>
> dtmf_mode = inband
>
> device_state_busy_at = 1600
>
> moh_passthrough = no
>
> identify_by = username,ip,header
>
> disallow = all
>
> allow = ulaw
>
> acl = acl1
>
> geoloc_incoming_call_profile = IS_prof_7
>
> geoloc_outgoing_call_profile = IS_prof_9
>
>
>
>
>
>
>
> Using a telnet session, I connect up via AMI and login.  Then I attempt to
> Originate.  The call goes through, but none of the location_info settings
> are being updated
>
>
>
> Action: Originate
>
> Channel: PJSIP/1234@192.168.33.31
>
> Exten: createcall
>
> Context: IS
>
> Priority: 1
>
> Timeout: 6
>
> CallerID: John Smith <8005551234>
>
> Variable:
> _MY_GEO_COUNTRY=US,_MY_GEO_NATSUB=Florida,_MY_GEO_CITY=Orlando,_MY_GEO_HNO=100,_MY_GEO_RD=Main,_MY_GEO_STS=Street,CALLERID(num-pres)=allowed_passed_screen
>
> Async: true
>
>
>
>
>
> [08/10 15:04:12.470] DEBUG[1774] manager.c: Running action 'Originate'
>
> [08/10 15:04:12.470] DEBUG[1907] chan_pjsip.c:  1234@192.168.33.31
> Topology:  <0:audio-0:audio:sendrecv (slin)>
>
> [08/10 15:04:12.470] DEBUG[1614] chan_pjsip.c:  1234@192.168.33.31
>
> [08/10 15:04:12.470] DEBUG[1614] res_pjsip_session.c:  192.168.33.31 1234
> Topology:  <0:audio-0:audio:sen

Re: [asterisk-users] [External] [External] [External] [External] Geo location 18.14.0-rc1 question

[George Joseph]

IP/192.168.33.31-0005 Method: INVITE Status: 100
>
> [08/10 15:04:12.473] DEBUG[1614] res_pjsip_session.c:
>
> [08/10 15:04:12.473] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-0005 TSX State: Proceeding  Inv State: CALLING
>
> [08/10 15:04:12.473] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-0005: Response is 100 Trying
>
> [08/10 15:04:12.473] DEBUG[1614] chan_pjsip.c:
> PJSIP/192.168.33.31-0005: Status: 100
>
> [08/10 15:04:12.473] DEBUG[1614] chan_pjsip.c:
> PJSIP/192.168.33.31-0005: Not queueing anything
>
> [08/10 15:04:12.473] DEBUG[1614] chan_pjsip.c:
> PJSIP/192.168.33.31-0005
>
> [08/10 15:04:12.473] DEBUG[1614] chan_pjsip.c:
> PJSIP/192.168.33.31-0005: Status: 100
>
> [08/10 15:04:12.473] DEBUG[1614] chan_pjsip.c:
> PJSIP/192.168.33.31-0005
>
> [08/10 15:04:12.473] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-0005
>
> [08/10 15:04:12.473] DEBUG[1614] res_pjsip_session.c:  Nothing delayed
>
> [08/10 15:04:12.473] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-0005 TSX State: Proceeding  Inv State: CALLING
>
> [08/10 15:04:12.473] DEBUG[1614] res_pjsip_session.c:  Topology: Pending:
> <0:audio-0:audio:sendrecv (ulaw)>  Active: (null topology)
>
> [08/10 15:04:12.473] DEBUG[1614] res_pjsip_session.c:
>
>
>
> The SIP INVITE is as follows (all the civic address settings are blank,
> despite the AMI VarSet events showing the variables were set on the channel)
>
>
>
> [08/10 15:04:12.472] VERBOSE[1614] res_pjsip_logger.c: <--- Transmitting
> SIP request (2265 bytes) to UDP:192.168.33.31:5060 --->
>
> INVITE sip:1234@192.168.33.31 SIP/2.0^M
>
> Via: SIP/2.0/UDP 192.168.33.33:5060
> ;rport;branch=z9hG4bKPj37ef75eb-0ddc-4802-baa7-0921ff30ff8a^M
>
> From: "John Smith"  >;tag=1720fabc-afd7-46ad-aa08-ff84140a7add^M
>
> To: ^M
>
> Contact: ^M
>
> Call-ID: c0a41f60-08d7-43e3-9901-39860a8001f5^M
>
> CSeq: 21131 INVITE^M
>
> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
> CANCEL, UPDATE, PRACK, MESSAGE, REFER^M
>
> Supported: 100rel, timer, replaces, norefersub, histinfo^M
>
> Session-Expires: 1800^M
>
> Min-SE: 90^M
>
> Geolocation: ^M
>
> Geolocation-Routing: no^M
>
> Max-Forwards: 70^M
>
> User-Agent: Asterisk PBX 18.14.0-rc1^M
>
> Content-Type:
> multipart/mixed;boundary=8542a9ce-7ac4-47f7-8c17-d42771ebd564^M
>
> Content-Length:  1470^M
>
> ^M
>
> ^M
>
> --8542a9ce-7ac4-47f7-8c17-d42771ebd564^M
>
> Content-Type: application/sdp^M
>
> Content-Length:   181^M
>
> ^M
>
> v=0^M
>
> o=- 823099435 823099435 IN IP4 192.168.33.33^M
>
> s=Asterisk^M
>
> c=IN IP4 192.168.33.33^M
>
> t=0 0^M
>
> m=audio 17214 RTP/AVP 0^M
>
> a=rtpmap:0 PCMU/8000^M
>
> a=ptime:20^M
>
> a=maxptime:150^M
>
> a=sendrecv^M
>
> ^M
>
> --8542a9ce-7ac4-47f7-8c17-d42771ebd564^M
>
> Content-ID: ^M
>
> Content-Type: application/pidf+xml^M
>
> Content-Length:  1009^M
>
> ^M
>
> 
>
>  xmlns:ca="urn:ietf:params:xml:ns:pidf:geopriv10:civicAddr"
> xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:fn="
> http://www.w3.org/2005/xpath-functions"; xmlns:gbp="urn:ietf:params:xml:\
>
> ns:pidf:geopriv10:basicPolicy" xmlns:gml="http://www.opengis.net/gml";
> xmlns:gp="urn:ietf:params:xml:ns:pidf:geopriv10" xmlns:gs="
> http://www.opengis.net/pidflo/1.0"; xmlns:date="
> http://exslt.org/dates-and-times"; entity="IS_prof_9">
>
>   
>
> 
>
>   
>
> 
>
>   
>
>   
>
>   
>
>   
>
>   
>
>   
>
>   
>
>   
>
> 
>
>   
>
>   
>
> yes
>
>   
>
> 
>
> 2022-08-10T20:04:12Z
>
>   
>
> 
>
> ^M
>
> --8542a9ce-7ac4-47f7-8c17-d42771ebd564--^M
>
>
>
>
>
> *From:* asterisk-users  *On
> Behalf Of *George Joseph
> *Sent:* Wednesday, August 10, 2022 1:34 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject:* Re: [External] [asterisk-users] [External] [External]
> [External] Geo location 18.14.0-rc1 question
>
>
>
>
>
>
>
> On Wed, Aug 10, 2022 at 11:25 AM Dan Cropp  wrote:
>
> Thank you George.
>
>
>
> Looking forward to working with the changes.  I will retrieve them when
> the next release candidate comes out.
>
>
>
>
>
> A quic

Re: [asterisk-users] [External] [External] [External] Geo location 18.14.0-rc1 question

[George Joseph]

On Wed, Aug 10, 2022 at 11:25 AM Dan Cropp  wrote:

> Thank you George.
>
>
>
> Looking forward to working with the changes.  I will retrieve them when
> the next release candidate comes out.
>
>
>
>
>
> A quick question on using variables to pass custom Geo Location settings
> on via an AMI Originate.
>
>
>
>
>
> If my AMI originate request looks something like this…
>
> Action: Originate
>
> Channel: PJSIP/1234@192.168.x.x
>
> Exten: createcall
>
> Context: mycontext
>
> Priority: 1
>
> Timeout: 6
>
> CallerID: John Smith <8005551234>
>
> Variable:
> _MY_GEO_COUNTRY=US,_MY_GEO_NATSUB=Florida,_MY_GEO_CITY=Orlando,_MY_GEO_HNO=100,_MY_GEO_RD=Main,_MY_GEO_STS=Street
>
> Async: true
>
>
>
> Do I need to program the location_variables in the profile like this?
>
>
>
> [1]
>
> type = profile
>
> pidf_element = device
>
> profile_action = discard_incoming
>
> usage_rules = retransmission_allowed=yes
>
> location_variables = country=${_MY_GEO_COUNTRY}
>
> location_variables = A1=${_MY_GEO_NATIONAL_SUBDIVISION}
>
> location_variables = A2=${_MY_GEO_NATSUB}
>
> location_variables = A3=${_MY_GEO_CITY}
>
> location_variables = HNO=${_MY_GEO_HNO}
>
> location_variables = RD=${_MY_GEO_RD}
>
> location_variables = STS=${_MY_GEO_STS}
>
> location_variables = PC=${_MY_GEO_PC}
>
>
>
> Or would I need to program the location_info_refinements in the profile to
> use those variables?
>

location_info_refinement is what you want.  location_variables defines *new*
variables to use in addition to those on the channel.  You'd use these if
you had variables that for some reason you didn't want on the channel
itself.

However...   The profile you defined above doesn't have a location
reference to refine so you'd need at least a dummy location with a format
of civicAddress.

[mylog]
type = location
format = civicAddress

Then in your profile...
[1]

type = profile

pidf_element = device

profile_action = discard_incoming

usage_rules = retransmission_allowed=yes
location_reference = myloc

location_variables = country=${_MY_GEO_COUNTRY}

location_variables = A1=${_MY_GEO_NATIONAL_SUBDIVISION}
...

You can also do this which might actually be faster...
[mylog]
type = location
format = civicAddress
location_info = country=${_MY_GEO_COUNTRY},
A1=${_MY_GEO_NATIONAL_SUBDIVISION}
location_info = A2=${_MY_GEO_NATSUB}, ...

[1]

type = profile

pidf_element = device

profile_action = discard_incoming

usage_rules = retransmission_allowed=yes
location_reference = myloc

This way you don't need to use location_info_refinement at all.
IIRC this saves having to parse location_info_refinement and bounce it
against
the original location_info which could be empty.







>
>
> Dan
>
>
>
>
>
> *From:* asterisk-users  *On
> Behalf Of *George Joseph
> *Sent:* Wednesday, August 10, 2022 8:58 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject:* Re: [External] [asterisk-users] [External] [External] Geo
> location 18.14.0-rc1 question
>
>
>
> Sorry for the delay but this turned out to be a bit more complex than I
> anticipated.
>
> There are reviews up on Gerrit for the 16 and 18 branches that address the
> issues below as well as clean up the implementation, plug some memory
> leaks, etc.
>
> 16: https://gerrit.asterisk.org/c/asterisk/+/18896
>
> 18: https://gerrit.asterisk.org/c/asterisk/+/18897
>
>
>
> I anticipate these will make it into the next set of release candidates
> which are due to be cut tomorrow.
>
>
>
> Give them a try.
>
>
>
> On Wed, Aug 3, 2022 at 1:51 PM George Joseph  wrote:
>
> Looks like it'll be tomorrow before I can get the patch up.  I ran into
> some strange issues.
>
>
>
> On Tue, Aug 2, 2022 at 1:43 PM Dan Cropp  wrote:
>
> Thank you George
>
>
>
> *From:* asterisk-users  *On
> Behalf Of *George Joseph
> *Sent:* Tuesday, August 2, 2022 2:40 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject:* Re: [External] [asterisk-users] [External] Geo location
> 18.14.0-rc1 question
>
>
>
>
>
>
>
> On Tue, Aug 2, 2022 at 1:35 PM George Joseph  wrote:
>
>
>
>
>
> On Tue, Aug 2, 2022 at 1:13 PM Dan Cropp  wrote:
>
> Is the allow_routing setting on the geolocation Wiki Profile also not
> fully implemented?
>
>
>
> Well, 99% of the code is there.  The 1% is parsing the config option.  Not
> sure how I missed that.
>
> I'll have a patch up first thing in the morning UTC-6.
>
> I'

Re: [asterisk-users] [External] [External] Geo location 18.14.0-rc1 question

[George Joseph]

Sorry for the delay but this turned out to be a bit more complex than I
anticipated.
There are reviews up on Gerrit for the 16 and 18 branches that address the
issues below as well as clean up the implementation, plug some memory
leaks, etc.
16: https://gerrit.asterisk.org/c/asterisk/+/18896
18: https://gerrit.asterisk.org/c/asterisk/+/18897

I anticipate these will make it into the next set of release candidates
which are due to be cut tomorrow.

Give them a try.

On Wed, Aug 3, 2022 at 1:51 PM George Joseph  wrote:

> Looks like it'll be tomorrow before I can get the patch up.  I ran into
> some strange issues.
>
> On Tue, Aug 2, 2022 at 1:43 PM Dan Cropp  wrote:
>
>> Thank you George
>>
>>
>>
>> *From:* asterisk-users  *On
>> Behalf Of *George Joseph
>> *Sent:* Tuesday, August 2, 2022 2:40 PM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
>> asterisk-users@lists.digium.com>
>> *Subject:* Re: [External] [asterisk-users] [External] Geo location
>> 18.14.0-rc1 question
>>
>>
>>
>>
>>
>>
>>
>> On Tue, Aug 2, 2022 at 1:35 PM George Joseph  wrote:
>>
>>
>>
>>
>>
>> On Tue, Aug 2, 2022 at 1:13 PM Dan Cropp  wrote:
>>
>> Is the allow_routing setting on the geolocation Wiki Profile also not
>> fully implemented?
>>
>>
>>
>> Well, 99% of the code is there.  The 1% is parsing the config option.
>> Not sure how I missed that.
>>
>> I'll have a patch up first thing in the morning UTC-6.
>>
>> I'll call it "allow_use_for_routing" in profile.
>>
>>
>>
>> Actually just "allow_routing_use"
>>
>>
>>
>>
>>
>>
>>
>> In the code, I see geolocation_routing used instead of allow_routing.
>>
>>
>>
>> Tried both and Asterisk indicates it cannot find suitable setting so it
>> doesn’t create the profile object.
>>
>>
>>
>> Dan
>>
>>
>>
>> *From:* Dan Cropp
>> *Sent:* Tuesday, August 2, 2022 10:04 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
>> asterisk-users@lists.digium.com>
>> *Subject:* RE: [External] [asterisk-users] Geo location 18.14.0-rc1
>> question
>>
>>
>>
>> Thank you George.
>>
>>
>>
>> *From:* asterisk-users  *On
>> Behalf Of *George Joseph
>> *Sent:* Tuesday, August 2, 2022 9:57 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
>> asterisk-users@lists.digium.com>
>> *Subject:* Re: [External] [asterisk-users] Geo location 18.14.0-rc1
>> question
>>
>>
>>
>>
>>
>>
>>
>> On Tue, Aug 2, 2022 at 8:46 AM Dan Cropp  wrote:
>>
>> I believe I have everything configured correctly, but Asterisk is
>> complaining about my configuration
>>
>>
>>
>> It is complaining about confidence settings.
>>
>>
>>
>> From the Asterisk Geolocation Implementation Wiki, I believe I have this
>> set correctly.
>>
>>
>>
>> Sub-parameters:
>>
>>- value: A percentage indicating the confidence or "unknown".
>>- pdf: "unknown", "normal" or "rectangular"
>>Example: confidence = value=80, pdf=unknown
>>If no confidence parameter is specified, the default is 95%.
>>See RFC7459
>>
>> <https://wiki.asterisk.org/wiki/display/AST/Geolocation+Reference+Information#GeolocationReferenceInformation-rfc7459>
>>  for
>>the exact definition of this parameter.
>>
>>
>>
>>
>>
>> [08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find
>> option suitable for category 'IS_loc_1' named 'confidence' at line 12 of
>>
>> [08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create
>> an object of type 'location' with id 'IS_loc_1' from configuration file
>> 'geolocation.conf'
>>
>>
>>
>> [IS_loc_1]
>>
>> type = location
>>
>> format = civicAddress
>>
>> confidence = value=95, pdf=unknown
>>
>> location_info = country=US,A1=Wisconsin,A3=Madison
>>
>> location_info = HNO=4800,RD=Main,STS=Drive,PC=53704
>>
>>
>>
>> Remove the confidence param for now.I documented it before I
>> implemented it. :)
>>
>>
>>
>>
>>
>>
>>
>> Also seeing problems with location_refinement setting.
>>
>

Re: [asterisk-users] [External] [External] Geo location 18.14.0-rc1 question

[George Joseph]

Looks like it'll be tomorrow before I can get the patch up.  I ran into
some strange issues.

On Tue, Aug 2, 2022 at 1:43 PM Dan Cropp  wrote:

> Thank you George
>
>
>
> *From:* asterisk-users  *On
> Behalf Of *George Joseph
> *Sent:* Tuesday, August 2, 2022 2:40 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject:* Re: [External] [asterisk-users] [External] Geo location
> 18.14.0-rc1 question
>
>
>
>
>
>
>
> On Tue, Aug 2, 2022 at 1:35 PM George Joseph  wrote:
>
>
>
>
>
> On Tue, Aug 2, 2022 at 1:13 PM Dan Cropp  wrote:
>
> Is the allow_routing setting on the geolocation Wiki Profile also not
> fully implemented?
>
>
>
> Well, 99% of the code is there.  The 1% is parsing the config option.  Not
> sure how I missed that.
>
> I'll have a patch up first thing in the morning UTC-6.
>
> I'll call it "allow_use_for_routing" in profile.
>
>
>
> Actually just "allow_routing_use"
>
>
>
>
>
>
>
> In the code, I see geolocation_routing used instead of allow_routing.
>
>
>
> Tried both and Asterisk indicates it cannot find suitable setting so it
> doesn’t create the profile object.
>
>
>
> Dan
>
>
>
> *From:* Dan Cropp
> *Sent:* Tuesday, August 2, 2022 10:04 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject:* RE: [External] [asterisk-users] Geo location 18.14.0-rc1
> question
>
>
>
> Thank you George.
>
>
>
> *From:* asterisk-users  *On
> Behalf Of *George Joseph
> *Sent:* Tuesday, August 2, 2022 9:57 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject:* Re: [External] [asterisk-users] Geo location 18.14.0-rc1
> question
>
>
>
>
>
>
>
> On Tue, Aug 2, 2022 at 8:46 AM Dan Cropp  wrote:
>
> I believe I have everything configured correctly, but Asterisk is
> complaining about my configuration
>
>
>
> It is complaining about confidence settings.
>
>
>
> From the Asterisk Geolocation Implementation Wiki, I believe I have this
> set correctly.
>
>
>
> Sub-parameters:
>
>- value: A percentage indicating the confidence or "unknown".
>- pdf: "unknown", "normal" or "rectangular"
>Example: confidence = value=80, pdf=unknown
>If no confidence parameter is specified, the default is 95%.
>See RFC7459
>
> <https://wiki.asterisk.org/wiki/display/AST/Geolocation+Reference+Information#GeolocationReferenceInformation-rfc7459>
>  for
>the exact definition of this parameter.
>
>
>
>
>
> [08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find option
> suitable for category 'IS_loc_1' named 'confidence' at line 12 of
>
> [08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create
> an object of type 'location' with id 'IS_loc_1' from configuration file
> 'geolocation.conf'
>
>
>
> [IS_loc_1]
>
> type = location
>
> format = civicAddress
>
> confidence = value=95, pdf=unknown
>
> location_info = country=US,A1=Wisconsin,A3=Madison
>
> location_info = HNO=4800,RD=Main,STS=Drive,PC=53704
>
>
>
> Remove the confidence param for now.I documented it before I
> implemented it. :)
>
>
>
>
>
>
>
> Also seeing problems with location_refinement setting.
>
> Again, I believe my setting matches what is on the Asterisk Geolocation
> Implementation wiki.
>
>
>
> [08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find option
> suitable for category 'IS_prof_20' named 'location_refinement' at line 56 of
>
> [08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create
> an object of type 'profile' with id 'IS_prof_20' from configuration file
> 'geolocation.conf'
>
>
>
> [IS_prof_20]
>
> type = profile
>
> profile_action = prefer_incoming
>
> pidf_element = person
>
> usage_rules = retransmission_allowed=no
>
> location_reference = IS_loc_22
>
> location_refinement = ROOM=292
>
> location_refinement = FLR=1
>
>
>
> Pffft.  I renamed this to "location_info_refinement" to better match the
> "location_info" parameter in the Location object.  I forgot to rename it in
> the wiki documentation.  If you just change the name it should work.
>
>
>
>
>
>
>
>
>
>
>
> Dan
>
> --
> ___

Re: [asterisk-users] [External] Geo location 18.14.0-rc1 question

[George Joseph]

On Tue, Aug 2, 2022 at 1:35 PM George Joseph  wrote:

>
>
> On Tue, Aug 2, 2022 at 1:13 PM Dan Cropp  wrote:
>
>> Is the allow_routing setting on the geolocation Wiki Profile also not
>> fully implemented?
>>
>
> Well, 99% of the code is there.  The 1% is parsing the config option.  Not
> sure how I missed that.
> I'll have a patch up first thing in the morning UTC-6.
> I'll call it "allow_use_for_routing" in profile.
>

Actually just "allow_routing_use"


>
>
>>
>>
>> In the code, I see geolocation_routing used instead of allow_routing.
>>
>>
>>
>> Tried both and Asterisk indicates it cannot find suitable setting so it
>> doesn’t create the profile object.
>>
>>
>>
>> Dan
>>
>>
>>
>> *From:* Dan Cropp
>> *Sent:* Tuesday, August 2, 2022 10:04 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
>> asterisk-users@lists.digium.com>
>> *Subject:* RE: [External] [asterisk-users] Geo location 18.14.0-rc1
>> question
>>
>>
>>
>> Thank you George.
>>
>>
>>
>> *From:* asterisk-users  *On
>> Behalf Of *George Joseph
>> *Sent:* Tuesday, August 2, 2022 9:57 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
>> asterisk-users@lists.digium.com>
>> *Subject:* Re: [External] [asterisk-users] Geo location 18.14.0-rc1
>> question
>>
>>
>>
>>
>>
>>
>>
>> On Tue, Aug 2, 2022 at 8:46 AM Dan Cropp  wrote:
>>
>> I believe I have everything configured correctly, but Asterisk is
>> complaining about my configuration
>>
>>
>>
>> It is complaining about confidence settings.
>>
>>
>>
>> From the Asterisk Geolocation Implementation Wiki, I believe I have this
>> set correctly.
>>
>>
>>
>> Sub-parameters:
>>
>>- value: A percentage indicating the confidence or "unknown".
>>- pdf: "unknown", "normal" or "rectangular"
>>Example: confidence = value=80, pdf=unknown
>>If no confidence parameter is specified, the default is 95%.
>>See RFC7459
>>
>> <https://wiki.asterisk.org/wiki/display/AST/Geolocation+Reference+Information#GeolocationReferenceInformation-rfc7459>
>>  for
>>the exact definition of this parameter.
>>
>>
>>
>>
>>
>> [08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find
>> option suitable for category 'IS_loc_1' named 'confidence' at line 12 of
>>
>> [08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create
>> an object of type 'location' with id 'IS_loc_1' from configuration file
>> 'geolocation.conf'
>>
>>
>>
>> [IS_loc_1]
>>
>> type = location
>>
>> format = civicAddress
>>
>> confidence = value=95, pdf=unknown
>>
>> location_info = country=US,A1=Wisconsin,A3=Madison
>>
>> location_info = HNO=4800,RD=Main,STS=Drive,PC=53704
>>
>>
>>
>> Remove the confidence param for now.I documented it before I
>> implemented it. :)
>>
>>
>>
>>
>>
>>
>>
>> Also seeing problems with location_refinement setting.
>>
>> Again, I believe my setting matches what is on the Asterisk Geolocation
>> Implementation wiki.
>>
>>
>>
>> [08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find
>> option suitable for category 'IS_prof_20' named 'location_refinement' at
>> line 56 of
>>
>> [08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create
>> an object of type 'profile' with id 'IS_prof_20' from configuration file
>> 'geolocation.conf'
>>
>>
>>
>> [IS_prof_20]
>>
>> type = profile
>>
>> profile_action = prefer_incoming
>>
>> pidf_element = person
>>
>> usage_rules = retransmission_allowed=no
>>
>> location_reference = IS_loc_22
>>
>> location_refinement = ROOM=292
>>
>> location_refinement = FLR=1
>>
>>
>>
>> Pffft.  I renamed this to "location_info_refinement" to better match the
>> "location_info" parameter in the Location object.  I forgot to rename it in
>> the wiki documentation.  If you just change the name it should work.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>

Re: [asterisk-users] [External] Geo location 18.14.0-rc1 question

[George Joseph]

On Tue, Aug 2, 2022 at 1:13 PM Dan Cropp  wrote:

> Is the allow_routing setting on the geolocation Wiki Profile also not
> fully implemented?
>

Well, 99% of the code is there.  The 1% is parsing the config option.  Not
sure how I missed that.
I'll have a patch up first thing in the morning UTC-6.
I'll call it "allow_use_for_routing" in profile.


>
>
> In the code, I see geolocation_routing used instead of allow_routing.
>
>
>
> Tried both and Asterisk indicates it cannot find suitable setting so it
> doesn’t create the profile object.
>
>
>
> Dan
>
>
>
> *From:* Dan Cropp
> *Sent:* Tuesday, August 2, 2022 10:04 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject:* RE: [External] [asterisk-users] Geo location 18.14.0-rc1
> question
>
>
>
> Thank you George.
>
>
>
> *From:* asterisk-users  *On
> Behalf Of *George Joseph
> *Sent:* Tuesday, August 2, 2022 9:57 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject:* Re: [External] [asterisk-users] Geo location 18.14.0-rc1
> question
>
>
>
>
>
>
>
> On Tue, Aug 2, 2022 at 8:46 AM Dan Cropp  wrote:
>
> I believe I have everything configured correctly, but Asterisk is
> complaining about my configuration
>
>
>
> It is complaining about confidence settings.
>
>
>
> From the Asterisk Geolocation Implementation Wiki, I believe I have this
> set correctly.
>
>
>
> Sub-parameters:
>
>- value: A percentage indicating the confidence or "unknown".
>- pdf: "unknown", "normal" or "rectangular"
>Example: confidence = value=80, pdf=unknown
>If no confidence parameter is specified, the default is 95%.
>See RFC7459
>
> <https://wiki.asterisk.org/wiki/display/AST/Geolocation+Reference+Information#GeolocationReferenceInformation-rfc7459>
>  for
>the exact definition of this parameter.
>
>
>
>
>
> [08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find option
> suitable for category 'IS_loc_1' named 'confidence' at line 12 of
>
> [08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create
> an object of type 'location' with id 'IS_loc_1' from configuration file
> 'geolocation.conf'
>
>
>
> [IS_loc_1]
>
> type = location
>
> format = civicAddress
>
> confidence = value=95, pdf=unknown
>
> location_info = country=US,A1=Wisconsin,A3=Madison
>
> location_info = HNO=4800,RD=Main,STS=Drive,PC=53704
>
>
>
> Remove the confidence param for now.I documented it before I
> implemented it. :)
>
>
>
>
>
>
>
> Also seeing problems with location_refinement setting.
>
> Again, I believe my setting matches what is on the Asterisk Geolocation
> Implementation wiki.
>
>
>
> [08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find option
> suitable for category 'IS_prof_20' named 'location_refinement' at line 56 of
>
> [08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create
> an object of type 'profile' with id 'IS_prof_20' from configuration file
> 'geolocation.conf'
>
>
>
> [IS_prof_20]
>
> type = profile
>
> profile_action = prefer_incoming
>
> pidf_element = person
>
> usage_rules = retransmission_allowed=no
>
> location_reference = IS_loc_22
>
> location_refinement = ROOM=292
>
> location_refinement = FLR=1
>
>
>
> Pffft.  I renamed this to "location_info_refinement" to better match the
> "location_info" parameter in the Location object.  I forgot to rename it in
> the wiki documentation.  If you just change the name it should work.
>
>
>
>
>
>
>
>
>
>
>
> Dan
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Geo location 18.14.0-rc1 question

[George Joseph]

On Tue, Aug 2, 2022 at 8:46 AM Dan Cropp  wrote:

> I believe I have everything configured correctly, but Asterisk is
> complaining about my configuration
>
>
>
> It is complaining about confidence settings.
>
>
>
> From the Asterisk Geolocation Implementation Wiki, I believe I have this
> set correctly.
>
>
>
> Sub-parameters:
>
>- value: A percentage indicating the confidence or "unknown".
>- pdf: "unknown", "normal" or "rectangular"
>Example: confidence = value=80, pdf=unknown
>If no confidence parameter is specified, the default is 95%.
>See RFC7459
>
> 
>  for
>the exact definition of this parameter.
>
>
>
>
>
> [08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find option
> suitable for category 'IS_loc_1' named 'confidence' at line 12 of
>
> [08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create
> an object of type 'location' with id 'IS_loc_1' from configuration file
> 'geolocation.conf'
>
>
>
> [IS_loc_1]
>
> type = location
>
> format = civicAddress
>
> confidence = value=95, pdf=unknown
>
> location_info = country=US,A1=Wisconsin,A3=Madison
>
> location_info = HNO=4800,RD=Main,STS=Drive,PC=53704
>

Remove the confidence param for now.I documented it before I
implemented it. :)


>
>
>
> Also seeing problems with location_refinement setting.
>
> Again, I believe my setting matches what is on the Asterisk Geolocation
> Implementation wiki.
>
>
>
> [08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find option
> suitable for category 'IS_prof_20' named 'location_refinement' at line 56 of
>
> [08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create
> an object of type 'profile' with id 'IS_prof_20' from configuration file
> 'geolocation.conf'
>
>
>
> [IS_prof_20]
>
> type = profile
>
> profile_action = prefer_incoming
>
> pidf_element = person
>
> usage_rules = retransmission_allowed=no
>
> location_reference = IS_loc_22
>
location_refinement = ROOM=292
>
> location_refinement = FLR=1
>

Pffft.  I renamed this to "location_info_refinement" to better match the
"location_info" parameter in the Location object.  I forgot to rename it in
the wiki documentation.  If you just change the name it should work.


>
>

>
>
>
>
> Dan
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Question about the Geo Location support being added

[George Joseph]

On Wed, Jul 27, 2022 at 11:02 AM Dan Cropp  wrote:

> Looking at the Asterisk wiki
>
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+Geolocation+Implementation
>

Just FYI, I'm in the process of clarifying and adding more info.  Should be
done Friday.

>
>
> I see the dial plan support the GeolocProfileCreate and there is support
> for GEOLOC_PROFILE settings to be set on the dial plan.
>
>
>
> We currently use AMI Originate support.  We may have dozens/hundreds of
> calls in the system and external to Asterisk, someone executes a behavior
> where we perform the Originate, if the party answers, we ConfBridge the
> necessary calls together.  It can be multiple calls and we never know when
> the total calls bridged together will need to be increased.  Because of the
> random increase in calls, we can’t use the Dial to bridge the parties
> together.
>
>
>
> The GEO Location information for the original caller can vary
> significantly because they could be WebRTC.  We are planning to require the
> setup of the Geo Location for each call to be provided to us (either via
> the incoming call or it may be provided from third party software).  Either
> way, we will know what the GEO Location to use for the Originate.  Trying
> to wrap my head around the best way to achieve this.
>

A real scenario to test!!!  Thanks!

>
>
> Using AMI Originate, is it possible to set the GEOLOC_PROFILE settings via
> the Variable header?
>

I've not tested this but you don't need to do it at all...

>
>
> My thought would be to configure an outgoing Geo Location profile for the
> PJSIP endpoint, but it would have the minimum settings.
>

Actually it would have a template specifying replacement channel variables.

When sending the AMI Originate, provide all the adjustments to the
> GEOLOC_PROFILE settings via the Variable.
>
>
>
> Is this possible or might there be a better way to achieve this?
>
>
>
It's possible but probably not needed.  Let's say you're using Civic
Address and a direct originate to the remote party via Dial.   In the
originate, you can specify regular, inherited channel variables with the
official Civic Address parameters preceded by '_'.  Let's use HNO (house
number) as an example.   You'd set _HNO=1633 in the originate and since it
has the '_' prefix it's going to be inherited by the outgoing channel.   In
the outgoing channel's profile/location, you'd set 'location_info =
HNO=${_HNO}.  Of course there'd be more than just the HNO parameter set but
it's the same technique.  The outgoing channel has a very generic location
template populated with values received from the incoming channel.

Now, this isn't going to work if you're originating both calls and adding
them to a bridge yourself but in this case, you have both channels at the
same time so you can just add the incoming channel's location info
directly to the outgoing channel's variables as you originate the outgoing
call.  Youdon';t need to create a new GEOLOC_PROFILE for the outgoing
channel.

All of this assumes that I actually understood your situation correctly. :)

How are you getting the caller's info in the first place?

>
>
> Alternatively, I could generate an internal local channel, configure the
> GeoLocProfile on it, configure all GEOLOC_PROFILE adjustments on it, then
> have it perform the Dial.  If the other end answers or not, treat it
> exactly as we currently do using the Originate.
>

Sounds more complicated than it needs to be.

>
>
>
>
> Dan
>
>
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>
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Re: [asterisk-users] Question on ExternalMedia and the codec

[George Joseph]

On Tue, Oct 12, 2021 at 2:54 PM Dan Cropp  wrote:

> We tell asterisk to use the slin format for ExternalMedia.  However, the
> unicast channel is selecting ulaw formatand the RTP data is indicating it’s
> ulaw format.
>
>
>
> Anyone know why ulaw format would be on chosen?
>

What do your ARI requests look like?  Are you just requesting "slin" or one
of the specific variants?



>
>
>
>
> [10/12 16:13:39.396] DEBUG[1665] http.c: HTTP Request URI is
> /ari/channels/externalMedia?app=a2519b4b-4d90-4d18-906b-717d02f8d569&external_host=192.168.32.148:8080
> &format=slin
>
> [10/12 16:13:39.396] DEBUG[1665] http.c: match request
> [ari/channels/externalMedia] with handler [static] len 6
>
> [10/12 16:13:39.396] DEBUG[1665] http.c: match request
> [ari/channels/externalMedia] with handler [ari] len 3
>
> [10/12 16:13:39.396] DEBUG[1665] http.c: Match made with [ari]
>
> [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Finding handler for
> channels/externalMedia
>
> [10/12 16:13:39.396] DEBUG[1665] res_ari.c:   Finding handler for channels
>
> [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking ari
> deviceStates:  Didn't match channels
>
> [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking ari
> applications:  Didn't match channels
>
> [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking ari
> channels:  Explicit match with channels
>
> [10/12 16:13:39.396] DEBUG[1665] res_ari.c:   Finding handler for
> externalMedia
>
> [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking channels
> create:  Didn't match externalMedia
>
> [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking channels
> channelId:  Matched wildcard.
>
> [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking channels
> externalMedia:  Explicit match with externalMedia
>
> [10/12 16:13:39.396] DEBUG[1665] acl.c: For destination '192.168.32.148',
> our source address is '192.168.33.34'.
>
> [10/12 16:13:39.396] DEBUG[1665] rtp_engine.c: Using engine 'asterisk' for
> RTP instance '0x7fef60018320'
>
> [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) RTP
> allocated port 12226
>
> [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE
> creating session 192.168.33.34:12226 (12226)
>
> [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE
> create
>
> [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE
> add system candidates
>
> [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE
> add candidate: 192.168.33.34:12226, 2130706431
>
> [10/12 16:13:39.396] DEBUG[1665] rtp_engine.c: RTP instance
> '0x7fef60018320' is setup and ready to go
>
> [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c:  :
> Formats: (none)
>
> [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c:  Channel is being
> initialized or destroyed
>
> [10/12 16:13:39.396] DEBUG[1665] stasis.c: Creating topic. name:
> channel:1634055219.4, detail:
>
> [10/12 16:13:39.396] DEBUG[1665] stasis.c: Topic 'channel:1634055219.4':
> 0x7fef6008d170 created
>
> [10/12 16:13:39.396] DEBUG[1665] channel.c: Channel 0x7fef6008a910
> 'UnicastRTP/192.168.32.148:8080-0x7fef60018320' allocated
>
> [10/12 16:13:39.396] DEBUG[1665] acl.c: For destination '192.168.32.148',
> our source address is '192.168.33.34'.
>
> [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c:
> UnicastRTP/192.168.32.148:8080-0x7fef60018320: Formats: (ulaw)
>
> [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c:  New topology set
>
> [10/12 16:13:39.396] DEBUG[1665] res_stasis.c:
> a2519b4b-4d90-4d18-906b-717d02f8d569: Subscribing to 1634055219.4
>
> [10/12 16:13:39.396] DEBUG[1665] stasis/app.c: Channel '1634055219.4' is 1
> interested in a2519b4b-4d90-4d18-906b-717d02f8d569
>
> [10/12 16:13:39.396] DEBUG[1665] http.c: HTTP keeping session open.
> status_code:200
>
> [10/12 16:13:39.396] DEBUG[1666] stasis/app.c: Channel '1634055219.4' is 2
> interested in a2519b4b-4d90-4d18-906b-717d02f8d569
>
>
>
> Have a good day!
>
> Dan
>
> This email is intended only for the use of the party to which it is
> addressed and may contain information that is privileged, confidential, or
> protected by law. If you are not the intended recipient you are hereby
> notified that any dissemination, copying or distribution of this email or
> its contents is strictly prohibited. If you have received this message in
> error, please notify us immediately by replying to the message and deleting
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> --
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>
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>
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>
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Re: [asterisk-users] Large system seeing single CPU core spiking

[George Joseph]

On Tue, Sep 14, 2021 at 9:19 AM Dan Cropp  wrote:

> Thank you George.
>
>
>
> It is using local file based configuration files.
>

Well, that's good at least.  It eliminates the database layer which can be
troublesome in virtualized environments, especially if a SAN and/or a
remote database server is used.

>
>
> Other factors.
>
> We run Asterisk in realtime mode to allow it to run as fast as possible.
>

Running at "realtime" level is usually NOT a good thing for Asterisk and
rarely needed when there are adequate resources.  Let's say you have a
local DNS resolver running.   If the system is stressed, Asterisk could
actually starve the resolver of resources, which then causes Asterisk to
back up waiting for DNS resolution to complete.  We've seen this happen
when using a database backend for configuration.  Someone thinks "I'll just
give Asterisk more resources" forgetting that Asterisk needs the database
engine to run.


>
>
> I just learned customer upgraded to 24 CPU cores.  Although, I’m not sure
> they actually assigned 24 physical cores to this machine or just increasing
> Hyper-V values.
>

How is this VM's priority versus other VMs on the same cluster?  Just
because it has 24 threads doesn't mean it's got 24 threads dedicated.  Does
using a realtime priority in the VM trickle down to Hyper-V's hypervisor's
resource management algorithms?


>
> I will monitor for additional information and see if the customer will
> allow me to capture a coredump when problems are happening.
>
> Waiting for them to report an incident.
>
>
>
> Here is a small sample of the system right now (24 cores), to the best of
> my knowledge it’s running fine.
>
>
>
> top -p 1509 -n 1 -H -b
>
> top - 15:06:32 up  9:06,  2 users,  load average: 6.02, 5.59, 5.26
>
> Threads: 1709 total,   8 running, 1701 sleeping,   0 stopped,   0 zombie
>
> %Cpu(s):  3.1 us,  2.5 sy,  0.0 ni, 94.3 id,  0.0 wa,  0.0 hi,  0.1 si,
> 0.0 st
>
> KiB Mem : 32143072 total, 29750072 free,  1016132 used,  1376868 buff/cache
>
> KiB Swap:  8388604 total,  8388604 free,0 used. 30697060 avail Mem
>
>
>
>PID USER  PR  NIVIRTRESSHR S %CPU %MEM TIME+ COMMAND
>
>   1830 root -11   0 13.741g 493680  28828 R 99.9  1.5 174:13.39
> asterisk
>
>   1541 root -11   0 13.741g 493680  28828 R 14.3  1.5  20:03.30
> asterisk
>
> 33601 root -11   0 13.741g 493680  28828 S  9.5  1.5   0:16.30 asterisk
>
> 46605 root -11   0 13.741g 493680  28828 S  9.5  1.5   0:30.06 asterisk
>
>   2295 root -11   0 13.741g 493680  28828 S  4.8  1.5  12:25.50
> asterisk
>
>   2297 root -11   0 13.741g 493680  28828 S  4.8  1.5   1:10.59
> asterisk
>

There's definitely one thread that's pegging a CPU.  If that thread is one
of the few "singleton" threads, that can be an issue.  What does "core show
taskprocessors" indicate?  Are there any that are hitting their limits?


>
>
>
>
> *From:* asterisk-users  *On
> Behalf Of *George Joseph
> *Sent:* Tuesday, September 14, 2021 9:39 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject:* Re: [asterisk-users] Large system seeing single CPU core
> spiking
>
>
>
>
>
>
>
> On Tue, Sep 14, 2021 at 8:07 AM Dan Cropp  wrote:
>
> I am working with a very large customer running Asterisk with PJSIP.
> Systems total channels have been over 2500 (which includes hundreds of
> local channels and ConfBridges) when the issues occur.
>
> It’s running on a Hyper-V VM with 12 CPU cores.
>
> Things work fine most of the time.
>
>
>
> They periodically see 10-30 minute periods where audio starts sounding
> like jitter buffer type issues.  Can literally have someone spelling their
> name and a ConfBridge recording of it shows the audio is missing a letter
> or two.
>
> The odd part is another system (not running Asterisk) was handling these
> calls previously.  The overall network has plenty of bandwidth (as
> evidenced by another system able to handle the call volume)
>
>
>
> One area that has perplexed us is when using htop, we see a single CPU
> core will spike to 100%.  Which core does keep changing.
>
>
>
> Asterisk is definitely the process using the vast majority of the CPU
> cycles.
>
>
>
> We recently found a setting on Hyper-V networking SR-IOV which improved
> things.  Prior to changing this setting, we were seeing SIP OPTIONS
> packets/responses would occasionally take more than 3 seconds causing
> devices to drop and come back online.
>
>
>
> We have configured a si

Re: [asterisk-users] Large system seeing single CPU core spiking

[George Joseph]

On Tue, Sep 14, 2021 at 8:07 AM Dan Cropp  wrote:

> I am working with a very large customer running Asterisk with PJSIP.
> Systems total channels have been over 2500 (which includes hundreds of
> local channels and ConfBridges) when the issues occur.
>
> It’s running on a Hyper-V VM with 12 CPU cores.
>
> Things work fine most of the time.
>
>
>
> They periodically see 10-30 minute periods where audio starts sounding
> like jitter buffer type issues.  Can literally have someone spelling their
> name and a ConfBridge recording of it shows the audio is missing a letter
> or two.
>
> The odd part is another system (not running Asterisk) was handling these
> calls previously.  The overall network has plenty of bandwidth (as
> evidenced by another system able to handle the call volume)
>
>
>
> One area that has perplexed us is when using htop, we see a single CPU
> core will spike to 100%.  Which core does keep changing.
>
>
>
> Asterisk is definitely the process using the vast majority of the CPU
> cycles.
>
>
>
> We recently found a setting on Hyper-V networking SR-IOV which improved
> things.  Prior to changing this setting, we were seeing SIP OPTIONS
> packets/responses would occasionally take more than 3 seconds causing
> devices to drop and come back online.
>
>
>
> We have configured a similar system running at Amazon handling far more
> traffic and can’t get the single CPU core to spike.  Only small static pops
> during the calls.
>
>
>
> The sheer scale of the system is making it hard to diagnose the problem.
>
>
>
> Any thoughts on how to diagnose what is causing the single CPU core to
> spike?
>
> Any thoughts on how to diagnose the problem?
>
> Any other thoughts/comments?
>


The first thing I'd do is see where the CPU is spending time: userspace,
system, nice, wait, etc.
Is it actually the asterisk process consuming the CPU?
Is Asterisk running with local file-based configs, local database, remote
database, etc?

If call quality is really bad already and your customer agrees, you could
try the following the next time it happens...
 1. Run "top -p `pidof asterisk` -n 1 -H -b" to get a list of all of
Asterisk's threads and their CPU utilization.
 2. Run ast_coredumper with the --RUNNING option.  This will pause Asterisk
while the dump is being generated!
 3. See if you can correlate the high cpu thread IDs from the top output to
the threads listed in the coredumper's -brief.txt file.

That _may_ give you an idea of where to look.



>
>
> Dan
>
> This email is intended only for the use of the party to which it is
> addressed and may contain information that is privileged, confidential, or
> protected by law. If you are not the intended recipient you are hereby
> notified that any dissemination, copying or distribution of this email or
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> error, please notify us immediately by replying to the message and deleting
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>
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Re: [asterisk-users] Between a dumb client and a capable server...

[George Joseph]

On Fri, Aug 20, 2021 at 2:33 PM Eric Wieling  wrote:

>
>
> On 8/20/21 4:24 PM, Antony Stone wrote:
> > On Friday 20 August 2021 at 19:06:09, George Joseph wrote:
> >
> >> On Fri, Aug 20, 2021 at 8:33 AM Antony Stone wrote:
> >>>
> >>> So, if I have Asterisk registered as a SIP client to some remote
> server,
> >>> how can I get Asterisk to tell that remote server to put the call on
> hold
> >>> (which a standard SIP telephone would normally do by sending a ReINVITE
> >>> with the SDP parameter 'sendonly')?
> >>
> >> On the outgoing pjsip endpoint, set "moh_passthrough = yes".   If you
> then
> >> put incoming call on hold, a reinvite with sendonly will be sent to the
> >> upstream server.
> >
> > So... how do I put the incoming call on hold, when the dumb client I'm
> > starting from cannot do that bit?
> >
> > I already know (from this list) that Asterisk as a SIP client cannot do
> ore
> > than (a) place a call, (b) answer a call, and (c) hang up a call.
> >
> > So, I'm still intrigued as to how you think this might be possible.
> >
> > If it *is* possible, I'd be really interested, but all my researches so
> far
> > suggest that Asterisk, acting in the middle like this, just cannot add
> the
> > necessary "put call on hold" which the original client cannot do.
> >
>
> With Asterisk, keep Asterisk in the media path with direct_media=yes and
> use DTMF to hold, transfer, and other features using features.conf.
> Asterisk has to stay in the media path when NAT is involved anyway.
>

You need to set direct_media=no to keep Asterisk in the media path.


>
> I doubt anything except Asterisk or other B2BUA software can do what you
> want.
>
> --
> http://help.nyigc.net/
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
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Re: [asterisk-users] Between a dumb client and a capable server...

[George Joseph]

On Fri, Aug 20, 2021 at 2:25 PM Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> On Friday 20 August 2021 at 19:06:09, George Joseph wrote:
>
> > On Fri, Aug 20, 2021 at 8:33 AM Antony Stone wrote:
> > >
> > > So, if I have Asterisk registered as a SIP client to some remote
> server,
> > > how can I get Asterisk to tell that remote server to put the call on
> hold
> > > (which a standard SIP telephone would normally do by sending a ReINVITE
> > > with the SDP parameter 'sendonly')?
> >
> > On the outgoing pjsip endpoint, set "moh_passthrough = yes".   If you
> then
> > put incoming call on hold, a reinvite with sendonly will be sent to the
> > upstream server.
>
> So... how do I put the incoming call on hold, when the dumb client I'm
> starting from cannot do that bit?
>
> I already know (from this list) that Asterisk as a SIP client cannot do
> ore
> than (a) place a call, (b) answer a call, and (c) hang up a call.
>

Yeah?  Who told you that? :)

>
> So, I'm still intrigued as to how you think this might be possible.
>
> If it *is* possible, I'd be really interested, but all my researches so
> far
> suggest that Asterisk, acting in the middle like this, just cannot add the
> necessary "put call on hold" which the original client cannot do.
>

Well, I can't tell you how you should do it but I can tell you how I might
attempt it based on what I know of your situation...

   - Create an ARI application in whatever scripting language you like that
   registers itself to Asterisk as "my3pcc" and also opens an HTTP listener
   for incoming 3PCC commands from your web app.
   - In your dialplan, send all incoming calls to Stasis(my3pcc).
   - The event listener (web socket) in your app tells you of the incoming
   call.
   - You create an outgoing channel to the upstream SIP server and bridge
   the two.
   - Your web app can get the channel id from your ARI app and tell your ARI
   app to do something with it, say place it on hold.
   - Your ARI app calls the  /channels/{channelId}/hold REST api.

It's actually a bit more complicated than that because you have to deal
with directionality.  You may have to create an intermediate local channel
to make sure the hold is sent in the correct direction.  You'd have to
experiment a bit.
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Re: [asterisk-users] Between a dumb client and a capable server...

[George Joseph]

On Fri, Aug 20, 2021 at 8:33 AM Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> On Friday 20 August 2021 at 16:14:44, George Joseph wrote:
>
> > On Wed, Aug 18, 2021 at 3:33 AM Antony Stone wrote:
> > > Hi.
> > >
> > > Just to summarise: I have a SIP client talking to a SIP server, and I
> > > need something which can send commands to that server to put calls,
> > > which were created by the existing client, on hold (that's the simplest
> > > scenario).  I do not want to build a SIP server / PBX myself which can
> > > itself perform call hold & transfer etc (I know how to do that with
> > > Asterisk) - I need those functions to be performed by the existing
> server.
> >
> > Sounds like you're looking for something to do 3rd Party Call Control
> > (3PCC).
>
> Okay, that sounds like useful terminology.
>
> > It also sounds like the 'SIP server" isn't Asterisk and you can't change
> > that either right?
>
> It *might* be Asterisk, but if it is, I have no access to it other than
> the
> SIP credentials a standard telephone would use to register to it.  Then
> again,
> I might not even *know* what it is - it's just a SIP-based PBX...
>
> > You could actually use a tiny Asterisk instance to do this.
>
> Hm, I'm very dubious about that, based on what I've seen in docs so far...
>
> > The dumb client would call Asterisk and Asterisk would simply send the
> call
> > to your existing SIP server.
>
> Okay, so far, so good, I can get Asterisk to do that.
>
> > You could then use AMI or ARI to watch for the call events and tell
> > Asterisk to transfer to some other extension on your SIP server or
> whatever.
>
> So, let's just take the simplest example - how can I get Asterisk to tell
> the
> other server to put a call on hold and play that other server's hold music
> to
> the remote party?
>
> > The big question is...  what triggers the action to take?
>
> That's easy, I have a web interface which is on the same machine as the
> dumb
> SIP softphone, and that can talk to this "tiny Asterisk server" you
> speculate
> about, for example by sending in AMI Originate commands to it, which can
> trigger dial plan actions, which can do anything Asterisk is capable of.
>
> My doubts are whether Asterisk as a SIP *client* is capable of this.
>
> So, if I have Asterisk registered as a SIP client to some remote server,
> how
> can I get Asterisk to tell that remote server to put the call on hold
> (which a
> standard SIP telephone would normally do by sending a ReINVITE with the
> SDP
> parameter 'sendonly')?
>

On the outgoing pjsip endpoint, set "moh_passthrough = yes".   If you then
put
incoming call on hold, a reinvite with sendonly will be sent to the upstream
server.


>
>
> Thanks,
>
>
> Antony.
>
> --
> "The future is already here.   It's just not evenly distributed yet."
>
>  - William Gibson
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
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> https://community.asterisk.org/
>
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Re: [asterisk-users] Between a dumb client and a capable server...

[George Joseph]

On Wed, Aug 18, 2021 at 3:33 AM Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> Hi.
>
> I wonder if anyone has some helpful advice or suggestions for me?
>
>
snip

I had thought that Kamailio might be what I was looking for, but I've asked
> on
> their mailing list and people are telling me that it isn't, and that I
> need
> something like Asterisk to do this.  I'm trying to get some specifics from
> them
> about *how* I would get Asterisk to do this (because I personally can't
> see
> how Asterisk could sit between a SIP client and a SIP server, and generate
> commands to manipulate the RTP stream and send them to the server, which
> is
> what the Kamailio people are saying I should do), but I thought it was
> worth
> asking here just in case what they're telling me is in fact quite easy
> when
> you only know enough about Asterisk.
>
> So, if someone here thinks this is possible using Asterisk, please could
> you
> point me at some documentation showing what commands I would use or the
> basics
> of how I should go about it?
>
> If anyone thinks there is another, perhaps better, way of achieving this,
> then
> I'm quite open to alternative solutions (as I say, I was initially
> thinking
> that Kamailio might be the way forward), so anything that shows me *how*
> such
> a thing might be achieved, with any tool at all, would be very welcome.
>
> Just to summarise: I have a SIP client talking to a SIP server, and I need
> something which can send commands to that server to put calls, which were
> created by the existing client, on hold (that's the simplest scenario).  I
> do
> not want to build a SIP server / PBX myself which can itself perform call
> hold
> & transfer etc (I know how to do that with Asterisk) - I need those
> functions
> to be performed by the existing server.
>
>
Sounds like you're looking for something to do 3rd Party Call Control
(3PCC).
It also sounds like the 'SIP server" isn't Asterisk and you can't change
that either
right?

You could actually use a tiny Asterisk instance to do this. The dumb client
would
call Asterisk and Asterisk would simply send the call to your existing SIP
server.
You could then use AMI or ARI to watch for the call events and tell
Asterisk to
transfer to some other extension on your SIP server or whatever.
The big question is...  what triggers the action to take?
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Re: [asterisk-users] AST-2021-008: Remote crash when using IAX2 channel driver

[George Joseph]

On Fri, Jul 23, 2021 at 6:12 AM Doug Lytle  wrote:

> >>> Asterisk Project Security Advisory - AST-2021-008
>
> Downloading asterisk-16-current.tar.gz is still showing Asterisk 16.19.0
>

Links should be fixed now.



>
> Doug
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Re: [asterisk-users] Patch to remove numbers from the logs

[George Joseph]

On Thu, Jul 8, 2021 at 3:58 PM Dovid Bender  wrote:

> Hi,
>
> We have a project where people will be making payments over the phone. I
> would like block Asterisk from logging any time the system is processing a
> card. So be it SayDigits(123456789), when the user enters DTMF or when I
> pass a card number as a variable to an AGI etc. I assume this affects
> others and I would like to have the patch created in a way that a. will be
> accepted by Sangoma and b. will work for anyone else that has this issue.
>

Are you talking strictly about normal messages generated by the dialplan or
all messages, even warnings, errors, etc generated internally?



> My idea was to have a channel variable
> for exampleSet(CHANNEL(LOG_DIGITS)=OFF) and then have ast_logger check to
> see if the variable is set.
>

Would you need to do this on a channel-by-channel basis or could you set a
global variable?


> The problem I faced that wherever the logger is called a string is passed.
> So any digits (e.g. channel ID, thread ID etc.) would have the digits
> removed which I assume would hurt people. My solution was to have a
> configuration file where you would put in regex strings that we would
> replace. For instance if I set LOG_DIGITS=OFF and in the Dialplan I had
> CARD=4111
> EXP_MOTH=12
> EXP_YEAR=2025
>

Are those variables set on the channel?


>
> In the configuration file I would have
> CARD=([0-9]{15,16})
> EXP_MONTH=([1-2]?[0-9])
> EXP_YEAR=(202[5-9])
>

I'd skip the config file and  make teh regexes global dialplan variables.


> The system would then look for any of the above expressions and then
> replace the numbers with an X. Does that seem like a patch that would get
> accepted? is that completely in left field? Any thoughts on a better way of
> doing it? I know I can change the verbosity to 0 but then I would get
> nothing at all in the logs.
>
> TIA.
>
> Dovid
>
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Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

[George Joseph]

On Thu, Oct 22, 2020 at 4:13 PM David Cunningham 
wrote:

> Hi George,
>
> Thank you for the response. I'm a little unclear on what you mean by a
> transport. We're using chan_sip, not pjsip.
>
> Do you mean a device in sip.conf, using bindaddr to set the address to
> bind for that device? We've only used bindaddr in the [general] section
> before, but if it will work in a device that could be the answer.
>

Sorry.  I just assume chan_pjsip these days.  Not sure how you'd do it for
chan_sip.



>
>
> On Fri, 23 Oct 2020 at 00:13, George Joseph  wrote:
>
>>
>>
>> On Wed, Oct 21, 2020 at 9:16 PM David Cunningham <
>> dcunning...@voisonics.com> wrote:
>>
>>> Hello,
>>>
>>> We have an Asterisk server with two public IP addresses, let's say
>>> 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with
>>> a call dialled from Asterisk to an external destination. The external
>>> destination sees the SIP packet as coming from 1.1.1.1 and the media
>>> address in the SDP is 1.1.1.1, which is great.
>>>
>>> However if we receive a call in to 2.2.2.2 then the call dialled from
>>> Asterisk to an external destination still comes from 1.1.1.1, whereas we
>>> want it to come from 2.2.2.2. The source of any dialled call (the IP packet
>>> and the SDP media address) should be the same as the address the related
>>> inbound call was received to.
>>>
>>> For example:
>>> INVITE received to 1.1.1.1:5060 -> Asterisk dials
>>> destinat...@termination.com -> INVITE sent from 1.1.1.1:5060 to
>>> termination.com
>>> INVITE received to 2.2.2.2:5060 -> Asterisk dials destinat...@pstn.com
>>> -> INVITE sent from 2.2.2.2:5060 to pstn.com
>>>
>>> Does anyone know how this can be achieved?
>>>
>>
>> If termination.com is only on 1.1.1.1 and pstn.com is only on 2.2.2.2,
>> create 2 transports, one specifically bound to 1.1.1.1, transport-1.1.1.1
>> for instance, and another to 2.2.2.2:  transport-2.2.2.2.  The names
>> aren't important as long as you can tell the difference.  Then explicitly
>> configure endpoint termination.com's "transport" parameter to
>> "transport-1.1.1.1" and pstn.com's "transport" parameter to
>> "transport-2.2.2.2".   In your dialplan, you can see which endpoint the
>> call came in on, and route it out the same endpoint.
>>
>> If both providers are available from both interfaces, you can create 2
>> endpoint for each provider: termination.com-1.1.1.1, pstn.com-1.1.1.1,
>> termination.com-2.2.2.2 and pstn.com-2.2.2.2;  Then configure each with the
>> same transports as above.
>>
>>
>>
>>
>>
>>>
>>> Thanks in advance for your help,
>>>
>>> --
>>> David Cunningham, Voisonics Limited
>>> http://voisonics.com/
>>> USA: +1 213 221 1092
>>> New Zealand: +64 (0)28 2558 3782
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
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>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>> --
>> George Joseph
>> Asterisk Software Developer
>> direct/fax +1 256 428 6012
>> Check us out at www.sangoma.com and www.asterisk.org
>> [image: image.png]
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
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>> https://community.asterisk.org/
>>
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>>
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>
>
>
> --
> David Cunningham, Voisonics Limited
> http://voisonics.com/
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782
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Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

[George Joseph]

On Wed, Oct 21, 2020 at 9:16 PM David Cunningham 
wrote:

> Hello,
>
> We have an Asterisk server with two public IP addresses, let's say 1.1.1.1
> and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with a call
> dialled from Asterisk to an external destination. The external destination
> sees the SIP packet as coming from 1.1.1.1 and the media address in the SDP
> is 1.1.1.1, which is great.
>
> However if we receive a call in to 2.2.2.2 then the call dialled from
> Asterisk to an external destination still comes from 1.1.1.1, whereas we
> want it to come from 2.2.2.2. The source of any dialled call (the IP packet
> and the SDP media address) should be the same as the address the related
> inbound call was received to.
>
> For example:
> INVITE received to 1.1.1.1:5060 -> Asterisk dials
> destinat...@termination.com -> INVITE sent from 1.1.1.1:5060 to
> termination.com
> INVITE received to 2.2.2.2:5060 -> Asterisk dials destinat...@pstn.com ->
> INVITE sent from 2.2.2.2:5060 to pstn.com
>
> Does anyone know how this can be achieved?
>

If termination.com is only on 1.1.1.1 and pstn.com is only on 2.2.2.2,
create 2 transports, one specifically bound to 1.1.1.1, transport-1.1.1.1
for instance, and another to 2.2.2.2:  transport-2.2.2.2.  The names aren't
important as long as you can tell the difference.  Then explicitly
configure endpoint termination.com's "transport" parameter to
"transport-1.1.1.1" and pstn.com's "transport" parameter to
"transport-2.2.2.2".   In your dialplan, you can see which endpoint the
call came in on, and route it out the same endpoint.

If both providers are available from both interfaces, you can create 2
endpoint for each provider: termination.com-1.1.1.1, pstn.com-1.1.1.1,
termination.com-2.2.2.2 and pstn.com-2.2.2.2;  Then configure each with the
same transports as above.





>
> Thanks in advance for your help,
>
> --
> David Cunningham, Voisonics Limited
> http://voisonics.com/
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
George Joseph
Asterisk Software Developer
direct/fax +1 256 428 6012
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] Codec question

[George Joseph]

On Wed, Jun 17, 2020 at 11:13 AM Jerry Geis  wrote:

> I see this device :
> Axis C8033 Audio Bridge Quick Specs:
> Communications Protocol: SIP.
> Ethernet Ports: 1x 10/100.
> PoE: 802.3af/at Type 1 Class 2.
> Additional Interfaces:
> Audio: one-way/two-way, mono.
> Audio Codecs: G.711, G.726, WAV, MP3.
> Edge Storage: microSD, microSDHC, microSDXC.
> Operating Temperature: 4°F - 122°F.
>
> What is Codec WAV and MP3  to asterisk ???
>

The literals "wav" and "mp3" are unknown as far as media handling goes.
 We'd need to see what payload types they translate them to.   If you can
get a real SDP from one of those devices we could say more.



>
> Jerry
>
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Re: [asterisk-users] x-ast-orig-host - How is this IP taken ?

[George Joseph]

On Wed, Jun 10, 2020 at 3:25 PM Administrator  wrote:

> Hi list,
>
> We have a strange behavior: a customer Snom300 behind a public FW has
> contact like
>
> contact  :
> sip:user@x.y.39.147:2048;x-ast-orig-host=169.254.252.1:2048


x-ast-orig-host is a header we add to incoming requests when
rewrite_contact is on AND the host we get the request from is different
from the host in the contact URI.  We do this so we can restore the
original contact URI when we send responses.  Here's the scenario...

A client behind a firewall sends Asterisk a REGISTER request.  The contact
URI is probably going to be a non-routable ip address like 192.168.0.1 but
the host the packet comes from will be the public ip address of the
firewall.   In order to properly route responses and subsequent requests,
the "rewrite_contact" option can be used to force Asterisk to substitute
the private ip address in the contact header with the public ip address we
actually got the packet from.  This way we send responses and new requests
to the public ip address. This all works well except for 1 scenario...
 When a client sends a REGISTER request, they can use the IP address in the
contact header of the response to match it to the request.  If we've
rewritten the contact header, they won't be able to match it.   So we save
off the contact host into that x-ast-orig-host header and when we send
responses back to the client, we still send it to the public ip address git
we reset the contact host back to what was in the original request.  We
then strip all x-ast* headers before we actually send the packets.


>
> The phone can place calls but not receive any. Also, qualify give
> unreachable which seems correct when looking the x-ast-orig-host IP.
> Problem is that the local IP of this phone is 192.168.1.75
>

Well, 169.254.x.x addresses are Automatic Assigned Ip Addresses assigned by
the device itself when it can't get a dhcp ip address.  It's highly
unlikely that things are going to function normally if the device doesn't
have a real ip address.


>
> Question: how asterisk sets this IP ? It looks for us like a FW issue as
> we have other customers with approaching local network organisation and
> which are not facing this problem.
>

See above.  You should also check that the router the phone is connected to
does NOT have SIP ALG turned on because that will mess with the SIP headers.


>
> Thanks for any hint.
>
> --
> Daniel
>
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Check us out at www.sangoma.com and www.asterisk.org
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[asterisk-users] Advanced Codec Negotiation: Need info and uses cases

[George Joseph]

s are pjsip or not?  Of course, if
*neither* is pjsip, none of the above applies and the old process is used.

I know this is a lot to take in but I'd implore you to read thoroughly,
respond with real life scenarios and ask questions if something isn't
clear.   We are NOT going to shove this into 18 without everyone
understanding the implications, and if the process gets too complex, we'll
NEVER put it in because it'll work no better and be no better understood
than the current process.

THANKS!


-- 
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direct/fax +1 256 428 6012
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Re: [asterisk-users] Asterisk and CentOS 8

[George Joseph]

On Sun, May 3, 2020 at 6:07 PM Patrick Wakano  wrote:

> Hello George,
> Hope this finds you well!
> I wonder if there has been any progress on this matter?
>

Sorry, I missed this last week.   We have an internal issue open to revisit
CentOS 8 and it's at the top of our backlog right now so we should have
something in the next few weeks.

Thanks for sharing your experience!


>
> I've been working to have Asterisk running on CentOS 8 and our jump from
> CentOS 6 to 8 doesn't look too bad The missing packages found are:
> gmime-devel, iksemel-devel, corosynclib-devel, libresample-devel, hoard and
> python-devel. Python-devel could be replaced by python2-devel or
> python3-devel (python36-devel), but I am not sure if there is any python
> incompatibility... gmime-devel, iksemel-devel and hoard seems of no use
> (and they were already missing for CentOS 6 too), so only corosynclib (if
> you are using it) and libresample may be a problem Does anyone knows
> what is the use of libresample-devel package?
>
> Just to share the experience so far, to build Asterisk it is needed to
> install the epel-release and enable PowerTools. By ignoring the missing
> packages, the compilation works and we can start Asterisk. My only concern
> was about the libresample-devel because in the past I think its absence
> caused issues when loading/starting pjsip modules, but it didn't happen
> this time (but I am not using pjsip, I just loaded it with the sample
> config files)
> Following on and making Asterisk a systemd service instead of using
> init.d, I first checked the service file under contrib/systemd but it seems
> to serve a different purpose, so it could not be used. Instead a very
> simple one does the job. Just make sure your exec cmd does not have the
> '-c' option (used in safe_asterisk), because this was causing me a 100% CPU
> usage (related to this:
> https://unix.stackexchange.com/questions/191621/systemd-service-using-100-of-my-cpu-when-it-doesnt-if-i-start-it-without-syste/368037#368037
> )
> Be aware that SELinux may cause permissions and access problems when
> running as systemd service It took me a whole day to figure out that
> under our configuration (that have included files outside the /etc/asterisk
> folder) the "problem" was SELinux.
> Finally, configure firewalld to allow the SIP and RTP ports and you should
> be ready to go.
> Also, we are using rpmbuild to create an Asterisk package. rpmbuild (or
> new red hat policies) is more strict and now complains about the python
> shebangs that do not have a version (files contrib/script/ref*.py -> what
> is this used for??). When using the DONT_OPTMIZE flag the compiler warns
> for every single file saying, so it is quite annoying but does not to cause
> issues:
> /usr/include/features.h:381:4: warning: #warning _FORTIFY_SOURCE requires
> compiling with optimization (-O) [-Wcpp].
> Anyway, these problems do not happen if you manually build with the simple
> configure and make commands.
>
> Cheers,
> Patrick Wakano
>
> On Fri, 18 Oct 2019 at 11:54, Carlos Chavez  wrote:
>
>> They only problem I have found so far is while trying to install
>> Alembic for SQLAlchemy (for realtime configs).  Those are the only packages
>> that I cannot get working properly.  Vanilla Asterisk works fine  with the
>> only extra package needed being libedit-devel that is not included in any
>> "official" repo.  You need to download the Fedora Core 29 packages to in
>> order to successfully compile Asterisk.  That being said, I would not
>> recommend trying to put this in production any time soon.
>> On 10/17/2019 11:19 AM, George Joseph wrote:
>>
>> At the current time, we do not recommend attempting to build Asterisk on
>> CentOS 8.  Many packages Asterisk uses are not yet available and would
>> require building from their sources.  The Asterisk packages are also not
>> available in the EPEL 8 or CentOS 8 repositories yet for the same reason.
>>
>> We'll update you when we think it's safe.
>>
>>
>> --
>> *George Joseph*
>> Digium - A Sangoma Company | Software Developer | Software Engineering
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>> direct/fax: +1 256 428 6012
>> Check us out at: https://digium.com · https://sangoma.com
>>
>>
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>>
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Re: [asterisk-users] PJSIP Lockup

[George Joseph]

gt;>> It uses the same underlying API and layer. It can do more frequent
>>> database access though due to queries and because PJSIP is multithreaded.
>>>
>>> --
>>> Joshua C. Colp
>>> Asterisk Technical Lead
>>> Sangoma Technologies
>>> Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] [asterisk-dev] Need feedback on the use of AMI events generated by MESSAGE requests

[George Joseph]

On Thu, Feb 6, 2020 at 12:34 PM sdut...@wazo.io  wrote:

> On 1/29/20 2:31 PM, George Joseph wrote:
> > For those of you who actually process SIP MESSAGE requests...  Do you
> > use any of the AMI events generated by the "Message/ast_msg_queue"
> > channel?   We want to change that channel to an "internal" channel that
> > doesn't generate AMI events (for performance reasons) but we need to
> > know if anyone's using them first.
> >
> > Thanks!
>
> Hi George,
>
> could you give us a summary list of the impacted AMI messages? More
> specifically, are there AMI messages explicitly generated by
> Message/ast_msq_queue? Or are we talking about Newchannel, NewExten and
> other messages implicitly sent on the AMI because Message is a channel
> like any other?
>
> Note: please keep me in CC, I am not subscribed to asterisk-users
> mailing list.
>

Here's a copy of the commit message which should explain things...

   message.c: Add option to suppress the Message channel AMI and ARI events

In order to reduce the amount of AMI and ARI events generated,
the global "Message/ast_msg_queue" channel can be set to suppress
it's normal channel housekeeping events such as "Newexten",
"VarSet", etc. This can greatly reduce load on the manager
and ARI applications when the Digium Phone Module for Asterisk
is in use.  To enable, set "hide_messaging_ami_events" in
asterisk.conf to "yes"  In Asterisk versions <18, the default
is "no" preserving existing behavior.  Beginning with
Asterisk 18, the option will default to "yes".

NOTE:  This change does not affect UserEvents or the ARI
TextMessageReceived events.

* Added the "hide_messaging_ami_events" option to asterisk.conf.

* Changed message.c to set the AST_CHAN_TP_INTERNAL property on
  the "Message/ast_msg_queue" channel if the option is set in
  asterisk.conf.  This suppresses the reporting of the events.




>
> --
> Sébastien Duthil
>


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Re: [asterisk-users] Need feedback on the use of AMI events generated by MESSAGE requests

[George Joseph]

On Thu, Jan 30, 2020 at 12:18 AM Jean Aunis  wrote:

> Hello,
>
> I use UserEvents generated by the Message/ast_message_queue channel with
> the UserEvent application.
>
> Regards
>
> Jean
>

Thanks Jean.  We're looking at alternatives.



> Le 29/01/2020 à 20:31, George Joseph a écrit :
>
> For those of you who actually process SIP MESSAGE requests...  Do you use
> any of the AMI events generated by the "Message/ast_msg_queue" channel?
>  We want to change that channel to an "internal" channel that doesn't
> generate AMI events (for performance reasons) but we need to know if
> anyone's using them first.
>
> Thanks!
> --
> George Joseph
> Asterisk Software Developer
> direct/fax +1 256 428 6012
> Check us out at www.sangoma.com and www.asterisk.org
> [image: image.png]
>
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> https://community.asterisk.org/
>
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



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[asterisk-users] Need feedback on the use of AMI events generated by MESSAGE requests

[George Joseph]

For those of you who actually process SIP MESSAGE requests...  Do you use
any of the AMI events generated by the "Message/ast_msg_queue" channel?
 We want to change that channel to an "internal" channel that doesn't
generate AMI events (for performance reasons) but we need to know if
anyone's using them first.

Thanks!
-- 
George Joseph
Asterisk Software Developer
direct/fax +1 256 428 6012
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] What is PJSIP equivalent of users.conf hassip setting ?

[George Joseph]

On Mon, Dec 30, 2019 at 4:02 AM Olivier  wrote:

> Hello,
>
> In /etc/asterisk/users.conf, you can set hassip=yes to declare a chansip
> entity.
> Is there any equivalent for PJSIP ?
>

No.  users.conf doesn't support pjsip and there are no plans for it to do
so.

If you're moving to the chan_pjsip channel driver, check out...
https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Wizard



>
> Best regards
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Re: [asterisk-users] pjsip: How is asterisk choosing the IP address to put in the Contact header?

[George Joseph]

On Fri, Nov 29, 2019 at 8:30 AM Benoit Panizzon 
wrote:

> Short update...
>
> After some more research I found:
>
> https://community.asterisk.org/t/box-with-2-interfaces-wrong-one-chosen-in-contact-header/74705/3
>
> And some more similar ones describing the same problem with chan_sip
> and pjsip.
>
> I attempted to set: external_signaling_address on my transports. Also
> trying to trick them there could be NAT (there is none) by setting.
> local_net=192.168.99.0/24
>
> Asterisk is still sending the wrong IP Address in the Contact header of
> 183 or 200 messages.
>
> Can anyone confirm this is a bug?
>

There's been a lot of work in this area since 13.18.  Without seeing packet
captures of the incoming request and the response and seeing the exact
configuration of the endpoints and transports it's hard to say what's going
on.  One thing you can try is setting "symmetric_transport=yes" on the two
transports.  That should ensure that the responses go out the same
transport as the one that received the request.  That, in turn, should
change the contact address accordingly.




>
> Mit freundlichen Grüssen
>
> -Benoît Panizzon-
> --
> I m p r o W a r e   A G-Leiter Commerce Kunden
> __
>
> Zurlindenstrasse 29 Tel  +41 61 826 93 00
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[asterisk-users] [asterisk-app-dev] Proposed change to External Media API

[George Joseph]

When we created the External Media addition to ARI we created an
ExternalMedia object to be returned from the channels/externalMedia REST
endpoint.  This object contained the channel object that was created plus
local_address and local_port attributes (which are also in the Channel
variables).  At the time, we thought that creating an ExternalMedia object
would give us more flexibility in the future but as we created the sample
speech to text application, we discovered that it doesn't work so well with
ARI client libraries that a) don't have the ExternalMedia object defined
and/or b) can't promote the embedded channel structure to a first-class
Channel object.

Example:

A common pattern using the node-ari-client is to create a new Channel
object, attach an event handler to it, then call originate on it like so...

chan = ari.channels.Channel();
chan.on('StasisStart', );
chan.originate(...);

With the current ExternalMedia API:

chan = ari.channels.Channel();
chan.on('StasisStart', );
chan.externalMedia(...);

This doesn't work however because the return from channels/externalMedia
isn't a Channel.  It's an ExternalMedia object with an chlld object that
looks like a Channel but has no Channel behavior attached to it.  The event
handler added to chan will never get called and you can't attach handlers
or perform any operations on ExternalMedia.channel because it's just a
plain object, not an instance of Channel.

Realistically, it doesn't make sense to force client library
implementations to create special logic to promote the
ExternalMedia.channel object into an instance of Channel and since External
Media is a new capability anyway, it seems that the least painful solution
is to remove the ExternalMedia object and have channels/externalMedia
return a Channel object directly, just like channels/create and
channels/originate.  As I described above, the only other attributes of
ExternalMedia were the local address and port and they're already available
in the Channel variables anyway.

I would think that this change would make things easier for ARI developers
but I wanted to make sure that you knew about it in advance and had a
chance to comment.  There will be a Gerrit review up for this change later
this morning.

Also...  I mentioned the "sample speech to text application" above.  It's
working and will be published next week.

Comments?  Questions?

-- 
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Digium - A Sangoma Company | Software Developer | Software Engineering
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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[asterisk-users] Asterisk and CentOS 8

[George Joseph]

At the current time, we do not recommend attempting to build Asterisk on
CentOS 8.  Many packages Asterisk uses are not yet available and would
require building from their sources.  The Asterisk packages are also not
available in the EPEL 8 or CentOS 8 repositories yet for the same reason.

We'll update you when we think it's safe.


-- 
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Re: [asterisk-users] problem with new install with asterisk 15.7.4

[George Joseph]

Oh, I forgot to mention that Asterisk 15 went End-Of-Life last Thursday.
:)   You should use Asterisk 16.

On Mon, Oct 7, 2019 at 5:58 AM George Joseph  wrote:

>
>
> On Fri, Oct 4, 2019 at 1:19 PM John Covici  wrote:
>
>> Hi.  I am trying to install asterisk 15.7.4 from git onto a Debian 10
>> system and I am running into the following problem.  I need to install
>> meetme (I know its old), and I have dahdi installed and the configure
>> script answers yes to all the edahdi questions, but the app_meetme
>> says depends on dahdi (e).  I did not install libpri as I have no
>> hardware of that type.
>>
>
> The (E) means "external" not "error".   Does the app_meetme entry in
> menuselect have "[ ]" before it or "XXX"?
> If "[ ]" you should be able to select it and build.
>
>
>>
>> I installed dahdi from git and have the kernel sources and it
>> installed without errors.
>>
>> How can I fix?
>>
>> Thanks in advance for any suggestions.
>>
>> --
>> Your life is like a penny.  You're going to lose it.  The question is:
>> How do
>> you spend it?
>>
>>  John Covici wb2una
>>  cov...@ccs.covici.com
>>
>> --
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>
>
>
> --
> *George Joseph*
> Digium - A Sangoma Company | Software Developer | Software Engineering
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> direct/fax: +1 256 428 6012
> Check us out at: https://digium.com · https://sangoma.com
>
>

-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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Re: [asterisk-users] trouble building dahdi on kernel 5.2.7

[Anthony Joseph Messina]

On Wednesday, August 14, 2019 5:12:52 PM CDT sean darcy wrote:
> On 8/14/19 6:00 PM, sean darcy wrote:
> > dahdi built fine on 5.1.20, but on 5.2.7:
> > 
> > .
> > 
> >CC [M]
> > 
> > /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader
> > /dahdi_vpmadt032_loader.o> 
> >SHIPPED
> > 
> > /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader
> > /vpmadt032_x86_64.o> 
> >LD [M]
> > 
> > /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/dahdi_vpmadt032_
> > loader.o> 
> >Building modules, stage 2.
> >MODPOST 15 modules
> > 
> > ERROR: "vpmadtreg_register"
> > [/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/dahdi_vpmadt032
> > _loader.ko] undefined!
> > ERROR: "vpmadtreg_unregister"
> > [/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/dahdi_vpmadt032
> > _loader.ko] undefined!
> > make[2]: *** [scripts/Makefile.modpost:91: __modpost] Error 1
> > make[1]: *** [Makefile:1605: modules] Error 2
> > make[1]: Leaving directory '/usr/src/kernels/5.2.7-100.fc29.x86_64'
> > make: *** [Makefile:74: modules] Error 2
> > error: Bad exit status from /var/tmp/rpm-tmp.F8F4dL (%prep)
> > 
> > Any ideas ?
> > 
> > sean
> 
> And yes, kernel-devel is installled.
> 
> kernel-5.1.20-200.fc29.x86_64
> kernel-5.1.21-200.fc29.x86_64
> kernel-5.2.7-100.fc29.x86_64
> kernel-core-5.1.20-200.fc29.x86_64
> kernel-core-5.1.21-200.fc29.x86_64
> kernel-core-5.2.7-100.fc29.x86_64
> kernel-devel-5.1.20-200.fc29.x86_64
> kernel-devel-5.1.21-200.fc29.x86_64
> kernel-devel-5.2.7-100.fc29.x86_64
> kernel-headers-5.2.7-100.fc29.x86_64
> kernel-modules-5.1.20-200.fc29.x86_64
> kernel-modules-5.1.21-200.fc29.x86_64
> kernel-modules-5.2.7-100.fc29.x86_64
> kernel-tools-5.2.7-100.fc29.x86_64
> kernel-tools-libs-5.2.7-100.fc29.x86_64
> 
> The same kernel packages as the 5.1 kernels.
> 
> sean

Other F30 ix86 build errors not appearing to be related to yours, Sean.  These 
are with DAHDI git master branch (at v3.1.0-rc1).  What DAHDI version are you 
building?

https://issues.asterisk.org/jira/browse/DAHLIN-371

make[1]: Entering directory '/usr/src/kernels/5.2.8-200.fc30.i686'
  Building modules, stage 2.
make[1]: Leaving directory '/usr/src/kernels/5.2.8-200.fc30.i686'
  MODPOST 27 modules
BUILDSTDERR: ERROR: "__divdi3" [/builddir/build/BUILD/dahdi-linux-kmod-3.0.0/
_kmod_build_5.2.8-200.fc30.i686/drivers/dahdi/xpp/xpp_usb.ko] undefined!
BUILDSTDERR: ERROR: "__udivdi3" [/builddir/build/BUILD/dahdi-linux-kmod-3.0.0/
_kmod_build_5.2.8-200.fc30.i686/drivers/dahdi/xpp/xpp_usb.ko] undefined!
BUILDSTDERR: ERROR: "__moddi3" [/builddir/build/BUILD/dahdi-linux-kmod-3.0.0/
_kmod_build_5.2.8-200.fc30.i686/drivers/dahdi/xpp/xpp.ko] undefined!
BUILDSTDERR: ERROR: "__divdi3" [/builddir/build/BUILD/dahdi-linux-kmod-3.0.0/
_kmod_build_5.2.8-200.fc30.i686/drivers/dahdi/xpp/xpp.ko] undefined!

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Re: [asterisk-users] trouble building dahdi on kernel 5.2.7

[Anthony Joseph Messina]

On Wednesday, August 14, 2019 5:12:52 PM CDT sean darcy wrote:
> On 8/14/19 6:00 PM, sean darcy wrote:
> > dahdi built fine on 5.1.20, but on 5.2.7:
> > 
> > .
> > 
> >CC [M]
> > 
> > /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader
> > /dahdi_vpmadt032_loader.o> 
> >SHIPPED
> > 
> > /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader
> > /vpmadt032_x86_64.o> 
> >LD [M]
> > 
> > /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/dahdi_vpmadt032_
> > loader.o> 
> >Building modules, stage 2.
> >MODPOST 15 modules
> > 
> > ERROR: "vpmadtreg_register"
> > [/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/dahdi_vpmadt032
> > _loader.ko] undefined!
> > ERROR: "vpmadtreg_unregister"
> > [/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/dahdi_vpmadt032
> > _loader.ko] undefined!
> > make[2]: *** [scripts/Makefile.modpost:91: __modpost] Error 1
> > make[1]: *** [Makefile:1605: modules] Error 2
> > make[1]: Leaving directory '/usr/src/kernels/5.2.7-100.fc29.x86_64'
> > make: *** [Makefile:74: modules] Error 2
> > error: Bad exit status from /var/tmp/rpm-tmp.F8F4dL (%prep)
> > 
> > Any ideas ?
> > 
> > sean
> 
> And yes, kernel-devel is installled.
> 
> kernel-5.1.20-200.fc29.x86_64
> kernel-5.1.21-200.fc29.x86_64
> kernel-5.2.7-100.fc29.x86_64
> kernel-core-5.1.20-200.fc29.x86_64
> kernel-core-5.1.21-200.fc29.x86_64
> kernel-core-5.2.7-100.fc29.x86_64
> kernel-devel-5.1.20-200.fc29.x86_64
> kernel-devel-5.1.21-200.fc29.x86_64
> kernel-devel-5.2.7-100.fc29.x86_64
> kernel-headers-5.2.7-100.fc29.x86_64
> kernel-modules-5.1.20-200.fc29.x86_64
> kernel-modules-5.1.21-200.fc29.x86_64
> kernel-modules-5.2.7-100.fc29.x86_64
> kernel-tools-5.2.7-100.fc29.x86_64
> kernel-tools-libs-5.2.7-100.fc29.x86_64
> 
> The same kernel packages as the 5.1 kernels.
> 
> sean

Hi Sean.  Unfortunately I can only add a +1 for the DAHDI kernel modules, but 
can confirm that the SipWise rtpengine kernel module also fails to build.  I'm 
waiting to try on 5.2.8 to see if anything is different before raising the 
flag.

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Re: [asterisk-users] PJSIP wizard reload not reloading ?

[George Joseph]

On Tue, Jul 30, 2019 at 10:23 PM Jean-Denis Girard 
wrote:

> Le 25/07/2019 à 18:49, Jean-Denis Girard a écrit :
> > Hi list,
> >
> > I'm having a strange problem when using pjsip wizard and reloading
> > ("pjsip reload" on CLI): some data (specifically endpoint/pickup_group)
> > is not modified.
> Am I the only one experiencing this problem? Or nobody uses call_group /
> pickup_group on Asterisk-16?
>
>
Go ahead and open an issue for this at https://issues.asterisk.org.

We have an internal issue for something similar so I'll link them together.




>
> Thanks,
> --
> Jean-Denis Girard
>
> SysNux   Systèmes   Linux   en   Polynésie  française
> https://www.sysnux.pf/   Tél: +689 40.50.10.40 / GSM: +689 87.797.527
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Re: [asterisk-users] pjsip endoint woes

[George Joseph]

On Tue, Apr 9, 2019 at 9:28 AM sean darcy  wrote:

> On 4/8/19 6:18 AM, Joshua C. Colp wrote:
> > On Sat, Apr 6, 2019, at 10:04 AM, sean darcy wrote:
> >> On 4/5/19 10:36 AM, sean darcy wrote:
> >>> I'm trying to set up pjsip to work with an obi202 and google voice. But
> >>> I can't configure the endpoint.
> >>>
> >>> pjsip:
> >>>
> >>> [obi202-auth](!)
> >>> type = auth
> >>> auth_type = userpass
> >>> password = 
> >>>
> >>> [obi202-aor](!)
> >>> type = aor
> >>> max_contacts = 2
> >>>
> >>> ; = endpoints  
> >>>
> >>> [gv-voice](obi202-endpoint)
> >>> auth = gv-voice
> >>> aors = gv-voice
> >>> identify_by=auth_username
> >>> ;identify_by=username ; I also tried this. Same result.
> >>> context = gv-voice
> >>>
> >>> [gv-voice](obi202-auth)
> >>> username = gv-voice
> >>>
> >>> [gv-voice](obi202-aor)
> >>>
>




> >>> Any help appreciated.
> >>>
> >>> sean
> >>>
> >>>
> >>
> >> I'm expecting gv-voice to be the "matching endpoint". The INVITE has
> >> gv-voice as the "Contact:" . Isn't this the "Username" in pjsip "auth" ?
> >
> > Nope. The Contact is never considered for that. The From username is
> what is matched for an endpoint using the "username" option. The
> authentication username is what is matched for an endpoint using the
> "auth_username" option but you also need to ensure it is enabled in
> "endpoint_identifier_order" global option.
>
> Thanks for the reply.
>
> auth_username seems to be enabled:
>
> asterisk*CLI> pjsip show identifiers
> Identifier Names:
> name not specified
> ip
> username
> anonymous
> header
> auth_username
>
> Is the order a problem ?
>
> I set:
>
> endpoint_identifier_order=auth_username,"name not
> specified",ip,username,anonymous,header
>
> restarted. No errors.
>
> But no effect on the identifier order.
>
>  From obi202 :
>
> SIP Credentials
> Parameter Name  Value
> AuthUserNamegv-voice
> AuthPasswordpassword
>
> >This also requires the endpoint to actually authenticate.
>
> Not sure what this means. Of course, I agree, but how do I make this
> happen?
>
> BTW, this is 16.3.0.
>
> sean
>

So you're using the obi with 2 itsp accounts, one Google Voice and the
other Asterisk yeah?  Then forwarding calls between them?
Are the credentials defined in the gv-voice auth object also configured in
the obi account that points to Asterisk?  Is the obi registering
successfully to Asterisk?

In this situation it might be better to treat the OBi like an ITSP in
Asterisk.  Instead of using registration and userid/password
authentication, add an "identify" object that matches on the obi's ip
address and points to the gv-voice endpoint...

[gv-voice]
type = identify
match = /
endpoint = gv-voice

This way you don't need to set up any auth objects at all and you just need
to set "identify = ip" on the endpoint.



>
>
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Re: [asterisk-users] Asterisk Dahdi Issue

[George Joseph]

On Fri, Feb 8, 2019 at 3:55 PM Alexander Perkins <
alexanderhenryperk...@gmail.com> wrote:

> Hi All.  I am trying to install dahdi, but I get the error below.  I have
> installed kernal devel from yum and also rebooted.  I've googled, but
> cannot seem to find an answer.  Any help would be appreciated.
>

Dahdi 3.0.0 requires a 4.0 linux kernel.  If you're building with CentOS
you'll need to use an earlier version of Dahdi.


> Thanks,
> Alex
>
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Re: [asterisk-users] INVITE from DID: No matching endpoint found but completes the call anyway

[George Joseph]

On Sat, Jan 26, 2019 at 10:56 AM Brian J. Murrell 
wrote:

> I have a trunk set up for the DID from my provider:
>
> [my_provider]
> type=registration
> outbound_auth=my_provider
> server_uri=sip:sip.example.com
> client_uri=sip:my_usern...@sip.example.com
> retry_interval=60
>
> [my_provider]
> type=auth
> auth_type=userpass
> password=123456
> username=my_username
>
> [my_provider]
> type=aor
> contact=sip:sip.example.com:5060
>
> [my_provider]
> type=endpoint
> context=from-my_provider
> disallow=all
> allow=ulaw
> outbound_auth=my_provider
> aors=my_provider
>
> [my_provider]
> type=identify
> endpoint=my_provider
> match=sip.example.com
>
> And it registers fine:
>
>
> 
>
> ==
>
>  mytrunk/sip:sip.example.com my_provider
>  Registered
>
>
> And when it gets an INVITE from my provider (192.168.0.1):
>
> <--- Received SIP request (997 bytes) from UDP:192.168.0.1:5060 --->
> INVITE sip:1235551212@10.75.22.5:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK06d035fd;rport
> Max-Forwards: 70
> From: "Fred Flintstone" ;tag=as539f9476
> To: 
> Contact: 
> Call-ID: 3ef877dc4477d8ce4aae29965c5d0875@192.168.0.1:5060
> CSeq: 102 INVITE
> User-Agent: foobar
> Date: Sat, 26 Jan 2019 17:40:00 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH, MESSAGE
> Supported: replaces, timer
> Remote-Party-ID: "Fred Flintstone"  >;party=calling;privacy=off;screen=no
> Content-Type: application/sdp
> Content-Length: 295
>
> [SDP redacted]
>
> It logs an error:
>
> [Jan 26 12:40:00] NOTICE[21775]: res_pjsip/pjsip_distributor.c:525
> log_failed_request: Request 'INVITE' from '"Fred Flintstone" <
> sip:4565551212@192.168.0.1>' failed for '192.168.0.1:5060' (callid:
> 3ef877dc4477d8ce4aae29965c5d0875@192.168.0.1:5060) - No matching endpoint
> found
>
> But then goes on to complete the call:
>
> <--- Transmitting SIP response (352 bytes) to UDP:192.168.0.1:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.0.1:5060
> ;rport=5060;received=192.168.0.1;branch=z9hG4bK06d035fd
> Call-ID: 3ef877dc4477d8ce4aae29965c5d0875@192.168.0.1:5060
> From: "Fred Flintstone" ;tag=as539f9476
> To: 
> CSeq: 102 INVITE
> Server: Asterisk PBX 13.11.1
> Content-Length:  0
>
> [ launch into dialplan ]
>
> So why the spurious error when it was able to complete the call?
>

What version of Asterisk and what's the value of the "identify_by"
parameter for the endpoint?
When you have an "identify" object configured, you should just use "ip" as
the "identify_by",





>
> Cheers,
> b.
>
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Re: [asterisk-users] [asterisk-app-dev] Who uses the ari/sounds resource?

[George Joseph]

On Mon, Jan 21, 2019 at 2:45 PM sdut...@wazo.io  wrote:

> On 2019-01-21 10:42 a.m., George Joseph wrote:
> > and what do you use it for?
>
> I assume that you're talking about the /sounds resource available in ARI
> [1].
>
> At Wazo [2], we use the /sounds resource to list the sound files that
> are register in Asterisk.
>
> It allows us to implement a REST API that gives an overview of what
> sound files are available and in what languages.
>
> This REST API is then used to display the list of sound files in a web
> page, or to select a sound file in an application, e.g. when creating an
> IVR and selecting a sound file to play. The REST API is also used to
> list the available languages for sound files.
>
> We have used the /sounds resource when we could, but we still had to
> access the file system directly at some point in order to offer more
> features in our REST API, like uploading custom sounds, streaming the
> sound files or accessing recording files.
>
> [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Sounds+REST+API
> [2] http://wazo.community/
>
> --
> Sébastien Duthil
> Wazo developer
>

Thanks Sébastien.

What kind of response time are you getting when you retrieve the full list?

The reason I'm asking is that when Asterisk starts, we pre-populate an
internal cache (media_index) with all the sounds which, depending on which
languages and formats are installed, could easily be 20,000 files.  In some
instances, users are creating symlinks inside the sounds language
directories to other places (like voicemail) which now will also get
indexed.  In one case, we had a customer with an effective list of over
800,000 sound files.  This takes quite a bit of memory.  What we're trying
to determine is the value of the cache considering that we're only caching
directory entries (not contents) which the filesystem has already got
cached.  In tests of the ari/sounds resource, 95% of the response time is
constructing the JSON response, even when not using the cache.  I.E.  The
cache made no appreciable difference in the response time.









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Re: [asterisk-users] [asterisk-app-dev] [asterisk-dev] Who uses the ari/sounds resource?

[George Joseph]

On Mon, Jan 21, 2019 at 11:55 AM Andrew Latham  wrote:

> Are you discussing
> https://wiki.asterisk.org/wiki/display/AST/ARI+and+Channels%3A+Simple+Media+Manipulation#ARIandChannels:SimpleMediaManipulation-Example:Playingbackasoundfile
> or something else?
>

I'm talking about the "sounds" resource itself.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Sounds+REST+API



> On Mon, Jan 21, 2019 at 9:42 AM George Joseph  wrote:
>
>> and what do you use it for?
>>
>>
>> --
>> *George Joseph*
>> Digium - A Sangoma Company | Software Developer | Software Engineering
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>> direct/fax: +1 256 428 6012
>> Check us out at: https://digium.com · https://sangoma.com
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>
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[asterisk-users] [asterisk-app-dev] Who uses the ari/sounds resource?

[George Joseph]

and what do you use it for?


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Re: [asterisk-users] Asterisk 15.6.1. Symbol pjsip_tls_transport_start2 not found

[George Joseph]

On Tue, Sep 25, 2018 at 2:18 PM Dmitriy Serov  wrote:

> Hello.
>
> After successful compilation 15.6.1 (bundled pjsip) and start asterisk i
> has error Symbol pjsip_tls_transport_start2 not found.
>
> /main/libasteriskpj.exports does not containg pjsip_tls_transport_start2
> and pjsip_tls_transport_start.
>
> More:
>
>- All versions before (including 15.5) has not such error on this
>computer (ubuntu 18.04).
>- with 15.6.0, 15.6.1 has error on this computer
>
>
I just tried building and running 15.6 on 18.04 and didn't have an issue.
Can  you double check that you don't have any libpj* files anywhere in
/usr/lib or /usr/local/lib?
You can also run "aptitude search pjsip pjproject" to make sure the
packages aren't installed.


>
>-
>- Other server (ubuntu 16.04) has not this error with 15.6.0
>
> What's wrong? Where to start searching for reasons?
>
> Thanks
> --
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Re: [asterisk-users] Asterisk crashing on AAAA lookup

[George Joseph]

On Wed, Jun 27, 2018 at 5:15 AM Dovid Bender  wrote:

>
>
> On Tue, Jun 26, 2018 at 7:59 PM, Richard Mudgett 
> wrote:
>
>>
>>
>> On Tue, Jun 26, 2018 at 6:15 PM, Dovid Bender 
>> wrote:
>>
>>> I have Asterisk running on a Ubuntu 18.0.4 on Digital Ocean. Every so
>>> often asterisk crashes and then restarts. I am not seeing any core dumps on
>>> the box. The only I thing I see every time is a second before Asterisk
>>> crashes there is a  lookup for the boxes hostname. As soon as it gets
>>> the response I see that asterisk is restarting. Any idea what would cause
>>> this and how would get a dump or further debug? I did build Asterisk with
>>> DONT_OPTIMIZE and BETTER_BACKTRACES but not seeing any traces anywhere. I
>>> am using Asterisk 15.4.1.
>>>
>>
>> You have to start asterisk with the -g option to make asterisk create
>> core files.
>> https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
>>
>> Richard
>>
>> It's very strange. So when I try to start asterisk via systemd I get:
> root@fingerprint1:/var/lib/lxcfs/cgroup/name=systemd/system.slice#
> systemctl start asterisk
> Job for asterisk.service failed because a timeout was exceeded.
> See "systemctl status asterisk.service" and "journalctl -xe" for details.
> root@fingerprint1:/var/lib/lxcfs/cgroup/name=systemd/system.slice#
> root@fingerprint1:/var/lib/lxcfs/cgroup/name=systemd/system.slice#
> root@fingerprint1:/var/lib/lxcfs/cgroup/name=systemd/system.slice# ps aux
> | grep aster
> root 14412  0.1  0.5  25084  5208 pts/2S+   10:52   0:00 nano
> /lib/systemd/system/asterisk.service
> asterisk 14425  9.3  4.7 1304352 48144 ?   Ssl  10:52   0:00
> /usr/sbin/asterisk -g -f -U asterisk
> root 14526  0.0  0.0  14856   976 pts/0S+   10:52   0:00 grep
> --color=auto aster
> root@fingerprint1:/var/lib/lxcfs/cgroup/name=systemd/system.slice#
>
> As you can see it's still working. If I then connect to the console
> asterisk is running fine, in this case Asterisk restarts randomly (every
> 1-2 minutes). If I then start asterisk myself by doing:
>  /usr/sbin/asterisk -g -f -U asterisk
>
> Then it starts fine and works with no issue. It would seem there is
> something with systemd that is causing Asterisk to restart. I don't think
> it's the actual script since I would then expect it to always restart at
> the same time though I am not able to find any dumps any where on the box.
>
> Any ideas?
>

Look in /var/lib/asterisk.  That's the home directory that the service file
sets.  Core files may be there.
Run "sysctl kernel.core_pattern".  That will tell you where the kernel will
place the files.

Did you previously install asterisk from apt and did you uninstall it
before compiling and installing 15.4.1?  The default version from apt is
13.18.3 so you may have a mixed installation that's causing issues.

Run "dmesg" after asterisk dies.  If it actually crashed, there'll be a
message in the kernel log.



>
>
> --
> _
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>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

[George Joseph]

On Wed, Jun 6, 2018 at 1:51 AM Olivier  wrote:

>
>
> 2018-06-05 20:29 GMT+02:00 George Joseph :
>
>>
>>
>> On Tue, Jun 5, 2018 at 10:59 AM Olivier  wrote:
>>
>>>
>>>
>>> 2018-06-05 15:27 GMT+02:00 George Joseph :
>>> Thank  you very much, George for replying.
>>>
>>>>
>>>>
>>>> On Tue, Jun 5, 2018 at 3:35 AM Olivier  wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> After a long discussion with a friend, I would like to ask here:
>>>>>
>>>>> 1.According SIP RFCs, is possible/recommended to have different values
>>>>> in From and P-Asserted-Id fields ?
>>>>> For instance, From field showing 123456789 and P-Asserted-Id showing
>>>>> 987654321 (beside privacy considerations) ?
>>>>>
>>>>
>>>> Possible? yes absolutely.
>>>>
>>>
>>> How would you then configure both headers, respectively ?
>>>
>>> From memory, in previous testings, whenever  CALLERID was set to
>>> WHATEVER, P-Asserted-Id was also set to WHATEVER and vice versa, so that I
>>> inferred from this that P-Asserted-Id was meant for Privacy
>>> considerations and nothing else (see [1])
>>>
>>
>> PAI should be used to indicate the calling party's identification
>> regardless of privacy concerns.  In the dialplan you can use the CALLERPRES
>> function to control privacy on a call by call basis.
>>
>>
>>
> I'm sorry but I still have a doubt ...
> Let me re-phrase my question:
>
> My setup is:
> Asterisk <--- PJSIP ---> Bob
>
> For a reason, I want Bob's phone to receive a call with the following
> headers:
>
> From: "Foo" ;tag=as75ee8c7c
> P-Asserted-Id: "Foo" >;whatever
>
> My dialplan is:
> same = n,Set(CALLERID(num)=999)
> XXX
> same = n,Dial(PJSIP/123456@bob)
>

> What shall I replace XXX with to allow me to set 8 in the user part of
> P-Asserted-Id URI (see example above) ?
>

You don't need anything for XXX.  Just the "Set(CALLERID(num)=999)" should
do it.


> CALLERPRES would change From or P-Asserted-Id but not having different
> user parts in URI, would it ?
>

CALLERPRES affects From and Contact but doesn't affect PAI at all.  It does
add a Privacy header though.
BTW, you should use CALLERID(pres) instead of CALLERPRES.


>
> To my knowledge, a possible way to implement what I'm after is to "turn
> off" P-Asserted-Id feature, add a custom P-Asserted-Id header with
> PJSIP_HEADER.
> Am I missing something ?
>

Either that or I am. :)   The example you have above should work.



>
>
>
>>
>>
>>>
>>>
>>> [1]
>>> https://www.voip-info.org/p-asserted-identity-and-remote-party-id-header/
>>>
>>>
>>>> Recommended? who knows?  Implementations are all over the place.  I've
>>>> always thought of the From header as identifying the user agent making the
>>>> request which kinda agrees with RFC3261.   The PAI header should contain
>>>> the identity of the original caller.
>>>>
>>>>
>>>>>
>>>>> 2. When Bob forwards to Cory a call coming from Alice, would expect
>>>>> Diversion/History-Info header to include Alice's number ?
>>>>>
>>>>
>>>> No.  The diversion header shows who the diverter is.
>>>> https://tools.ietf.org/html/rfc5806
>>>>
>>>
>
> Thank for  this reference: I think I confused diverter/caller/callee roles
> when I first read this document.
>
> So, if Bob forwards to Cory a call from Alice, in which headers would you
> expect Alice and Bob numbers to respectively appear ?
>
>
Well, if you just have 3 user agents without asterisk in the middle
Alice sends INVITE to Bob.
Bob returns a 302 to Alice with Cory as the Contact and Bob as the Diversion
Alice sends an ACK to Bob.
Alice sends a new INVITE to Cory.

If Asterisk is in the middle then...
Alice sends INVITE to Bob via Asterisk
Asterisk sends INVITE to Bob with Alice in From/PAI
Bob returns a 302 to Asterisk with Cory in Contact and Bob in Diversion
Asterisk returns a 181 "Call is being forwarded" to Alice.
Asterisk goes back to the dialplan to find Cory.
Asterisk sends an INVITE to Cory with Alice in From/PAI and Bob in
Diversion.
When Cory answers, Asterisk sends back a 200 OK to Alice with Cory in PAI
and Bob in Diversion


>
>
>
>>
>> Best regards
>> --
>> __

Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

[George Joseph]

On Tue, Jun 5, 2018 at 10:59 AM Olivier  wrote:

>
>
> 2018-06-05 15:27 GMT+02:00 George Joseph :
> Thank  you very much, George for replying.
>
>>
>>
>> On Tue, Jun 5, 2018 at 3:35 AM Olivier  wrote:
>>
>>> Hi,
>>>
>>> After a long discussion with a friend, I would like to ask here:
>>>
>>> 1.According SIP RFCs, is possible/recommended to have different values
>>> in From and P-Asserted-Id fields ?
>>> For instance, From field showing 123456789 and P-Asserted-Id showing
>>> 987654321 (beside privacy considerations) ?
>>>
>>
>> Possible? yes absolutely.
>>
>
> How would you then configure both headers, respectively ?
>
> From memory, in previous testings, whenever  CALLERID was set to WHATEVER,
> P-Asserted-Id was also set to WHATEVER and vice versa, so that I inferred
> from this that P-Asserted-Id was meant for Privacy considerations and
> nothing else (see [1])
>

PAI should be used to indicate the calling party's identification
regardless of privacy concerns.  In the dialplan you can use the CALLERPRES
function to control privacy on a call by call basis.




>
>
> [1]
> https://www.voip-info.org/p-asserted-identity-and-remote-party-id-header/
>
>
>> Recommended? who knows?  Implementations are all over the place.  I've
>> always thought of the From header as identifying the user agent making the
>> request which kinda agrees with RFC3261.   The PAI header should contain
>> the identity of the original caller.
>>
>>
>>>
>>> 2. When Bob forwards to Cory a call coming from Alice, would expect
>>> Diversion/History-Info header to include Alice's number ?
>>>
>>
>> No.  The diversion header shows who the diverter is.
>> https://tools.ietf.org/html/rfc5806
>>
>>
>>
>>
>>>
>>> Best regards
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>> --
>> George Joseph
>> Digium, Inc. | Software Developer
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>> Check us out at: www.digium.com & www.asterisk.org
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

[George Joseph]

On Tue, Jun 5, 2018 at 3:35 AM Olivier  wrote:

> Hi,
>
> After a long discussion with a friend, I would like to ask here:
>
> 1.According SIP RFCs, is possible/recommended to have different values in
> From and P-Asserted-Id fields ?
> For instance, From field showing 123456789 and P-Asserted-Id showing
> 987654321 (beside privacy considerations) ?
>

Possible? yes absolutely.  Recommended? who knows?  Implementations are all
over the place.  I've always thought of the From header as identifying the
user agent making the request which kinda agrees with RFC3261.   The PAI
header should contain the identity of the original caller.


>
> 2. When Bob forwards to Cory a call coming from Alice, would expect
> Diversion/History-Info header to include Alice's number ?
>

No.  The diversion header shows who the diverter is.
https://tools.ietf.org/html/rfc5806




>
> Best regards
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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_
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Re: [asterisk-users] Possibility to access PJSIP variables from dialplan

[George Joseph]

On Tue, Apr 17, 2018 at 7:20 AM, Administrator TOOTAI 
wrote:

> Hi all,
>
> is it possible to access PJSIP configuration variables from the dialplan ?
> Exemple: I want to get the username of a type = auth context.
>

The AST_CONFIG dialplan function will let you pull things form any standard
asterisk config file but it's not optimized for reading files where the
same "category" occurs more than once and it doesn't understand "type".  So
if you have both an aor and an auth named "1000" you might have an issue.




>
> Thanks for any hint
>
> Daniel
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
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>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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Re: [asterisk-users] Asterisk 15.3.0-rc1 regression

[George Joseph]

On Mon, Feb 26, 2018 at 8:22 AM, Dmitriy Serov  wrote:

> Asterisk 15.2.0, pjsip
>
> Yesterday I installed 15.3.0-rc1 (after 15.2.0). Today I had to rollback
> to 15.2.0 (not 15.2.2).
>
> The reason is: the loaded server very often hung on the same places:
> SUBSCRIBE/NOTIFY processing. Unloading modules was the solution at that
> time.
>

Please open an issue ASAP!



>
> ; Off SUBSCRIBE + NOTIFY
>
> noload => res_pjsip_pubsub.so
> noload => res_pjsip_exten_state.so
> noload => res_pjsip_dialog_info_body_generator.so
> noload => res_pjsip_mwi_body_generator.so
> noload => res_pjsip_pidf_body_generator.so
> noload => res_pjsip_xpidf_body_generator.so
>
> At 15.3.0 asterisk stopped processing authorization, because it started to
> issue an error on the SUBSCRIBE packet. More precisely: the server believed
> that the client is authorized, and the client (PhonerLite) believed that
> authorization failed.
>
> Try to load the modules   res_pjsip_pubsub  and res_pjsip_exten_state has
> led to renewed hangs.
>
> Rollback to 15.2.2 did not bring success. Registering without
> res_pjsip_pubsub led to the client thinking that it was authorized, and the
> server did not think so.
>
> Whether the res_pjsip_pubsub module is mandatory for successful work?
>
>
> P.S. ASTERISK-27474. Bug not fixed in 15.3.0-rc1. This is very sad,
> because encryption should be mandatory in the modern world.
>
>
> Dmitriy Serov.
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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Re: [asterisk-users] [OT] How to use audio files with SIPp

[George Joseph]

On Fri, Feb 9, 2018 at 8:04 AM, Olivier  wrote:

> Thank you very much George for replying.
>
> 2018-02-09 14:39 GMT+01:00 George Joseph :
>
>>
>>
>> On Fri, Feb 9, 2018 at 6:27 AM, Olivier  wrote:
>>
>>> Hello,
>>>
>>> SIPp's PCAP play feature can replay pre-recorded audio stream towards
>>> destination (see [1]).
>>> Doc mentions tcpdump and Wireshark as tools to record such RTP streams
>>> without further details.
>>>
>>>
>>> Looking at SIPp 3.2 source archive, I found PCAP samples in a pcap/
>>> directory.
>>> Sample pcap/g711a.pcap file includes RTP from 10.1.3.1:5000 to
>>> 10.1.6.18:2006
>>>
>>> 1. How can you "forge" IPs and/or ports of a pcap file ?
>>>
>>
>> You don't have to.  sipp only takes the rtp payload from the packets in
>> the pcap then just sends the datagrams to the remote in the scenario.
>>
>
> That is exactly what I'm after !
>
> Before diving into this, can I ask which SIPp version and feature are we
> talking about here ?
>

We're on 3.5.


>
> Since I posted my question, I've read this [1] thread mentionning a new
> WAV file playing capability but this feature required SIPp 3.4 and above.
> On Debian Stetch I'm playing with, packaged SIPp is 3.2.
>

That's pretty old.  I'd recommend compiling from source yourself.  It's
very easy to build.


>
>
> [1] https://stackoverflow.com/questions/20122607/playing-
> audio-file-using-sipp/20123193
>
>
>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>>
>>
>>
>> --
>> George Joseph
>> Digium, Inc. | Software Developer
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>> Check us out at: www.digium.com & www.asterisk.org
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
-- 
_
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Re: [asterisk-users] [OT] How to use audio files with SIPp

[George Joseph]

On Fri, Feb 9, 2018 at 6:27 AM, Olivier  wrote:

> Hello,
>
> SIPp's PCAP play feature can replay pre-recorded audio stream towards
> destination (see [1]).
> Doc mentions tcpdump and Wireshark as tools to record such RTP streams
> without further details.
>
>
> Looking at SIPp 3.2 source archive, I found PCAP samples in a pcap/
> directory.
> Sample pcap/g711a.pcap file includes RTP from 10.1.3.1:5000 to
> 10.1.6.18:2006
>
> 1. How can you "forge" IPs and/or ports of a pcap file ?
>

You don't have to.  sipp only takes the rtp payload from the packets in the
pcap then just sends the datagrams to the remote in the scenario.


>
> 2. When generating simultaneous calls from one source device to a single
> target device, do you need to have specific PCAP files (one specific for
> each call) with specific source port ?
>

Nope.  See above.


>
> 3. How do you capture an RTP flux with thark or tcpdump ?
>

This is a little tricky.  tcpdump isn't much help if you don't know the
ports.  With tshark though you can create a "read" filter that can capture
only RTP packets but it's very expensive.  Usually, I just capture all UDP
packets between the hosts in question with tcpdump, then use the Wireshark
gui to filter the rtp stream.  Then you can just export those packets.



>
> Best regards
>
> [1] http://sipp.sourceforge.net/doc/reference.html#PCAP+Play
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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Re: [asterisk-users] pjsip trunking configuration issue

[George Joseph]

On Thu, Feb 8, 2018 at 12:53 AM, Kevin Long 
wrote:

>
>
> Greetings !
>
>
> My goal is to get Twilio trunking working, and with TLS/SRTP.
>
> I see this concerning message in my log:
>
> [Feb  7 16:50:26] ERROR[20596] res_sorcery_config.c: Could not create an
> object of type 'endpoint' with id ’twilio' from configuration file
> ‘pjsip.conf’
>
>
>
> Thus, ‘pjsip show endpoints’  does not show the endpoint for the Twilio
> trunk.
>
>
> Hoping for a sanity check of my pjsip.conf file, and what could be causing
> this.
>
> A test call form Twilio’s system hits the PBX (over TLS), but always says
> “No matching endpoint found” in the asterisk log.
>
>
>
> pjsip.conf
>
> [transport-tls]
> type = transport
> protocol = tls
> bind = 0.0.0.0:5061
> cert_file=cert_file
> priv_key_file=key_file
> method=tlsv1
> external_media_address=X.Y.Z.D
> external_signaling_address=X.Y.Z.D
> verify_client=no
> verify_server=no
> allow_reload=yes
>
> [twilio](!)
> type=endpoint
> transport=transport-tls
> context=from-twilio
> disallow=all
> allow=ulaw
> dtmf_mode=inband
>

Are you sure you want "inband" and not "rfc4733"?



> media_encryption=sdes
> rtp_symmetric=yes
> rewrite_contact=yes
> force_rport=yes
> canreinvite=no
> tlsdontverifyserver=yes
>

"canreinvite" and "tlsdontverifyserver" aren't valid endpoint parameters
which is why the endpoint is failing to load.


>
>
> [auth-out](!)
> type=auth
> auth_type=userpass
>
> [twilio]
> aors=twilio-aors
>
> [twilio-aors]
> type=aor
> contact=sips:trunkname.pstn.twilio.com:5061 ;tried with sip: also
>
> [twilio]
> type=identify
> endpoint=twilio
> match=54.172.60.0
> match=54.172.60.1
> match=54.172.60.2
> match=54.172.60.3
>
> [endpoint-basic](!)
> type=endpoint
> transport=transport-tls
> context=from-phones
> disallow=all
> allow=ulaw
>
> [auth-userpass](!)
> type=auth
> auth_type=userpass
>
> [aor-single-reg](!)
> type=aor
> max_contacts=20
>
> [1001](endpoint-basic)
> auth=auth1001
> aors=1001
>
> [auth1001](auth-userpass)
> password=password123
> username=1001
>
> [1001](aor-single-reg)
>
>
> Extensions.conf
>
> [from-twilio]
> exten => _+1NX,1,Dial(PJSIP/1001)
>
> [from-phones]
> exten => _NXXNXX,1,Set(CALLERID(all)="David" <78451234>)
> same => n,Dial(PJSIP/+1${EXTEN}@twilio)
> --
> _____
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] What is the status of world wide e164 DUNDI

[Anthony Joseph Messina]

On Friday, February 2, 2018 3:15:22 AM CST Benoit Panizzon wrote:
> Hello List
> 
> I have a still two connected DUNDI peers, but they seem to flap from
> time to time.
> 
> A couple of years ago I was able to look up quite some, mostly free
> call numbers via DUNDI all over the world and I als saw incomming
> lookups.
> 
> But not anymore. I wonder if I am stranded on a no longer world-wide
> connected DUNDI island of me and the two remaining peers I have.
> 
> http://www.dundi.com/ only shows a default website.
> 
> My last request for peers on the DUNDI Mailinglist from March 2017 was
> unanswered.
> 
> Is anybody still interconnected via DUNDI or has this service silently
> died?
> 
> Mit freundlichen Grüssen
> 
> -Benoît Panizzon-

I'm in the US where things seemed to die off dramatically some years back:
https://messinet.com/post/voip/2013/09/10/leaving-the-dundi-e.164-network/

-- 
Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery
F9B6 560E 68EA 037D 8C3D  D1C9 FF31 3BDB D9D8 99B6


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Re: [asterisk-users] res_pjsip_transport_management.c: Shutting down transport

[George Joseph]

On Wed, Jan 24, 2018 at 7:07 AM, marek cervenka  wrote:

> hello,
>
> i met with this interesting situation
>
> [Jan 24 13:48:37] NOTICE[1049] res_pjsip_transport_management.c: Shutting
> down transport '8' since no request was received in 32 seconds
>
> [Jan 24 13:48:37] NOTICE[1049] res_pjsip_transport_management.c: Shutting
> down transport '8' since no request was received in 32 seconds
> [Jan 24 13:48:41] NOTICE[1049] res_pjsip_transport_management.c: Shutting
> down transport '' since no request was received in 32 seconds
> [Jan 24 13:48:44] NOTICE[1049] res_pjsip_transport_management.c: Shutting
> down transport 'e=" 79-ad2e-c47e6a3db178>";expires=60
> u▒l^' since no request was received in 32 seconds
> [Jan 24 13:48:44] NOTICE[1049] res_pjsip_transport_management.c: Shutting
> down transport '' since no request was received in 32 seconds
> [Jan 24 13:48:45] NOTICE[1049] res_pjsip_transport_management.c: Shutting
> down transport '' since no request was received in 32 seconds
> [Jan 24 13:48:47] NOTICE[1049] res_pjsip_transport_management.c: Shutting
> down transport 'e=" 37-966d-9a936a350728>";expires=60
> ' since no request was received in 32 seconds
> [Jan 24 13:48:49] NOTICE[1049] res_pjsip_transport_management.c: Shutting
> down transport '' since no request was received in 32 seconds
> [Jan 24 13:48:50] NOTICE[1049] res_pjsip_transport_management.c: Shutting
> down transport '.0
> Date: Wed, 24 Jan 2018 12:48:18 GMT
> Allow: INVITE, ACK, CAN' since no request was received in 32 seconds
> [Jan 24 13:48:53] NOTICE[1049] res_pjsip_transport_management.c: Shutting
> down transport '' since no request was received in 32 seconds
> [Jan 24 13:48:53] NOTICE[1049] res_pjsip_transport_management.c: Shutting
> down transport ' SUBSCRIBE, INFO' since no request was received in 32
> seconds
> [Jan 24 13:48:53] NOTICE[1049] res_pjsip_transport_management.c: Shutting
> down transport 'c732305e-f905-489a-a6f4-5164f0809c8a>";expires=60
> Expires: 60
> @u▒^' since no request was received in 32 seconds
> [Jan 24 13:49:27] NOTICE[1049] res_pjsip_transport_management.c: Shutting
> down transport '▒▒<%▒▒*W▒▒▒$@▒▒▒&r{▒X_DL▒▒▒1▒▒"▒`$▒zC▒l▒o▒O▒3▒▒c:133
> idle_sched_cb: Shutting down transport '=" b-b2ca-6292f151c7c2>";expires=60
>
> asterisk went crazy and had to be restarted
>


That module does 2 things.  First it handles the keepalives
if keep_alive_interval is > 0 in the pjsip.conf/global.  It also attempts
to mitigate DOS attacks if an attacker floods asterisk with TCP (or TLS)
connections but doesn't send any actual messages within the time set in
pjsip.conf/system/timer_b.   When a connection is opened, a timer is
started and if there is no recognizable SIP message before the timer
expires, you get the "Shutting down transport" message.


>
> topology
>
> asterisk 13.18.2 + pjsip realtime  + mariadb  (mariadb is on different
> network!) + jssip via wss as client
>
> extconfig.conf
>
> ps_endpoints => odbc,configDb
> ps_auths => odbc,configDb
> ps_aors => odbc,configDb
> ps_domain_aliases => odbc,configDb
>
> sorcery.conf
>
> [res_pjsip] ; Realtime PJSIP configuration wizard
> endpoint/cache=memory_cache,expire_on_reload=yes,full_backend_cache=yes
> endpoint=realtime,ps_endpoints
> auth/cache=memory_cache,expire_on_reload=yes,full_backend_cache=yes
> auth=realtime,ps_auths
> aor/cache=memory_cache,expire_on_reload=yes,full_backend_cache=yes
> aor=realtime,ps_aors
> domain_alias=realtime,ps_domain_aliases
>
>
> there was net interruption on ~13:48
>
> do you have any ideas what can be cause of "res_pjsip_transport_management.c:
> Shutting down transport" ?
>

Yep, it was probably that network interruption.  The incoming messages were
being corrupted and not recognized as real SIP messages so the timer
expired and the transports were shut down.


>
> my idea was that Asterisk with cache doesnt need realtime connectivity
> with mariadb (can survive short internet interruptions)
>
> Marek
>
>
>
>
> --
> _____
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
George Joseph
Digium, Inc. | Softw

Re: [asterisk-users] PJSIP: identify endpoint by authentication username?

[George Joseph]

On Tue, Jan 9, 2018 at 5:38 AM, Benoit Panizzon 
wrote:

> Dear fellow list readers
>
> This is the situation:
>
> ISDN Devices => Patton ISDN to SIP GW => Asterisk PJSIP
>
> The Patton GW resides on a dynamic IP address, so I cannot really use
> match=ip in the identify section.
>
> The Patton does not send a line parameter.
>
> The ISDN Devices behind the patton have different MSN and should be
> able to send them in the From: Header, so the default endpoint
> identification mechanism which matches the From username with the
> endplaint fails.
>
> So what are the options to solve that issue?
>
> I see the asterisk sending out a challenge and getting a proper reply
> from the patton, but then stills complains about the endpoint not
> matching.
>
> According to the manual there is no
>
> type=identify
> match=authentication_username
>

There is no need for a separate "identify" object in this case.  In the
pjsip.conf "global" section set "endpoint_identifier_order" to include
"auth_username" and in each endpoint's section set "identify_by" to include
"auth_username".

[global]
endpoint_identifier_order = auth_username,username,ip,anonymous

[endpoint_x]
identify_by = auth_username





>
> or similar.
>
> Mit freundlichen Grüssen
>
> -Benoît Panizzon-
> --
> I m p r o W a r e   A G-Leiter Commerce Kunden
> __
>
> Zurlindenstrasse 29 Tel  +41 61 826 93 00
> CH-4133 PrattelnFax  +41 61 826 93 01
> Schweiz Web  http://www.imp.ch
> __
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system?

[George Joseph]

On Thu, Jan 4, 2018 at 11:07 AM, Dan Cropp  wrote:

> Thank you George.
>
>
>
> I will pass along the rfc information to those responsible for the other
> switch.
>
>
>
> I missed the match_header addition to Asterisk.
>
> Unfortunately, the only header field that seems appropriate is the To
> header.
>
>
>
> On a separate box I am now trying to configure the endpoint recognition.
> Planning on multiple endpoints to the same switch, so I am trying to use
> the match_header field.
>
>
>
> I tried programming the match_header with the To: header information.
> Unfortunately, it didn’t work.  Apparently the To header doesn’t work with
> the match_header field.
>
> The Asterisk debug shows the following…
>
>
>
> DEBUG[2778] res_pjsip_endpoint_identifier_ip.c: SIP message contains
> header 'To' but value '' does not match value ''
> for endpoint '286'
>

Rats.  Apparently the code assumes the header being searched for is a
"generic string" header but the To header is its own non-generic type.

I created an issue for that...
https://issues.asterisk.org/jira/browse/ASTERISK-27548



>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] *On Behalf Of *George Joseph
> *Sent:* Tuesday, December 19, 2017 7:57 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Is it possible to have two endpoints to
> the same IP address where one uses IP based authentication and the other
> requires asterisk to register to that system?
>
>
>
>
>
>
>
> On Mon, Dec 18, 2017 at 9:04 AM, Dan Cropp  wrote:
>
> Thanks George
>
>
>
> I originally didn’t have the 1002@ for the identify.  Changed that when
> things were not working.  I changed it back.
>
>
>
> Unfortunately, the system I am connecting with doesn’t seem to support the
> line support.  Looking at the SIP packets, I see Asterisk send it.
> Unfortunately, they do not send the line information as part of the
> INVITE.  I checked with some developers of that system and they do not know
> anything about the line setting.
>
> Is there any rfcs I could refer them to?
>
>
>
> Yeah, I've found that some providers do and some providers don't.
>
>
>
>
>
> https://tools.ietf.org/html/rfc3261#section-19.1
>
> An implementation MUST include any provided transport, maddr, ttl, or
> user parameter in the Request-URI of the formed request. If the URI
> contains a method parameter, its value MUST be used as the method of
> the request. The method parameter MUST NOT be placed in the
> Request-URI.
>
> *​​*
>
>
> *Unknown URI parameters MUST be placed in the message'sRequest-URI*.
>
>
>
> The identify object also has the capability to match against a specific
> header and value but it looks like it only tries to match on header if it
> can't find a match by ip address.  Here's some info on it anyway.
>
>
>
> If you're provider puts something unique and constant in the headers, like
> a User-Agent string that doesn't change, you can also try using the
> "match_header" parameter to an identify object.
>
>
>
> [my_provider]
>
> type = identify
>
> match_header = User-Agent: Something Unique 1.0.0
>
> endpoint = provider
>
>
>
> It has to be an exact match though, no wildcards or regular expressions.
>
>
>
> I opened an issue[1] on separating ip matching from header matching so
> they can be re-ordered.
>
>
>
>
>
>
>
>
>
> [1] https://issues.asterisk.org/jira/browse/ASTERISK-27491
>
>
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] *On Behalf Of *George Joseph
> *Sent:* Thursday, December 14, 2017 10:59 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Is it possible to have two endpoints to
> the same IP address where one uses IP based authentication and the other
> requires asterisk to register to that system?
>
>
>
>
>
>
>
> On Wed, Dec 13, 2017 at 10:51 AM, Dan Cropp  wrote:
>
> Currently using PJSIP.  First, they want me to get this working with the
> existing PJSIP configuration, but then setup a second box using chan_sip
> performing similar work.
>
>
>
> For PJSIP…
>
> I currently have an endpoint configured to a system using IP based
> authentication.  It is configured with a match setting in the endpoint
> section.
>
> All channels coming from that IP address go to this endpoint.
>
>
&

Re: [asterisk-users] Asterisk 13.18.4 - New Error PJLIB_UTIL_EDNS_REFUSED

[George Joseph]

On Thu, Dec 21, 2017 at 8:18 AM, Bryant Zimmerman 
wrote:

> We just updated from 13.17.1 to 13.18.4 and are noticing a new error
>
> [2017-12-21 10:12:48] ERROR[32343]: res_pjsip.c:3850 endpt_send_request:
> Error 320055 'DNS "Refused" (PJLIB_UTIL_EDNS_REFUSED)' sending OPTIONS
> request to endpoint 6162480909.8009
>
> The DNS on the system seems to be working find. Anyone have an idea what
> could be triggering this issue?
>

I don't see anything offhand between 13.17.1 and 13.18.4 that would cause
these.  How often are they happening?



>
> Thanks
>
> Bryant Zimmerman (ZK Tech Inc.)
> 616-855-1030 Ext. 2003
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system?

[George Joseph]

On Mon, Dec 18, 2017 at 9:04 AM, Dan Cropp  wrote:

> Thanks George
>
>
>
> I originally didn’t have the 1002@ for the identify.  Changed that when
> things were not working.  I changed it back.
>
>
>
> Unfortunately, the system I am connecting with doesn’t seem to support the
> line support.  Looking at the SIP packets, I see Asterisk send it.
> Unfortunately, they do not send the line information as part of the
> INVITE.  I checked with some developers of that system and they do not know
> anything about the line setting.
>
> Is there any rfcs I could refer them to?
>

Yeah, I've found that some providers do and some providers don't.


https://tools.ietf.org/html/rfc3261#section-19.1

An implementation MUST include any provided transport, maddr, ttl, or
user parameter in the Request-URI of the formed request. If the URI
contains a method parameter, its value MUST be used as the method of
the request. The method parameter MUST NOT be placed in the
Request-URI.

*​​Unknown URI parameters MUST be placed in the message'sRequest-URI*.

The identify object also has the capability to match against a specific
header and value but it looks like it only tries to match on header if it
can't find a match by ip address.  Here's some info on it anyway.

If you're provider puts something unique and constant in the headers, like
a User-Agent string that doesn't change, you can also try using the
"match_header" parameter to an identify object.

[my_provider]
type = identify
match_header = User-Agent: Something Unique 1.0.0
endpoint = provider

It has to be an exact match though, no wildcards or regular expressions.

I opened an issue[1] on separating ip matching from header matching so they
can be re-ordered.




[1] https://issues.asterisk.org/jira/browse/ASTERISK-27491


>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] *On Behalf Of *George Joseph
> *Sent:* Thursday, December 14, 2017 10:59 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Is it possible to have two endpoints to
> the same IP address where one uses IP based authentication and the other
> requires asterisk to register to that system?
>
>
>
>
>
>
>
> On Wed, Dec 13, 2017 at 10:51 AM, Dan Cropp  wrote:
>
> Currently using PJSIP.  First, they want me to get this working with the
> existing PJSIP configuration, but then setup a second box using chan_sip
> performing similar work.
>
>
>
> For PJSIP…
>
> I currently have an endpoint configured to a system using IP based
> authentication.  It is configured with a match setting in the endpoint
> section.
>
> All channels coming from that IP address go to this endpoint.
>
>
>
> They want me to keep this endpoint, but add a new endpoint where we
> register with them.
>
>
>
> Existing…
>
> [transport1]
>
> type = transport
>
> bind = 0.0.0.0
>
> protocol = udp
>
>
>
> [1002]
>
> type = aor
>
> remove_existing = yes
>
> contact = sip:1...@xxx.xxx.xxx.xxx
>
>
>
> [1002]
>
> type = endpoint
>
> context = mycontext
>
> transport = transport1
>
> accountcode = 6
>
> dtmf_mode = inband
>
> device_state_busy_at = 48
>
> force_rport = no
>
> identify_by = username
>
> from_user = 1002
>
> disallow = all
>
> allow = ulaw
>
> acl = acl1
>
>
>
> [identify112]
>
> type = identify
>
> endpoint = 1002
>
> match = 1...@xxx.xxx.xxx.xxx
>
>
>
>
>
> Check this first...  identify112 probably failed to load because the match
> parameter can only take an ip address
>
> plus an optional netmask, or a hostname.  The '1002@' is invalid.
>
>
>
>
>
>
>
>
>
> I setup the registration and the endpoint.
>
>
>
> [286]
>
> type = aor
>
> remove_existing = yes
>
> contact = sip:2...@xxx.xxx.xxx.xxx
>
> qualify_frequency = 60
>
>
>
> [auth8]
>
> type = auth
>
> username = 286
>
> password = yyy
>
>
>
> [286]
>
> type = endpoint
>
> context = mycontext
>
> transport = transport1
>
> outbound_auth = auth8
>
> aors = 286
>
> accountcode = 22
>
> dtmf_mode = inband
>
> device_state_busy_at = 48
>
> force_rport = no
>
> disallow = all
>
> allow = ulaw
>
> acl = acl1
>
>
>
> [registration3]
>
> type = registration
>
> transport = transport1
>
> client_uri = sip:2...@zzz.zzz.zzz.zzz
>
> server_uri = sip:xxx.xxx.xxx.xxx
>
> contact_user = 286
>
> outbound_auth = auth8
>
> expiratio

Re: [asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system?

[George Joseph]

On Wed, Dec 13, 2017 at 10:51 AM, Dan Cropp  wrote:

> Currently using PJSIP.  First, they want me to get this working with the
> existing PJSIP configuration, but then setup a second box using chan_sip
> performing similar work.
>
>
>
> For PJSIP…
>
> I currently have an endpoint configured to a system using IP based
> authentication.  It is configured with a match setting in the endpoint
> section.
>
> All channels coming from that IP address go to this endpoint.
>
>
>
> They want me to keep this endpoint, but add a new endpoint where we
> register with them.
>
>
>
> Existing…
>
> [transport1]
>
> type = transport
>
> bind = 0.0.0.0
>
> protocol = udp
>
>
>
> [1002]
>
> type = aor
>
> remove_existing = yes
>
> contact = sip:1...@xxx.xxx.xxx.xxx
>
>
>
> [1002]
>
> type = endpoint
>
> context = mycontext
>
> transport = transport1
>
> accountcode = 6
>
> dtmf_mode = inband
>
> device_state_busy_at = 48
>
> force_rport = no
>
> identify_by = username
>
> from_user = 1002
>
> disallow = all
>
> allow = ulaw
>
> acl = acl1
>
>
>
> [identify112]
>
> type = identify
>
> endpoint = 1002
>
> match = 1...@xxx.xxx.xxx.xxx
>


Check this first...  identify112 probably failed to load because the match
parameter can only take an ip address
plus an optional netmask, or a hostname.  The '1002@' is invalid.




>
>
> I setup the registration and the endpoint.
>
>
>
> [286]
>
> type = aor
>
> remove_existing = yes
>
> contact = sip:2...@xxx.xxx.xxx.xxx
>
> qualify_frequency = 60
>
>
>
> [auth8]
>
> type = auth
>
> username = 286
>
> password = yyy
>
>
>
> [286]
>
> type = endpoint
>
> context = mycontext
>
> transport = transport1
>
> outbound_auth = auth8
>
> aors = 286
>
> accountcode = 22
>
> dtmf_mode = inband
>
> device_state_busy_at = 48
>
> force_rport = no
>
> disallow = all
>
> allow = ulaw
>
> acl = acl1
>
>
>
> [registration3]
>
> type = registration
>
> transport = transport1
>
> client_uri = sip:2...@zzz.zzz.zzz.zzz
>
> server_uri = sip:xxx.xxx.xxx.xxx
>
> contact_user = 286
>
> outbound_auth = auth8
>
> expiration = 3600
>
>
>
> The registration for the second endpoint works fine.  However, when I call
> through the other system for 286, it is failing.  For the INVITE from the
> other switch, the from_user varies depending on who is calling.  Asterisk
> logs report “No matching endpoint found” when it processes the INVITE for
> 286.
>
>
>
> I believe the reason INVITEs work for the other channel is because they
> are programmed to support the match for this IP address.
>
>
>
> Can anyone offer some suggestions?
>

You may be able to use the 'line and 'endpoint' registration parameters...
[registration3]
type = registration
...
line = yes
endpoint = 286

This causes asterisk to put the encoded endpoint name in the outgoing
Contact header.  If the provider properly echos back Contact parameters
when sending responses or new requests, asterisk will use the line
parameter to match an endpoint.  I'll have to double check but I believe we
do that BEFORE checking any identify object for a match.





>
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>
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Re: [asterisk-users] Explain how to maintain a compiled from source Asterisk instance ?

[George Joseph]

On Fri, Dec 8, 2017 at 8:08 AM, Olivier  wrote:

> Hello,
>
> When compiling Asterisk from source, the classical ./configure, make and
> make install commands are issued.
>
>
> If a vulnerabilty is found within Asterisk code, then Asterisk source code
> is patched and depending on what files were touched parts or all of above
> commands need to be re-issued.
>
> What should be done for Asterisk runtime dependencies ?
>
> Which of the following sentences is or are correct ?
>
> 1. All Asterisk runtime dependencies are delivered as .so files. Is this
> correct ?
>

Generally, yes but not delivered by the Asterisk team of course.  There may
be exceptions such as libsrtp.  Some older distributions only provide an
outdated version so we recommend that you compile and install it yourself
from source.


> 2. I don't need to re-configure, re-compile or even re-start asterisk when
> a such .so file is updated
>

This really depends on the nature of the change.  Generally you don't need
to re-configure or re-compile but if the dependent library changed some
public API that clients like us rely on, asterisk may behave very badly if
if the steps aren't re-run.  As for re-start, you do need to re-start.  The
operating system takes care of loading dependent libraries that aren't
actually asterisk modules and they are usually held in memory until the
process using them ends.  If they weren't and you did a package update to
replace a shared library, every process using it would crash.


> 3. Script install_prereq gives an approximate list of both build and
> runtime dependencies.
>

True.



>
> Best regards
>
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> org/
>
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>
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Re: [asterisk-users] How can I check backtrace files ?

[George Joseph]

On Wed, Dec 6, 2017 at 11:13 AM, Olivier  wrote:

>
>
> 2017-12-06 15:52 GMT+01:00 George Joseph :
>
>>
>>
>> On Tue, Dec 5, 2017 at 9:20 AM, Olivier  wrote:
>>
>>> Hello,
>>>
>>> I carefully read [1] which details how backtrace files can be produced.
>>>
>>> Maybe this seems natural to some, but how can I go one step futher, and
>>> check that produced XXX-thread1.txt, XXX-brief.txt, ... files are OK ?
>>>
>>> In other words, where can I find an example on how to use one of those
>>> files and check by myself, that if a system ever fails, I won't have to
>>> wait for another failure to provide required data to support teams ?
>>>
>>
>> It's a great question but I could spend a week answering it and not
>> scratch the surface. :)
>>
>
> Thanks very much for trying, anyway ;-)
>
>
>>  It's not a straightforward thing unless you know the code in question.
>> The most common is a segmentation fault (segfault or SEGV).
>>
>
> True ! I experienced segfaults lately and I could not configure the
> platform I used then (Debian Jessie) to produce core files in a directory
> Asterisk can write into.
> Now, with Debian Stretch, I can produce core file at will (with a kill -s
> SIGSEGV ).
> I checked ast_coredumped worked OK as it produced thread.txt files and so
> on.
>
> Ideally, I would like to go one step further: check now that a future .txt
> file would be "workable" (and not "you should have compiled with option XXX
> or configured with option YYY) .
>
>
>
>>   In that case, the thread1.txt file is the place to start.  Since most
>> of the objects passed around are really pointers to objects, the most
>> obvious cause would be a 0x0 for a value.  So for instance "chan=0x0".
>> That would be a pointer to a channel object that was not set when it
>> probably should have been.  Unfortunately, it's not only 0x0 that could
>> cause a segv.   Anytime a program tries to access memory it doesn't own,
>> that signal is raised.  So let's say there a 256 byte buffer which the
>> process owns.  If there's a bug somewhere that causes the program to try
>> and access bytes beyond the end of the buffer, you MAY get a segv if that
>> process doesn't also own that memory.  If this case, the backtrace won't
>> show anything obvious because the pointers all look valid.  There probably
>> would be an index variable (i or ix, etc) that may be set to 257 but you'd
>> have to know that the buffer was only 256 bytes to realize that that was
>> the issue.
>>
>
> So, with an artificial kill -s SIGSEGV , does the bellow
> output prove I have a workable .txt files (having .txt files that let
> people find the root cause of the issue is another story as we probably can
> only hope for the best here) ?
>
>
> # head core-brief.txt
> !@!@!@! brief.txt !@!@!@!
>
>
> Thread 38 (Thread 0x7f2aa5dd0700 (LWP 992)):
> #0  pthread_cond_timedwait@@GLIBC_2.3.2 () at ../sysdeps/unix/sysv/linux/
> x86_64/pthread_cond_timedwait.S:225
> #1  0x55cdcb69ae84 in __ast_cond_timedwait (filename=0x55cdcb7d4910
> "threadpool.c", lineno=1131, func=0x55cdcb7d4ea8 <__PRETTY_FUNCTION__.8978>
> "worker_idle", cond_name=0x55cdcb7d4b7f "&worker->cond",
> mutex_name=0x55cdcb7d4b71 "&worker->lock", cond=0x7f2abc000978,
> t=0x7f2abc0009a8, abstime=0x7f2aa5dcfc30) at lock.c:668
> #2  0x55cdcb75d153 in worker_idle (worker=0x7f2abc000970) at
> threadpool.c:1131
> #3  0x55cdcb75ce61 in worker_start (arg=0x7f2abc000970) at
> threadpool.c:1022
> #4  0x55cdcb769a8c in dummy_start (data=0x7f2abc000a80) at utils.c:1238
> #5  0x7f2aeddad494 in start_thread (arg=0x7f2aa5dd0700) at
> pthread_create.c:333
>


That's it!  The key pieces of information are the function names
(worker_idle, worker_start, etc.), the filename (threadpool.c, etc) and the
line numbers (1131, 1022, etc).




>
>
>> Deadlocks are even harder to troubleshoot.  For that, you need to look at
>> full.txt to see where the threads are stuck and find the 1 thread that's
>> holding the lock that the others are stuck on.
>>
>> Sorry.  I wish I had a better answer because it'd help a lot if folks
>> could do more investigation themselves.
>>
>>
>>
>>
>>
>>>
>>>
>>>
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Re: [asterisk-users] How can I check backtrace files ?

[George Joseph]

On Tue, Dec 5, 2017 at 9:20 AM, Olivier  wrote:

> Hello,
>
> I carefully read [1] which details how backtrace files can be produced.
>
> Maybe this seems natural to some, but how can I go one step futher, and
> check that produced XXX-thread1.txt, XXX-brief.txt, ... files are OK ?
>
> In other words, where can I find an example on how to use one of those
> files and check by myself, that if a system ever fails, I won't have to
> wait for another failure to provide required data to support teams ?
>

It's a great question but I could spend a week answering it and not scratch
the surface. :)   It's not a straightforward thing unless you know the code
in question.  The most common is a segmentation fault (segfault or SEGV).
In that case, the thread1.txt file is the place to start.  Since most of
the objects passed around are really pointers to objects, the most obvious
cause would be a 0x0 for a value.  So for instance "chan=0x0".  That would
be a pointer to a channel object that was not set when it probably should
have been.  Unfortunately, it's not only 0x0 that could cause a segv.
 Anytime a program tries to access memory it doesn't own, that signal is
raised.  So let's say there a 256 byte buffer which the process owns.  If
there's a bug somewhere that causes the program to try and access bytes
beyond the end of the buffer, you MAY get a segv if that process doesn't
also own that memory.  If this case, the backtrace won't show anything
obvious because the pointers all look valid.  There probably would be an
index variable (i or ix, etc) that may be set to 257 but you'd have to know
that the buffer was only 256 bytes to realize that that was the issue.

Deadlocks are even harder to troubleshoot.  For that, you need to look at
full.txt to see where the threads are stuck and find the 1 thread that's
holding the lock that the others are stuck on.

Sorry.  I wish I had a better answer because it'd help a lot if folks could
do more investigation themselves.





>
> Best regards
>
> [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>



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Re: [asterisk-users] taskprocessor.c: The 'sorcery/contact-00000015' task processor queue reached 1500 scheduled tasks.

[George Joseph]

On Thu, Nov 23, 2017 at 3:25 PM, Brian Capouch  wrote:

> Running 15.1.2.  I have four devices transitioned to use pjsip.
>
> After about 1-2 days of uptime, psjip stops accepting registrations,
> and the messages log contains the entry as per the subject.
>
> At any given time, "pjsip show contacts" only shows the four devices.
>
> Could someone point me to a fix, short of rebooting the server every day?
>
> Thanks.
>
> b.
>

Each pjsip object type (endpoint, aor, contact, etc) has a queue
(taskprocessor) to notify registered observers in a separate thread
whenever an item changes.  In your case, one of those observers has locked
and prevented the rest from running.  What kind of activity is happening on
those 4 contacts and are you using realtime or config files for pjsip
configuration?

Also, "core show taskprocessors" might give you some idea of what's going
on.


>
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>
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Re: [asterisk-users] Many tests from TestSuite fail with "Asterisk 127.0.0.1 received error: FRACK!"

[George Joseph]

On Tue, Oct 31, 2017 at 7:02 AM, Olivier  wrote:

> Hello,
>
> Thanks to Tzafrir and George help, I could run Asterisk TestSuite for the
> first time on a fresh Debian Stretch setup.
> TestSuite is installed with "apt-get install asterisk-testsuite" and
> Asterisk itself is stopped (with "service asterisk stop") before executing
> (as root) asterisk-tests-run.
>
> On console, I can see 14 instances of the following error
>
> asterisk:126 errReceived: Asterisk 127.0.0.1 received error: FRACK!,
> Failed assertion user_data is NULL (0) at line 121 in INTERNAL_OBJ of
> astobj2.c
> asterisk:126 errReceived: Asterisk 127.0.0.1 received error: FRACK!,
> Failed assertion user_data is NULL (0) at line 121 in INTERNAL_OBJ of
> astobj2.c
> asterisk:126 errReceived: Asterisk 127.0.0.1 received error: FRACK!,
> Failed assertion user_data is NULL (0) at line 121 in INTERNAL_OBJ of
> astobj2.c
>


Can you post the results from a single test?


>
> Is it worth reporting this ?
>
> Best regards
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
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>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>



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Re: [asterisk-users] How tu run runtests.py on Debian Stretch ?

[George Joseph]

On Tue, Oct 31, 2017 at 5:07 AM, Olivier  wrote:

> Hello,
>
> I'm giving asterisk-testsuite package a try on a fresh Debian Stretch
> setup.
>
> I've got this:
> # /usr/share/asterisk-testsuite/runtests.py
> Traceback (most recent call last):
>   File "/usr/share/asterisk-testsuite/runtests.py", line 24, in 
> from asterisk.version import AsteriskVersion
> ImportError: No module named asterisk.version
>

What version of python are you using?

Try
PYTHONPATH=./lib/python ./runtests.py
and see if that makes a difference


>
> 1. Which documentation shall I look for ?
> I've found [1]
>


That and
https://wiki.asterisk.org/wiki/display/AST/Installing+the+Asterisk+Test+Suite


>
>
> 2. How shall I run runtests.py ? As asterisk user ? As root ?
>

Most tests can be run as a normal user.  A few that use the pcap library
need root.  They should be ones that require yappcap.


>
>
> Best
>
>
> [1] https://wiki.asterisk.org/wiki/display/AST/Running+the+
> Asterisk+Test+Suite
>
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> org/
>
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>
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>



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[asterisk-users] Connect Two Existing Channels and Stop Listening

[Joseph Smith]

Hello all,

In my scenario I have two channels connected to Asterisk and in a stasis app.

I can put them both in a bridge and audio between them works as expected.  
However, I would like to free up the resource and no longer have Asterisk 
involved in the call if possible.


I'm currently playing with "redirect" with and without creating the bridge but 
not have any success.


Do you have advice on how to accomplish this?


Thanks
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Re: [asterisk-users] ERROR during high volume MoH dialplan

[Joseph Smith]

Thank you for the response Mike,

I did run into a CDR bottleneck as well and have already disabled it,

> module show like cdr
Module Description  Use 
Count  Status  Support Level
0 modules loaded

# grep enable= /etc/asterisk/cdr.conf
enable=no

At this point I'm really just not sure what the current bottleneck is and how 
to prevent the tasks for pooling.  I expected that the CPU would cap out before 
this occurred.  I do feel like there must be something I'm missing but just 
can't to it.

Any further suggestions are very welcome.

Thanks
Joseph

From: asterisk-users-boun...@lists.digium.com 
 on behalf of Mike 

Sent: Friday, September 1, 2017 3:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

I had that problem before – I believe “task processor queue reached 500 
scheduled tasks” crashing means your CDR records (queue) are being written as 
the call ends, and if you had many thousands of entries being written to disk 
it crashes asterisk (each ring to one phone is an entry, so it goes up fast – 
for example 10 busy phones, with a between-ring delay of 1 second means every 
second there are 10 entries being put in memory)

I was using a MySQL CDR, but I had left the “CSV” type of CDR on. I 
removed/disabled the CSV CDR module, kept on the SQL CDR only and things have 
been working fine ever since.

Mike

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Smith
Sent: September 1, 2017 16:41
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan


Thanks for the suggestion Tony,


I installed each codec for MoH, core sounds, and extra sound packages.  
Unfortunately the tests produce the same results.

[Sep  1 20:36:45] ERROR[10081][C-7fe5]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x20380b0 (

continuously for a while followed by a

[Sep  1 20:36:46] WARNING[7761][C-770d]: taskprocessor.c:888 
taskprocessor_push: The 'subp:PJSIP/sipp-0020' task processor queue reached 
500 scheduled tasks.

Then this time Asterisk actually crashed. :(


From: asterisk-users-boun...@lists.digium.com 
 on behalf of Tony Mountifield 

Sent: Friday, September 1, 2017 11:01 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

In article 
,
Joseph Smith  wrote:
>
> Thanks for the feedback.
>
> I do agree with having multiple smaller servers.  When I was first approached 
> with this task I mentioned as much.
> However, the current desire is to work with already existing hardware.  That 
> is out of my hands at the moment unless it
> just can't be done.  I will explore Freeswitch a bit soon to compare it as 
> well.
>
>
> I am struggling to find what the bottle neck is in this scenario.  Does 
> anyone have any advice on what that could be or
> on steps to discover it?   Do you think that tasks are pooling up because of 
> transcoding?  If so would it help to change
> the codec that is being used?  I am not sure about the MoH but the audio 
> files I am using are gsm.

You will find it less taxing on the server if you have MoH files and sounds 
files
available in all the possible native formats. Then Asterisk can use the 
appropriate
one for the channel without transcoding.

On my systems I have MoH and sounds installed in wav, ulaw, alaw, gsm and g729.

They will also sound better than transcoding from the gsm versions.

Cheers
Tony
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Re: [asterisk-users] ERROR during high volume MoH dialplan

[Joseph Smith]

Thanks for the suggestion Tony,


I installed each codec for MoH, core sounds, and extra sound packages.  
Unfortunately the tests produce the same results.

[Sep  1 20:36:45] ERROR[10081][C-7fe5]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x20380b0 (

continuously for a while followed by a

[Sep  1 20:36:46] WARNING[7761][C-770d]: taskprocessor.c:888 
taskprocessor_push: The 'subp:PJSIP/sipp-0020' task processor queue reached 
500 scheduled tasks.

Then this time Asterisk actually crashed. :(


From: asterisk-users-boun...@lists.digium.com 
 on behalf of Tony Mountifield 

Sent: Friday, September 1, 2017 11:01 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

In article 
,
Joseph Smith  wrote:
>
> Thanks for the feedback.
>
> I do agree with having multiple smaller servers.  When I was first approached 
> with this task I mentioned as much.
> However, the current desire is to work with already existing hardware.  That 
> is out of my hands at the moment unless it
> just can't be done.  I will explore Freeswitch a bit soon to compare it as 
> well.
>
>
> I am struggling to find what the bottle neck is in this scenario.  Does 
> anyone have any advice on what that could be or
> on steps to discover it?   Do you think that tasks are pooling up because of 
> transcoding?  If so would it help to change
> the codec that is being used?  I am not sure about the MoH but the audio 
> files I am using are gsm.

You will find it less taxing on the server if you have MoH files and sounds 
files
available in all the possible native formats. Then Asterisk can use the 
appropriate
one for the channel without transcoding.

On my systems I have MoH and sounds installed in wav, ulaw, alaw, gsm and g729.

They will also sound better than transcoding from the gsm versions.

Cheers
Tony
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Re: [asterisk-users] ERROR during high volume MoH dialplan

[Joseph Smith]

Thanks for the feedback.

I do agree with having multiple smaller servers.  When I was first approached 
with this task I mentioned as much.  However, the current desire is to work 
with already existing hardware.  That is out of my hands at the moment unless 
it just can't be done.  I will explore Freeswitch a bit soon to compare it as 
well.


I am struggling to find what the bottle neck is in this scenario.  Does anyone 
have any advice on what that could be or on steps to discover it?   Do you 
think that tasks are pooling up because of transcoding?  If so would it help to 
change the codec that is being used?  I am not sure about the MoH but the audio 
files I am using are gsm.


Thanks

Joseph



From: asterisk-users-boun...@lists.digium.com 
 on behalf of Pete Mundy 

Sent: Thursday, August 31, 2017 2:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

>> On Thu, 31 Aug 2017, Joseph Smith wrote:
>>
>> So I am looking for a better way to allow several thousand callers to listen 
>> to this IVR menu at the same time.


> On 1/09/2017, at 7:10 AM, Steve Edwards  wrote:
>
> I'm thinking multiple hosts.
>
> I'm not a fan of 4,000 eggs in one basket.


+1 for horizontal scaling as the best solution in this situation.

Pete

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Re: [asterisk-users] ERROR during high volume MoH dialplan

[Joseph Smith]

It is meant to simulate simultaneous calls on an IVR.  I have also tested with 
a separate set of audio files closer to what the actual IVR menu.  This 
produced the same result.


I apologize for not clearly stating the use case up front.  I will try to give 
a bit more detail on that now.


I have an IVR menu and submenu that users may dial into. I initially tested 
with the IVR audio files.  When I began experiencing this issue I used MoH as 
an attempt to narrow down the problem to the simplest dialplan possible.


If I continue my test at this volume or a higher volume, I begin to get errors 
about reaching the maximum queue size for that particular taskprocessor.  
Since, these error proceeded that I thought that they may be the key to 
preventing the queue from maxing out.


It sounds like Richard is saying that these refcount logs may not actually be 
errors and can be ignored in this scenario.  If that is the case then is there 
anything that can be done about the task processor queue size?  Is that simply 
a side effect of having so many callers listening to the IVR at the same time?

pjsip.conf is currently setup with a trunk allowing incoming calls from a 
specific IP.  This is the task processor that is maxing out.

So I am looking for a better way to allow several thousand callers to listen to 
this IVR menu at the same time.

Thank you for the feedback thus far.

Any info and advice is helpful.

Thanks
Joseph




From: asterisk-users-boun...@lists.digium.com 
 on behalf of Antony Stone 

Sent: Thursday, August 31, 2017 11:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

On Thursday 31 August 2017 at 18:15:54, Joseph Smith wrote:

> I was hoping Asterisk would handle more than 4k simultaneous calls.

I know from experience that Asterisk can handle more than 4k simultaneous
calls, however it's an extreme case to have all of them playing music on hold.

I think that if you tested 4k simultaneous calls with standard media streams
on the majority of them, you would not experience the problem.

Is this a real problem for you - that Asterisk can't manage 4k MoH sessions
simultaneously, even though it can manage 4k standard phone calls?


Antony.

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Re: [asterisk-users] ERROR during high volume MoH dialplan

[Joseph Smith]

Is there any more information I can provide to give insight to these errors?

Any further advice on avoiding these during high call volume?


I was hoping Asterisk would handle more than 4k simultaneous calls.

Thanks

Joseph



From: asterisk-users-boun...@lists.digium.com 
 on behalf of Joseph Smith 

Sent: Monday, August 28, 2017 5:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan


Hi Richard,

Thank you for the reply


Correct, I did mean 13.15.


I set no optimize and better backtrace through "make menuselect" and the output 
is now


[Aug 28 21:41:16] ERROR[17171][C-392d]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x21962b0 (0)

Got 26 backtrace records

#0: [0x61923f] main/utils.c:2475 __ast_assert_failed() (0x6191bb+84)

#1: [0x45ffc9] main/astobj2.c:543 __ao2_ref() (0x45fc3d+38C)

#2: [0x5320ce] main/frame.c:345 ast_frdup() (0x531e4c+282)

#3: [0x531a99] main/frame.c:196 ast_frisolate() (0x531a76+23)

#4: [0x60be51] main/translate.c:459 ast_trans_frameout() (0x60bd6e+E3)

#5: [0x60be75] main/translate.c:464 default_frameout()

#6: [0x60c46a] main/translate.c:579 ast_translate() (0x60c192+2D8)

#7: [0x4c0bf1] main/channel.c:5290 ast_write() (0x4bfb3e+10B3)

#8: [0x7fdef8345486] res/res_musiconhold.c:455 moh_files_generator()

#9: [0x4ba212] main/channel.c:3014 generator_force()

#10: [0x4bc23d] main/channel.c:3872 __ast_read()

#11: [0x4be29b] main/channel.c:4399 ast_read() (0x4be27e+1D)

#12: [0x4b6312] main/channel.c:1568 ast_safe_sleep_conditional() (0x4b6229+E9)

#13: [0x4b64c9] main/channel.c:1613 ast_safe_sleep() (0x4b64a1+28)

#14: [0x7fdef8346caa] res/res_musiconhold.c:834 play_moh_exec()

#15: [0x5970a3] main/pbx_app.c:491 pbx_exec() (0x596f87+11C)

#16: [0x582edf] main/pbx.c:2923 pbx_extension_helper()

#17: [0x586c30] main/pbx.c:4155 ast_spawn_extension() (0x586bcc+64)

#18: [0x5878bb] main/pbx.c:4328 __ast_pbx_run()

#19: [0x589061] main/pbx.c:4651 pbx_thread()

#20: [0x61624e] main/utils.c:1233 dummy_start()



* What codecs are you using in this setup?
In pjsip.conf I have disallow=all and allow=ulaw.  If I can provide more 
information or a better response to this question please guide me on how to do 
that.


Thanks
Joseph



From: asterisk-users-boun...@lists.digium.com 
 on behalf of Richard Mudgett 

Sent: Monday, August 28, 2017 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan



On Mon, Aug 28, 2017 at 1:04 PM, Joseph Smith 
mailto:warlock1...@hotmail.com>> wrote:

Hello,

I've recently setup a small load test against an instance of Asterisks.  I've 
tested on asterisk 13.5 and 14.6 with the same results.

I think you mean 13.15.0 as the excessive ref count trap is not in 13.5.0.

I am using PJSIP.  My dial plan is,

[test]

exten => 1001,1,Answer

exten => 1001,n,MusicOnHold(15)

exten => 1001,n,Hangup

I am using SIPP to test.  I can share XML if desired but it simply waits on the 
line while music plays for 8 seconds.  I used sippycup to generate it with the 
following steps in the yaml file.


steps:

  - invite

  - wait_for_answer

  - ack_answer

  - sleep 8

  - send_bye

At around 500 calls per second I begin to see the following ERRORs,


[Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup: Excessive 
refcount 10 reached on ao2 object 0x26bffc0

[Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x26bffc0 (0)

Got 19 backtrace records

#0: [0x45d229] /usr/sbin/asterisk(__ao2_ref+0x1a9) [0x45d229]

#1: [0x526ce6] /usr/sbin/asterisk(ast_frdup+0x116) [0x526ce6]

#2: [0x5fa616] /usr/sbin/asterisk(ast_translate+0x306) [0x5fa616]

#3: [0x4bf16b] /usr/sbin/asterisk(ast_write+0x104b) [0x4bf16b]

#4: [0x7efeb578230b] /usr/lib/asterisk/modules/res_musiconhold.so(+0x430b) 
[0x7efeb578230b]

#5: [0x4b5b52] /usr/sbin/asterisk() [0x4b5b52]

#6: [0x4c259c] /usr/sbin/asterisk() [0x4c259c]

#7: [0x4c4a45] /usr/sbin/asterisk() [0x4c4a45]

#8: [0x7efeb578478d] /usr/lib/asterisk/modules/res_musiconhold.so(+0x678d) 
[0x7efeb578478d]

#9: [0x58ec79] /usr/sbin/asterisk(pbx_exec+0xb9) [0x58ec79]

#10: [0x582e84] /usr/sbin/asterisk() [0x582e84]

#11: [0x584e7c] /usr/sbin/asterisk() [0x584e7c]

#12: [0x5863fb] /usr/sbin/asterisk() [0x5863fb]

#13: [0x60002a] /usr/sbin/asterisk() [0x60002a]

This inline backtrace would be more useful if you had BETTER_BACKTRACES enabled.



I've also seen similar behavior when using playback instead of MusicOnHold.  
CPU usage gets around 50%.  Can anyone enlighten me on the meaning and cause of 
the error?  Is there some steps (config etc) that can be taken to alleviate the 
issue?

This particular FRACK is meant to help fi

Re: [asterisk-users] ERROR during high volume MoH dialplan

[Joseph Smith]

Hi Richard,

Thank you for the reply


Correct, I did mean 13.15.


I set no optimize and better backtrace through "make menuselect" and the output 
is now


[Aug 28 21:41:16] ERROR[17171][C-392d]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x21962b0 (0)

Got 26 backtrace records

#0: [0x61923f] main/utils.c:2475 __ast_assert_failed() (0x6191bb+84)

#1: [0x45ffc9] main/astobj2.c:543 __ao2_ref() (0x45fc3d+38C)

#2: [0x5320ce] main/frame.c:345 ast_frdup() (0x531e4c+282)

#3: [0x531a99] main/frame.c:196 ast_frisolate() (0x531a76+23)

#4: [0x60be51] main/translate.c:459 ast_trans_frameout() (0x60bd6e+E3)

#5: [0x60be75] main/translate.c:464 default_frameout()

#6: [0x60c46a] main/translate.c:579 ast_translate() (0x60c192+2D8)

#7: [0x4c0bf1] main/channel.c:5290 ast_write() (0x4bfb3e+10B3)

#8: [0x7fdef8345486] res/res_musiconhold.c:455 moh_files_generator()

#9: [0x4ba212] main/channel.c:3014 generator_force()

#10: [0x4bc23d] main/channel.c:3872 __ast_read()

#11: [0x4be29b] main/channel.c:4399 ast_read() (0x4be27e+1D)

#12: [0x4b6312] main/channel.c:1568 ast_safe_sleep_conditional() (0x4b6229+E9)

#13: [0x4b64c9] main/channel.c:1613 ast_safe_sleep() (0x4b64a1+28)

#14: [0x7fdef8346caa] res/res_musiconhold.c:834 play_moh_exec()

#15: [0x5970a3] main/pbx_app.c:491 pbx_exec() (0x596f87+11C)

#16: [0x582edf] main/pbx.c:2923 pbx_extension_helper()

#17: [0x586c30] main/pbx.c:4155 ast_spawn_extension() (0x586bcc+64)

#18: [0x5878bb] main/pbx.c:4328 __ast_pbx_run()

#19: [0x589061] main/pbx.c:4651 pbx_thread()

#20: [0x61624e] main/utils.c:1233 dummy_start()



* What codecs are you using in this setup?
In pjsip.conf I have disallow=all and allow=ulaw.  If I can provide more 
information or a better response to this question please guide me on how to do 
that.


Thanks
Joseph



From: asterisk-users-boun...@lists.digium.com 
 on behalf of Richard Mudgett 

Sent: Monday, August 28, 2017 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan



On Mon, Aug 28, 2017 at 1:04 PM, Joseph Smith 
mailto:warlock1...@hotmail.com>> wrote:

Hello,

I've recently setup a small load test against an instance of Asterisks.  I've 
tested on asterisk 13.5 and 14.6 with the same results.

I think you mean 13.15.0 as the excessive ref count trap is not in 13.5.0.

I am using PJSIP.  My dial plan is,

[test]

exten => 1001,1,Answer

exten => 1001,n,MusicOnHold(15)

exten => 1001,n,Hangup

I am using SIPP to test.  I can share XML if desired but it simply waits on the 
line while music plays for 8 seconds.  I used sippycup to generate it with the 
following steps in the yaml file.


steps:

  - invite

  - wait_for_answer

  - ack_answer

  - sleep 8

  - send_bye

At around 500 calls per second I begin to see the following ERRORs,


[Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup: Excessive 
refcount 10 reached on ao2 object 0x26bffc0

[Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x26bffc0 (0)

Got 19 backtrace records

#0: [0x45d229] /usr/sbin/asterisk(__ao2_ref+0x1a9) [0x45d229]

#1: [0x526ce6] /usr/sbin/asterisk(ast_frdup+0x116) [0x526ce6]

#2: [0x5fa616] /usr/sbin/asterisk(ast_translate+0x306) [0x5fa616]

#3: [0x4bf16b] /usr/sbin/asterisk(ast_write+0x104b) [0x4bf16b]

#4: [0x7efeb578230b] /usr/lib/asterisk/modules/res_musiconhold.so(+0x430b) 
[0x7efeb578230b]

#5: [0x4b5b52] /usr/sbin/asterisk() [0x4b5b52]

#6: [0x4c259c] /usr/sbin/asterisk() [0x4c259c]

#7: [0x4c4a45] /usr/sbin/asterisk() [0x4c4a45]

#8: [0x7efeb578478d] /usr/lib/asterisk/modules/res_musiconhold.so(+0x678d) 
[0x7efeb578478d]

#9: [0x58ec79] /usr/sbin/asterisk(pbx_exec+0xb9) [0x58ec79]

#10: [0x582e84] /usr/sbin/asterisk() [0x582e84]

#11: [0x584e7c] /usr/sbin/asterisk() [0x584e7c]

#12: [0x5863fb] /usr/sbin/asterisk() [0x5863fb]

#13: [0x60002a] /usr/sbin/asterisk() [0x60002a]

This inline backtrace would be more useful if you had BETTER_BACKTRACES enabled.



I've also seen similar behavior when using playback instead of MusicOnHold.  
CPU usage gets around 50%.  Can anyone enlighten me on the meaning and cause of 
the error?  Is there some steps (config etc) that can be taken to alleviate the 
issue?

This particular FRACK is meant to help find ao2 object reference leaks.  It 
acts as an early warning for excessive references to any particular ao2 object 
used in the code.  The FRACK itself is benign.  Based upon the inline backtrace 
the ao2 object is likely to be a codec format.

* What codecs are you using in this setup?

* With 500 calls/sec and the calls lasting 8 seconds that comes to 4000 active 
channels.  Hitting the FRACK would result in an average of 25 references to the 
format per channel.  This is a simplistic calculation a

Re: [asterisk-users] What version of Linux?

[Joseph Smith]

Hello Ira,

 I recently installed on AMI to test out a bit before moving to physical 
hardware.  I had to install a number of packages to get it working their. You 
might be having a similar issue.  I installed the following packages before 
getting a completed configure and make.


** Caused error during configure

gcc-c++
ncurses-devel
libuuid-devel
libxml2-devel
sqlite-devel
patch
jansson-devel

**caused error duing make
openssl-devel
m2crypto


If you still have problems, paste some of the error generated by configure for 
reference


Good Luck
Joseph


From: asterisk-users-boun...@lists.digium.com 
 on behalf of Ira 

Sent: Monday, August 28, 2017 2:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] What version of Linux?

Hello Asterisk,

I've been running CentOS since 2006 or so and support for the 32
bit version recently ended. CentOS no longer offers a 32 bit
version so I thought I'd try Fedora 26 as they have 32 bit and
support. Got it installed, then downloaded Asterisk 14.6.0 but
can't seem to get it built. The configure script fails with some
error about CPP not working correctly? I did discover that
kernel-devel was not installed so I fixed that but I'm still
stuck.

Is the latest Fedora a good choice for an Asterisk box or should
I try something else. The machine is an Intel Atom board with a
Digium PCI analog board for my one last analog line.

I believe the board is limited to a 32 bit OS.

So two questions, is Fedora a good choice and if not, what
should I use for a machine running only Asterisk and Samba?

Is there a list of dependencies I need to install before
Asterisk will compile?

Thanks, Ira


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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
Getting Started - Asterisk Project - Asterisk Project 
Wiki<https://wiki.asterisk.org/wiki/display/AST/Getting+Started>
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When learning Asterisk it is important to start off on the right foot, so this 
section of the wiki covers orientation for learning Asterisk as well as 
installation ...




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[asterisk-users] ERROR during high volume MoH dialplan

[Joseph Smith]

Hello,

I've recently setup a small load test against an instance of Asterisks.  I've 
tested on asterisk 13.5 and 14.6 with the same results.

I am using PJSIP.  My dial plan is,

[test]

exten => 1001,1,Answer

exten => 1001,n,MusicOnHold(15)

exten => 1001,n,Hangup

I am using SIPP to test.  I can share XML if desired but it simply waits on the 
line while music plays for 8 seconds.  I used sippycup to generate it with the 
following steps in the yaml file.


steps:

  - invite

  - wait_for_answer

  - ack_answer

  - sleep 8

  - send_bye

At around 500 calls per second I begin to see the following ERRORs,


[Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup: Excessive 
refcount 10 reached on ao2 object 0x26bffc0

[Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x26bffc0 (0)

Got 19 backtrace records

#0: [0x45d229] /usr/sbin/asterisk(__ao2_ref+0x1a9) [0x45d229]

#1: [0x526ce6] /usr/sbin/asterisk(ast_frdup+0x116) [0x526ce6]

#2: [0x5fa616] /usr/sbin/asterisk(ast_translate+0x306) [0x5fa616]

#3: [0x4bf16b] /usr/sbin/asterisk(ast_write+0x104b) [0x4bf16b]

#4: [0x7efeb578230b] /usr/lib/asterisk/modules/res_musiconhold.so(+0x430b) 
[0x7efeb578230b]

#5: [0x4b5b52] /usr/sbin/asterisk() [0x4b5b52]

#6: [0x4c259c] /usr/sbin/asterisk() [0x4c259c]

#7: [0x4c4a45] /usr/sbin/asterisk() [0x4c4a45]

#8: [0x7efeb578478d] /usr/lib/asterisk/modules/res_musiconhold.so(+0x678d) 
[0x7efeb578478d]

#9: [0x58ec79] /usr/sbin/asterisk(pbx_exec+0xb9) [0x58ec79]

#10: [0x582e84] /usr/sbin/asterisk() [0x582e84]

#11: [0x584e7c] /usr/sbin/asterisk() [0x584e7c]

#12: [0x5863fb] /usr/sbin/asterisk() [0x5863fb]

#13: [0x60002a] /usr/sbin/asterisk() [0x60002a]


I've also seen similar behavior when using playback instead of MusicOnHold.  
CPU usage gets around 50%.  Can anyone enlighten me on the meaning and cause of 
the error?  Is there some steps (config etc) that can be taken to alleviate the 
issue?

Thanks
Joseph




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Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

[George Joseph]

On Tue, Aug 1, 2017 at 12:53 PM, Carlos Chavez  wrote:

>  I am having a very tough time trying to replace an Elastix 2.X
> install running as a virtual machine on ESXI 4.  I tried using the Freepbx
> 14 ISO that installs CentOS 6 along with Asterisk 13.16 but I keep getting
> random segfaults:
>
> [175711.476685] asterisk[2942]: segfault at 188 ip 7fc6c41abffc sp
> 7fc608575890 error 4 in libasteriskpj.so.2[7fc6c4144000+14c000]
>

The messages that get dumped to the kernel log aren't of much use.  See
below for more info.


>
>  I then proceeded to install a CentOS 7.3 VM and compiled Asterisk
> 13.17.0 by hand.


That *should* be a good combination.


> We are still using Freepbx 14 for the front end.  We did some testing over
> the weekend and calls were coming in and out and all extensions were
> registered.  Come Monday Asterisk started segfaulting again with exactly
> the same error.  Maybe VMware is too old to support the newer CentOS and
> Asterisk?  The Elastix install is based on CentOS 5 and Asterisk 1.6.  I
> have no idea how to approach this.  It only segfaults when there are only
> more than a couple simultaneous calls, that is why testing with only a
> couple of calls worked.
>
>  There are several core dump files but I really do not know how to use
> them for debugging Asterisk.  Any ideas?


If you still have the coredump files, you can use the ast_coredumper
utility located in /var/lib/asterisk/scripts to extract the human-readable
stack traces.  "sudo /var/lib/asterisk/scripts/ast_coredumper --help" will
get you more info on how to run the command.  The *-thread1.txt file it
produces will be the most helpful but all 4 should be attached to the
Asterisk issue should you decide to create one.



> I will try using chan_sip instead of PJSIP to get things running but
> confidence is not high.
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez
> dCAP #1349
> +52 (55)8116-9161
>
> --
> _
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>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users




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Re: [asterisk-users] dahdi kernel module

[Anthony Joseph Messina]

On Sunday, July 30, 2017 4:49:31 PM CDT Greg Woods wrote:
> Does anyone know if there are any plans to update the dahdi-linux kernel
> module code? It no longer compiles with recent kernels, and the last
> release of dahdi-linux appears to have been around March of 2016. I am
> currently running 4.6.3-300.fc24.x86_64 (on a Fedora system obviously) and
> the dahdi-linux-complete-2.11.1+2.11.1 release builds and runs under this
> kernel, but if I try to build it under any Fedora kernel more recent than
> this, I get:
> 
> [root@worldsys dahdi-linux-master]# make
> make -C drivers/dahdi/firmware firmware-loaders
> make[1]: Entering directory
> '/local/src/dahdi-linux-master/drivers/dahdi/firmware'
> make[1]: Leaving directory
> '/local/src/dahdi-linux-master/drivers/dahdi/firmware'
> make -C /lib/modules/4.11.12-100.fc24.x86_64/build
> SUBDIRS=/local/src/dahdi-linux-master/drivers/dahdi
> DAHDI_INCLUDE=/local/src/dahdi-linux-master/include DAHDI_MODULES_EXTRA=" "
> HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m
> make[1]: Entering directory '/usr/src/kernels/4.11.12-100.fc24.x86_64'
>   CC [M]  /local/src/dahdi-linux-master/drivers/dahdi/dahdi-base.o
> /local/src/dahdi-linux-master/drivers/dahdi/dahdi-base.c: In function
> ‘dahdi_ioctl_iomux’:
> /local/src/dahdi-linux-master/drivers/dahdi/dahdi-base.c:5954:7: error:
> implicit declaration of function ‘signal_pending’
> [-Werror=implicit-function-declaration]
>if (signal_pending(current)) {
>^~
> cc1: some warnings being treated as errors
> scripts/Makefile.build:294: recipe for target
> '/local/src/dahdi-linux-master/drivers/dahdi/dahdi-base.o' failed
> make[2]: *** [/local/src/dahdi-linux-master/drivers/dahdi/dahdi-base.o]
> Error 1
> Makefile:1496: recipe for target
> '_module_/local/src/dahdi-linux-master/drivers/dahdi' failed
> make[1]: *** [_module_/local/src/dahdi-linux-master/drivers/dahdi] Error 2
> make[1]: Leaving directory '/usr/src/kernels/4.11.12-100.fc24.x86_64'
> 
> (This particular run was using the master download from github, but the
> results are the same if I try to build the 2.11.1+2.11.1  release from
> Digium's downloads site).
> 
> If I can't find a way around this, my only options are to junk a $600
> telephony card (I shudder to think how much it would cost to replace it now
> with one that has a maintained driver) or keep running a non-updateable
> kernel.
> 
> Thanks,
> --Greg

https://issues.asterisk.org/jira/browse/DAHLIN-354

I've patched my builds here:
https://messinet.com/rpms/browser/dahdi-linux-kmod/dahdi-linux-kmod.spec

It looks like you're using F24, so you might be able to rebuild using the 
SRPMs https://messinet.com/pub/fedora/linux/updates/26/SRPMS/


-- 
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F9B6 560E 68EA 037D 8C3D  D1C9 FF31 3BDB D9D8 99B6


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[asterisk-users] Asterisk community servers are currently unavailable

[George Joseph]

We're investigating

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[asterisk-users] PJProject 2.6

[George Joseph]

>From the blog...  http://blogs.asterisk.org/2017/03/01/pjproject-2-6/

This week, we’re pleased to say that we’ve updated the Asterisk 13, 14 and
master branches’ bundled version of pjproject to 2.6.

Here’s a short recap of the steps we took to get here:

   - All of the the patches we were applying to 2.5.5 were verified to be
   in 2.6.
   - We looked for any other functional or API changes that might affect
   how Asterisk uses pjproject.
  - We found 1 minor improvement in how memory pools are released.
  This resulted in 3 minor code tweaks that are backwards compatible.
   - We tested the build process looking for issues that might change how
   Asterisk compiles and links pjproject.
  - We found an issue with WebRTC…  In this release, pjproject now
  includes a WebRTC implementation in its third-party directory
which we had
  to disable.  These were all minor updates but required to get
pjproject to
  build successfully and they do not affect Asterisk’s ability to process
  WebRTC calls.
 - One of pjproject’s configure options was changed from
 ‘–disable-webrtc’ to ‘–without-external-webrtc’
 - The build of WebRTC was removed from the Makefile
 - A #define was added to config_site.h to prevent pjmedia from
 requiring WebRTC.
  - We ran the Asterisk Testsuite a few dozen times to make sure the
   functional tests still passed.
   - Finally, for the first time, we were able to run stress tests to look
   for any new performance or stability issues that might have crept in.  We
   didn’t find any.

Of course we could have missed something, which is why it’s important for
the community to test for themselves.  If you’re using the bundled version
of pjproject, and you should be :), checkout the Asterisk 13 or 14 branch
and test it in your environment.  If you build pjproject yourself, you can
try it with recent Asterisk releases.

For more information related to Astrerisk’s use of pjproject, visit Building
and Installing pjproject
<https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject>
on
the Asterisk Wiki <https://wiki.asterisk.org/wiki/display/AST/Home>.


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Re: [asterisk-users] Advices when Asterisk segfaults and nothing useful in logs

[George Joseph]

On Fri, Feb 17, 2017 at 5:17 AM, Olivier  wrote:

> Hi George,
>
> How does ast_coredumper compare to ast_grab_core ) ?
> Is it worth learning to use both or shall favor one ?
>
> PS: As I don't know either program, yet, my question may seem silly.
> Please, forgive me for this
>

Not silly at all.

ast_grab_core actually kills asterisk to get the core file while
ast_coredumper dumps the core file and let's asterisk continue.  If
asterisk is truly deadlocked this may not matter but in some situations you
might not want to kill asterisk.

If asterisk was compiled with DEBUG_THREADS, ast_coredumper dumps the locks
table which is critical in debugging deadlock scenarios.  It does this from
the coredump, as opposed to running "core show locks", which makes the
locks table consistent with the rest of the backtraces.

ast_coiredumper can also find and process existing coredumps and it can
create a tarball containing the backtraces and lock tables.



>
> 2017-02-14 22:52 GMT+01:00 George Joseph :
>
>>
>>
>> On Tue, Feb 14, 2017 at 2:51 PM, George Joseph 
>> wrote:
>>
>>>
>>>
>>> On Tue, Feb 14, 2017 at 10:21 AM, Olivier  wrote:
>>>
>>>> Hello,
>>>>
>>>> I've got a 13.13.1 system using PJSIP stack on debian Jessie.
>>>> It runs from 50 to 100 simultaneous calls (so 100 to 200  PJSIP
>>>> channels) all day long.
>>>> From time to time, roughly meaning once a month, it segfaults  with
>>>> lines (from dmesg -T output) like this:
>>>> asterisk[1160]: segfault at 7efe ip 005881d6 sp
>>>> 7fec95c33910 error 4 in asterisk[40+2a2000]
>>>>
>>>>
>>>> Debug level was unfortunately not set in asterisk.conf but verbose
>>>> level was set to 5.
>>>> Asterisk runs with:
>>>> /usr/sbin/asterisk -U asterisk -G asterisk -g
>>>>
>>>> Asterisk is compiled with DONT_OPTIMIZE and BETTER_BACKTRACES options.
>>>>
>>>> "core show settings" outputs:
>>>> * Directories
>>>>   -
>>>>   Configuration file:
>>>>   Configuration directory: /etc/asterisk
>>>>   Module directory:/usr/lib/asterisk/modules
>>>>   Spool directory: /var/spool/asterisk
>>>>   Log directory:   /var/log/asterisk
>>>>   Run/Sockets directory:   /var/run/asterisk
>>>>   PID file:/var/run/asterisk/asterisk.pid
>>>>   VarLib directory:/var/lib/asterisk
>>>>   Data directory:  /var/lib/asterisk
>>>>   ASTDB:   /var/lib/asterisk/astdb
>>>>   IAX2 Keys directory: /var/lib/asterisk/keys
>>>>
>>>>
>>>>
>>>> 1. Am I correct to expect a coredump file to be produced anytime
>>>> asterisk segfaults ?
>>>>
>>>
>>> Yes if -g is set and the user that's running asterisk has permissions to
>>> set ulimit -c.
>>>
>>>
>>>>
>>>> 2. Does Asterisk prints any WARNING or ERROR message whenever it
>>>> detects, at startup preferably, that it has not required permissions to
>>>> write a coredump file ?
>>>>
>>>
>>> No because it's the system that determines where a coredump goes and
>>> actually writes it, not asterisk.
>>> It's the sysctl kernel.core_pattern setting.
>>>
>>>
>>>>
>>>> 3. Among above directories, which one is choosen to save coredump files
>>>> ? Is it something that can/should be configured in /etc/asterisk (I've seen
>>>> related options in some debian  /etc/default/asterisk files but I would be
>>>> curious to know if such things exist
>>>>
>>>
>>> See above.
>>>
>>>
>>>>
>>>> 4. Is there anything useful I can do with a line such as :
>>>> asterisk[1160]: segfault at 7efe ip 005881d6 sp
>>>> 7fec95c33910 error 4 in asterisk[40+2a2000] ? Any pointer ?
>>>>
>>>
>>> Nope.  Not a thing.  Sorry.
>>>
>>>
>>>
>>>>
>>>> 5. Suggestions ?
>>>>
>>>
>>> If you can at least get the system to write a coredump file, there are
>>> new utilities in /var/lib/asterisk/scripts, namely ast_coredumper which can
>>> help create the backtraces if it can at least find the core file.  Just run
>>

Re: [asterisk-users] Advices when Asterisk segfaults and nothing useful in logs

[George Joseph]

On Tue, Feb 14, 2017 at 2:51 PM, George Joseph  wrote:

>
>
> On Tue, Feb 14, 2017 at 10:21 AM, Olivier  wrote:
>
>> Hello,
>>
>> I've got a 13.13.1 system using PJSIP stack on debian Jessie.
>> It runs from 50 to 100 simultaneous calls (so 100 to 200  PJSIP channels)
>> all day long.
>> From time to time, roughly meaning once a month, it segfaults  with lines
>> (from dmesg -T output) like this:
>> asterisk[1160]: segfault at 7efe ip 005881d6 sp
>> 7fec95c33910 error 4 in asterisk[40+2a2000]
>>
>>
>> Debug level was unfortunately not set in asterisk.conf but verbose level
>> was set to 5.
>> Asterisk runs with:
>> /usr/sbin/asterisk -U asterisk -G asterisk -g
>>
>> Asterisk is compiled with DONT_OPTIMIZE and BETTER_BACKTRACES options.
>>
>> "core show settings" outputs:
>> * Directories
>>   -
>>   Configuration file:
>>   Configuration directory: /etc/asterisk
>>   Module directory:/usr/lib/asterisk/modules
>>   Spool directory: /var/spool/asterisk
>>   Log directory:   /var/log/asterisk
>>   Run/Sockets directory:   /var/run/asterisk
>>   PID file:/var/run/asterisk/asterisk.pid
>>   VarLib directory:/var/lib/asterisk
>>   Data directory:  /var/lib/asterisk
>>   ASTDB:   /var/lib/asterisk/astdb
>>   IAX2 Keys directory: /var/lib/asterisk/keys
>>
>>
>>
>> 1. Am I correct to expect a coredump file to be produced anytime asterisk
>> segfaults ?
>>
>
> Yes if -g is set and the user that's running asterisk has permissions to
> set ulimit -c.
>
>
>>
>> 2. Does Asterisk prints any WARNING or ERROR message whenever it detects,
>> at startup preferably, that it has not required permissions to write a
>> coredump file ?
>>
>
> No because it's the system that determines where a coredump goes and
> actually writes it, not asterisk.
> It's the sysctl kernel.core_pattern setting.
>
>
>>
>> 3. Among above directories, which one is choosen to save coredump files ?
>> Is it something that can/should be configured in /etc/asterisk (I've seen
>> related options in some debian  /etc/default/asterisk files but I would be
>> curious to know if such things exist
>>
>
> See above.
>
>
>>
>> 4. Is there anything useful I can do with a line such as :
>> asterisk[1160]: segfault at 7efe ip 005881d6 sp
>> 7fec95c33910 error 4 in asterisk[40+2a2000] ? Any pointer ?
>>
>
> Nope.  Not a thing.  Sorry.
>
>
>
>>
>> 5. Suggestions ?
>>
>
> If you can at least get the system to write a coredump file, there are new
> utilities in /var/lib/asterisk/scripts, namely ast_coredumper which can
> help create the backtraces if it can at least find the core file.  Just run
> "./ast_coredumper --help" for more info.   You should also be able to use
> those utilities with earlier Asterisk 13 versions.
>
>
>

Oh yeah, and it's on my list to publish instructions on how ot use those
utilities but they were just released yesterday.



>
>> Best regards
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> George Joseph
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
>


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Re: [asterisk-users] Advices when Asterisk segfaults and nothing useful in logs

[George Joseph]

On Tue, Feb 14, 2017 at 10:21 AM, Olivier  wrote:

> Hello,
>
> I've got a 13.13.1 system using PJSIP stack on debian Jessie.
> It runs from 50 to 100 simultaneous calls (so 100 to 200  PJSIP channels)
> all day long.
> From time to time, roughly meaning once a month, it segfaults  with lines
> (from dmesg -T output) like this:
> asterisk[1160]: segfault at 7efe ip 005881d6 sp
> 7fec95c33910 error 4 in asterisk[40+2a2000]
>
>
> Debug level was unfortunately not set in asterisk.conf but verbose level
> was set to 5.
> Asterisk runs with:
> /usr/sbin/asterisk -U asterisk -G asterisk -g
>
> Asterisk is compiled with DONT_OPTIMIZE and BETTER_BACKTRACES options.
>
> "core show settings" outputs:
> * Directories
>   -
>   Configuration file:
>   Configuration directory: /etc/asterisk
>   Module directory:/usr/lib/asterisk/modules
>   Spool directory: /var/spool/asterisk
>   Log directory:   /var/log/asterisk
>   Run/Sockets directory:   /var/run/asterisk
>   PID file:/var/run/asterisk/asterisk.pid
>   VarLib directory:/var/lib/asterisk
>   Data directory:  /var/lib/asterisk
>   ASTDB:   /var/lib/asterisk/astdb
>   IAX2 Keys directory: /var/lib/asterisk/keys
>
>
>
> 1. Am I correct to expect a coredump file to be produced anytime asterisk
> segfaults ?
>

Yes if -g is set and the user that's running asterisk has permissions to
set ulimit -c.


>
> 2. Does Asterisk prints any WARNING or ERROR message whenever it detects,
> at startup preferably, that it has not required permissions to write a
> coredump file ?
>

No because it's the system that determines where a coredump goes and
actually writes it, not asterisk.
It's the sysctl kernel.core_pattern setting.


>
> 3. Among above directories, which one is choosen to save coredump files ?
> Is it something that can/should be configured in /etc/asterisk (I've seen
> related options in some debian  /etc/default/asterisk files but I would be
> curious to know if such things exist
>

See above.


>
> 4. Is there anything useful I can do with a line such as :
> asterisk[1160]: segfault at 7efe ip 005881d6 sp
> 7fec95c33910 error 4 in asterisk[40+2a2000] ? Any pointer ?
>

Nope.  Not a thing.  Sorry.



>
> 5. Suggestions ?
>

If you can at least get the system to write a coredump file, there are new
utilities in /var/lib/asterisk/scripts, namely ast_coredumper which can
help create the backtraces if it can at least find the core file.  Just run
"./ast_coredumper --help" for more info.   You should also be able to use
those utilities with earlier Asterisk 13 versions.



>
> Best regards
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
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Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] asterisk callerid issue PJSIP Realtime

[George Joseph]

On Thu, Feb 2, 2017 at 4:06 AM, Zakir Mahomedy  wrote:

> Yes, from_user was set, removing those entries solved the problem.
>
> Can someone please explain to me the correct use for fromuser field?
>

from_user forces the user portion of the From header to a specific value on
calls that go TO the device represented by the endpoint.  Most often it's
used with a service provider when the service provider requires that all
calls it accepts have some sort of account identifier in the From header
instead of the original caller's info.  I can't think of a scenario where
you'd need to use from_user with a phone.


>
> thanks
> Zakir
>
>
> On Wednesday, February 1, 2017 8:00 PM, "asterisk-users-request@lists.
> digium.com"  wrote:
>
>
> Send asterisk-users mailing list submissions to
> asterisk-users@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
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>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> Today's Topics:
>
>   1. asterisk  callerid issue PJSIP Realtime (Zakir Mahomedy)
>   2. Re: asterisk callerid issue PJSIP Realtime (George Joseph)
>
>
> --
>
> Message: 1
> Date: Wed, 1 Feb 2017 13:50:57 + (UTC)
> From: Zakir Mahomedy 
> To: "asterisk-users@lists.digium.com"
> 
> Subject: [asterisk-users] asterisk  callerid issue PJSIP Realtime
> Message-ID: <1998594554.250932.1485957057...@mail.yahoo.com>
> Content-Type: text/plain; charset="utf-8"
>
> I recently rolled out a new server with asterisk 14. ?On the Called user
> phone, the caller ID is the same as the Called User.
> eg) ext ?406 ?with callerid 406 ? calls ext 405 ,??on the caller id on the
> ext 405 phone displaying 405.
>
>
> We are using realtime PJSIP, I set the callerid field in the database but
> no luck.?
> - Executing [405@common:1] NoOp("PJSIP/406-000f", ""DEBUGGING PJSIP
> CLID"") in new stack
> - Executing [405@common:2] NoOp("PJSIP/406-000f", "CALLERID = ?"ross"
> <406>") in new stack- Executing [405@common:3] Dial("PJSIP/406-000f",
> "PJSIP/405") in new stack
> In the above dialplan, the callerid is been taken from the database PJSIP
> endpoints.?
> Here is the sip debugger files
> INVITE sip:405@192.168.1.27 SIP/2.0Via: SIP/2.0/UDP 192.168.1.82:5060
> ;branch=z9hG4bK714210067;rportFrom: "zak" 
> ;tag=2071662084To:
> Call-ID: 50172054-506...@bjc.bgi.B.ICCSeq: 21
> INVITEContact: "zak" Authorization: Digest
> username="406", realm="asterisk", nonce="1485956409/
> e852b2a5e081f01421212d9a6ca954fa", uri="sip:405@192.168.1.27", response="
> ef94bae123f16dc5d9314a43922c949d", algorithm=md5, cnonce="13226017",
> opaque="50d490d233efd03e", qop=auth, nc=0003
>
> INVITE sip:405@192.168.1.209:36767;ob SIP/2.0Via: SIP/2.0/UDP
> 197.245.99.113:5060;rport;branch=z9hG4bKPj2f9d3dde-5ec4-49e1-b92d-7b4091b3138bFrom:
> ;tag=e4a0ecf6-c74e-4ab5-8438-bac5c073e328To:  405@192.168.1.209;ob>Contact: Call-ID:
> b4a83465-9105-4c70-9da1-11f410c37657
>
> <--- Received SIP response (515 bytes) from UDP:192.168.1.209:36767
> --->SIP/2.0 180 RingingVia: SIP/2.0/UDP 197.245.99.113:5060;rport=
> 5060;received=192.168.1.27;branch=z9hG4bKPj70fb8ef9-d99c-4e5b-88a5-eecbf7dd7682Call-ID:
> f0b31a86-0ac3-47f0-8b13-487d54982e9bFrom: ;tag=
> 77ea8869-273a-4f65-8128-e334b445f970To: ;
> tag=jurMewPN-95CgNyoQbhRCFpbH90hKw1dCSeq: 12221 INVITEContact:  405@192.168.1.209:36767;ob>Allow: PRACK, INVITE, ACK, B
>
>
> ?ParameterName ? ? ? ? ? ? ? ? ? ? ?: ParameterValue?===
> ==?callerid ? ? ? ? ? ? ? ? ? ? ?
> ? ? : "john doe" <405>?callerid_privacy ? ? ? ? ? ? : allowed?callerid_tag
> ? ? ? ? ? ? ? ? ? ?:
> Zakir
>
> -- next part --
> An HTML attachment was scrubbed...
> URL: <http://lists.digium.com/pipermail/asterisk-users/
> attachments/20170201/ede9ff18/attachment-0001.html>
>
> --
>
> Message: 2
> Date: Wed, 1 Feb 2017 08:52:59 -0700
> From: George Joseph 
> To: Zakir Mahomedy ,  Asterisk Users Mailing List
> - Non-Commercial Disc

Re: [asterisk-users] asterisk callerid issue PJSIP Realtime

[George Joseph]

On Wed, Feb 1, 2017 at 6:50 AM, Zakir Mahomedy  wrote:

> I recently rolled out a new server with asterisk 14.
> On the Called user phone, the caller ID is the same as the Called User.
>
> eg) ext  406  with callerid 406   calls ext 405 ,
>
> on the caller id on the ext 405 phone displaying 405.
>
>
>
> We are using realtime PJSIP, I set the callerid field in the database but
> no luck.
>
> - Executing [405@common:1] NoOp("PJSIP/406-000f", ""DEBUGGING PJSIP
> CLID"") in new stack
> - Executing [405@common:2] NoOp("PJSIP/406-000f", "CALLERID =  "ross"
> <406>") in new stack
> - Executing [405@common:3] Dial("PJSIP/406-000f", "PJSIP/405") in new
> stack
>
> In the above dialplan, the callerid is been taken from the database PJSIP
> endpoints.
>
> Here is the sip debugger files
>
> INVITE sip:405@192.168.1.27 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.82:5060;branch=z9hG4bK714210067;rport
> From: "zak" ;tag=2071662084
> To: 
> Call-ID: 50172054-506...@bjc.bgi.b.ic
> CSeq: 21 INVITE
> Contact: "zak" 
> Authorization: Digest username="406", realm="asterisk", nonce="1485956409/
> e852b2a5e081f01421212d9a6ca954fa", uri="sip:405@192.168.1.27", response="
> ef94bae123f16dc5d9314a43922c949d", algorithm=md5, cnonce="13226017",
> opaque="50d490d233efd03e", qop=auth, nc=0003
>
>
> INVITE sip:405@192.168.1.209:36767;ob SIP/2.0
> Via: SIP/2.0/UDP 197.245.99.113:5060;rport;branch=z9hG4bKPj2f9d3dde-5ec4-
> 49e1-b92d-7b4091b3138b
> From: ;tag=e4a0ecf6-c74e-4ab5-8438-bac5c073e328
>


On 405's endpoiint, you're not forcing from_user to 405 are you?




> To: 
> Contact: 
> Call-ID: b4a83465-9105-4c70-9da1-11f410c37657
>
>
> <--- Received SIP response (515 bytes) from UDP:192.168.1.209:36767 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 197.245.99.113:5060;rport=5060;received=192.168.1.27;
> branch=z9hG4bKPj70fb8ef9-d99c-4e5b-88a5-eecbf7dd7682
> Call-ID: f0b31a86-0ac3-47f0-8b13-487d54982e9b
> From: ;tag=77ea8869-273a-4f65-8128-e334b445f970
> To: ;tag=jurMewPN-95CgNyoQbhRCFpbH90hKw1d
> CSeq: 12221 INVITE
> Contact: 
> Allow: PRACK, INVITE, ACK, B
>
>
>
>  ParameterName  : ParameterValue
>  =
>  callerid   : "john doe" <405>
>  callerid_privacy : allowed
>  callerid_tag:
>
> Zakir
>
>
>
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> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] Can't comile bundled PJSIP on CentOS 7

[George Joseph]

On Mon, Jan 9, 2017 at 5:15 PM, Olivier  wrote:

> Hello,
>
> I'm setting up an Asterisk 13.13.1 cluster on two CentOS boxes.
>
> I followed this:
> cd /usr/src
> wget  ... asterisk-13.13.1.tar.gz
> tar zxf asterisk-13.13.1.tar.gz
> cd asterisk-13.13.1
> ASTERISK_CONFIGURE="--libdir=/usr/lib64 --prefix=/usr"
> ./configure ${ASTERISK_CONFIGURE}  --with-pjproject-bundled
> make menuselect (shows res-srtp is available)
> make
>
> latest make command fails with (see [1]:
>
>[GENERATE] libasteriskpj.exports
>[LD] libasteriskpj.o -> libasteriskpj.so.2
> /bin/ld: cannot find -lsrtp-x86_64-unknown-linux-gnu
>
>
> So I also build a simple VM box in which I also install CentOS and
> Asterisk 13.3.1 but I could run the above instructions successfully.
>
>
> Suggestions ?
>

Could you have had installed pjproject into the system locations
previously?  Specifically, are there any libpj* files in /usr/lib64 or pj*
files or directories in /usr/include?  If so, try uninstalling it before
building --with-pjproject-bundled.  There may be a conflict.

The newest versions of pjproject require libsrtp 2 which doesn't ship with
most distributions so pjproject attempts to build it's own internal
version.   Because we handle srtp in Asterisk itself not pjproject, we made
a change to the bundled version of pjproject to not build that internal
libsrtp.  If you look in third-party/pjproject/patches/config_site.h you
should see "#define PJMEDIA_HAS_SRTP 0".   It's possible that when the
bundled version is being built it's picking up something from the system
directories it shouldn't and looking for the internal version of libsrtp
which it didn't build.

If removing the system pjproject fixes your issue, let me know and I'll
open a ticket to look further at the conflict.











>
> Best regards
>
> [1] Original error message was localised.
>
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>
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>
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>



-- 
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Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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[asterisk-users] Recommended change to pjproject config_site.h

[George Joseph]

"*A problem was reported to the Asterisk project where transactions were
not **found when they should have been [1].  In the issue an incoming
INVITE transaction is CANCELed but the INVITE transaction cannot be found
so a 481 response is returned for the CANCEL.  The problematic calls have a
'_' **character in the Via branch parameter. *"
--Credit: Richard Mudgett

In the next Asterisk release, we have disabled PJ_HASH_USE_OWN_TOLOWER in
the bundled pjproject's config_site.h file.  If you build or package
pjproject yourself, we suggest you do the same.  We've also updated the
wiki[2] page.

/*
 * Do not ever enable PJ_HASH_USE_OWN_TOLOWER because the algorithm is
 * inconsistently used when calculating the hash value and doesn't
 * convert the same characters as pj_tolower()/tolower().  Thus you
 * can get different hash values if the string hashed has certain
 * characters in it.  (ASCII '@', '[', '\\', ']', '^', and '_')
 */
#undef PJ_HASH_USE_OWN_TOLOWER

[1] https://issues.asterisk.org/jira/browse/ASTERISK-26490
[2]
https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject

-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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Re: [asterisk-users] Opus codec in codecs.conf

[George Joseph]

On Tue, Oct 25, 2016 at 11:39 PM, Igor Goncharovsky <
igor.goncharov...@gmail.com> wrote:

> Hello,
>
> George, thank you for pointing this, but there is other question. It is
> not clear for some parameters names, what is possible values?
>
> For example this parameters:
> packet_loss
> max_bandwidth
> signal
> application
>
> Is there example of configured opus with full set of parameters?
>

I've opened an internal issue to update the sample codecs.conf file.  In
the mean time, you can simply add the parameter name to the end of the
'config show help' command as follows:

*CLI> config show help codec_opus opus max_bandwidth
[opus]
max_bandwidth = [] (Default: full) (Regex: False)

Encoder's maximum bandwidth allowed.

Sets an upper bandwidth bound on the encoder. Can be any of the following:
narrow
medium
wide
    super_wide
full






>
>
> 2016-10-25 18:42 GMT+06:00 George Joseph :
>
>>
>>
>> On Mon, Oct 24, 2016 at 6:54 PM, Igor Goncharovsky <
>> igor.goncharov...@gmail.com> wrote:
>>
>>> Hello,
>>>
>>> I am trying to configure new opus codec in asterisk 14, but unable to
>>> find any examples of codecs.conf settings for this codec.
>>>
>>> All I am trying to do - setup peer with using opus in narrow band mode
>>> (8kHz sampling rate). Does anybody know how to configure chan_opus?
>>>
>>>
>> If you run "config show help condec_opus opus" from teh Asterisk command
>> line, you'll get a list of the configuration options
>>
>> pbx1*CLI> config show help codec_opus opus
>> opus: [category !~ /.?/]
>>
>> Codec opus module for Asterisk options
>>
>> type  -- Must be of type 'opus'
>> sample_rate   -- Codec's sample rate.
>> packet_loss   -- Encoder's packet loss percentage.
>>
>> complexity-- Encoder's computational complexity.
>>
>> max_bandwidth -- Encoder's maximum bandwidth allowed.
>>
>> signal-- Encoder's signal type.
>> application   -- Encoder's application type.
>> max_playback_rate -- Encoder's maximum playback rate.
>>
>> max_ptime -- Encoder's maximum packetization rate.
>>
>> ptime -- Encoder's packetization rate.
>>
>> bitrate   -- Encoder's bit rate.
>> cbr   -- Encoder's constant bit rate value.
>> fec   -- Encoder's forward error correction value.
>> dtx   -- Encoder's discontinuous transmission value.
>>
>>
>>
>>> --
>>> Regards, Igor Goncharovsky
>>> Unistim Dev: http://unistim.igorg.ru
>>>
>>>
>>> --
>>> _____
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>>>   http://www.asterisk.org/community/astricon-user-conference
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>>
>> --
>> George Joseph
>> Digium, Inc. | Software Developer
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>> Check us out at: www.digium.com & www.asterisk.org
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>>   http://www.asterisk.org/community/astricon-user-conference
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Regards, Igor Goncharovsky
> Unistim Dev: http://unistim.igorg.ru
> Blog: http://igorg.ru
>
> --
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> Check out the new Aste

Re: [asterisk-users] Opus codec in codecs.conf

[George Joseph]

On Mon, Oct 24, 2016 at 6:54 PM, Igor Goncharovsky <
igor.goncharov...@gmail.com> wrote:

> Hello,
>
> I am trying to configure new opus codec in asterisk 14, but unable to find
> any examples of codecs.conf settings for this codec.
>
> All I am trying to do - setup peer with using opus in narrow band mode
> (8kHz sampling rate). Does anybody know how to configure chan_opus?
>
>
If you run "config show help condec_opus opus" from teh Asterisk command
line, you'll get a list of the configuration options

pbx1*CLI> config show help codec_opus opus
opus: [category !~ /.?/]

Codec opus module for Asterisk options

type  -- Must be of type 'opus'
sample_rate   -- Codec's sample rate.
packet_loss   -- Encoder's packet loss percentage.

complexity-- Encoder's computational complexity.

max_bandwidth -- Encoder's maximum bandwidth allowed.

signal-- Encoder's signal type.
application   -- Encoder's application type.
max_playback_rate -- Encoder's maximum playback rate.

max_ptime -- Encoder's maximum packetization rate.

ptime -- Encoder's packetization rate.
bitrate   -- Encoder's bit rate.
cbr   -- Encoder's constant bit rate value.
fec   -- Encoder's forward error correction value.
dtx   -- Encoder's discontinuous transmission value.



> --
> Regards, Igor Goncharovsky
> Unistim Dev: http://unistim.igorg.ru
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>   http://www.asterisk.org/community/astricon-user-conference
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
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Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] Asterisk 14.0.2 opens a high numbered UDP port

[George Joseph]

On Thu, Oct 13, 2016 at 12:35 PM, Brandon B.  wrote:

> What part of Asterisk 14.0.2 opens the random, high numbered (33094
> currently) UDP port? This port is opened even without any channel drivers
> loaded.
>
> $ sudo netstat -ltunp | grep asterisk
> udp0  0 0.0.0.0:51488 0.0.0.0:*
>  13830/asterisk
> udp0  0 0.0.0.0:5060 0.0.0.0:*
>  13830/asterisk
> udp0  0 :::42516 :::*
> 13830/asterisk
>
>
>
Those ports are used by the underlying pjproject DNS resolver.  The
resolver is always listening on those ports for DNS query responses.  1 for
IPV4 and 1 for IPV6.  Your firewall should only be allowing responses to
flow through to those ports that match outgoing requests.





>
> --
> _
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>
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>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] Finding the user agent of a channel using PJSIP?

[Anthony Joseph Messina]

On Monday, September 26, 2016 4:30:05 PM CDT John Kiniston wrote:
> I'm working on my sip to pjsip translation.
> 
> Right now I do some functionality based on what the user agent is on the
> calling phone using:
> 
> ${SIPPEER(${CHANNEL(peername)},useragent)}
> 
> I'm trying to replace it with
> 
> PJSIP_CONTACT(${CHANNEL(contact)},user-agent}) but I'm not getting any data
> returned when I query ${CHANNEL(contact)}
> 
> Is there a different function I should use to get my needed user agent of
> the active call?

How about ${PJSIP_HEADER(read,User-Agent)}

-- 
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F9B6 560E 68EA 037D 8C3D  D1C9 FF31 3BDB D9D8 99B6


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Re: [asterisk-users] Asterisk 13 externip

[George Joseph]

On Fri, Sep 16, 2016 at 7:29 AM,  wrote:

> Hi
>
> Can i put fromdomain in endpoint configuration?
>

Yes, from_domain goes in the endpoint or a template for an endpoint .


>
> Sent from my iPhone
>
> On Sep 16, 2016, at 18:51, George Joseph  wrote:
>
>
>
> On Fri, Sep 16, 2016 at 5:55 AM, Madushan Geethanga <
> mgliyanage...@gmail.com> wrote:
>
>> Hi,
>>
>> Tried with both softphone (Ekiga) and snom IP phone, contact header
>> contains the public IP. but from header contains the private IP.  after
>> OPTIONS method sent by the server. client sends an Register with expires 0.
>>
>
>
> Ok, did setting from_domain work?
>
>
>>
>> Best Regards,
>> Madushan
>>
>> On Thu, Sep 15, 2016 at 8:13 PM, George Joseph 
>> wrote:
>>
>>>
>>>
>>> On Thu, Sep 15, 2016 at 8:38 AM, Madushan Geethanga <
>>> mgliyanage...@gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> Thanks for the reply.
>>>>
>>>> Yes my PABX is on the cloud when I try to register to the server, the
>>>> server  sends registration OK with public address but  OPTION method
>>>> includes the private address of the server  in from header not the public
>>>> address. I have include both
>>>>
>>>> external_media_address=XX.XX.XX.XX
>>>> external_signaling_address=XX.XX.XX.XX
>>>> local_net=XX.XX.XX.XX
>>>>
>>>>
>>>> The client AOR is not getting registered.
>>>>
>>>
>>> What type of device/softphone is the client?
>>> Is the client trying to respond back to the address in the From header
>>> instead of the Contact header?
>>>
>>>
>>>>
>>>> Best Regards,
>>>> Madushan
>>>>
>>>> On Thu, Sep 15, 2016 at 7:41 PM, George Joseph 
>>>> wrote:
>>>>
>>>>>
>>>>>
>>>>> On Thu, Sep 15, 2016 at 7:17 AM, Faheem Muhammad >>>> > wrote:
>>>>>
>>>>>>
>>>>>>
>>>>>> On Wednesday, 14 September 2016, Madushan Geethanga <
>>>>>> mgliyanage...@gmail.com> wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> What is the equal option for externip in asterisk 13 with pjsip. I
>>>>>>> have tried
>>>>>>>
>>>>>>> external_media_address=XX.XX.XX.XX
>>>>>>> external_signaling_address=XX.XX.XX.XX
>>>>>>>
>>>>>>> but asterisk 13 writes local ip to the from header. any suggestions?
>>>>>>>
>>>>>>
>>>>> Setting 'from_domain' on the endpoint will do it.  Are you having
>>>>> issues with an internal address being used in the "From" header?
>>>>>
>>>>>
>>>>>
>>>>>>
>>>>>>> Best Regards,
>>>>>>> Madushan
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> --
>>>>>> Sent from Gmail Mobile
>>>>>>
>>>>>> --
>>>>>> _
>>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>>>
>>>>>> Join the Asterisk Community at the 13th AstriCon, September 27-29,
>>>>>> 2016
>>>>>>   http://www.asterisk.org/community/astricon-user-conference
>>>>>>
>>>>>> New to Asterisk? Start here:
>>>>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>>>>
>>>>>> asterisk-users mailing list
>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> George Joseph
>>>>> Digium, Inc. | Software Developer
>>>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>>>>> Check us out at: www.digium.com & www.asterisk.org
>>>>>
>>>>>
>>>>> --
>>>>> _
>>>>&g

Re: [asterisk-users] Asterisk 13 externip

[George Joseph]

On Fri, Sep 16, 2016 at 5:55 AM, Madushan Geethanga  wrote:

> Hi,
>
> Tried with both softphone (Ekiga) and snom IP phone, contact header
> contains the public IP. but from header contains the private IP.  after
> OPTIONS method sent by the server. client sends an Register with expires 0.
>


Ok, did setting from_domain work?


>
> Best Regards,
> Madushan
>
> On Thu, Sep 15, 2016 at 8:13 PM, George Joseph  wrote:
>
>>
>>
>> On Thu, Sep 15, 2016 at 8:38 AM, Madushan Geethanga <
>> mgliyanage...@gmail.com> wrote:
>>
>>> Hi,
>>>
>>> Thanks for the reply.
>>>
>>> Yes my PABX is on the cloud when I try to register to the server, the
>>> server  sends registration OK with public address but  OPTION method
>>> includes the private address of the server  in from header not the public
>>> address. I have include both
>>>
>>> external_media_address=XX.XX.XX.XX
>>> external_signaling_address=XX.XX.XX.XX
>>> local_net=XX.XX.XX.XX
>>>
>>>
>>> The client AOR is not getting registered.
>>>
>>
>> What type of device/softphone is the client?
>> Is the client trying to respond back to the address in the From header
>> instead of the Contact header?
>>
>>
>>>
>>> Best Regards,
>>> Madushan
>>>
>>> On Thu, Sep 15, 2016 at 7:41 PM, George Joseph 
>>> wrote:
>>>
>>>>
>>>>
>>>> On Thu, Sep 15, 2016 at 7:17 AM, Faheem Muhammad 
>>>> wrote:
>>>>
>>>>>
>>>>>
>>>>> On Wednesday, 14 September 2016, Madushan Geethanga <
>>>>> mgliyanage...@gmail.com> wrote:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> What is the equal option for externip in asterisk 13 with pjsip. I
>>>>>> have tried
>>>>>>
>>>>>> external_media_address=XX.XX.XX.XX
>>>>>> external_signaling_address=XX.XX.XX.XX
>>>>>>
>>>>>> but asterisk 13 writes local ip to the from header. any suggestions?
>>>>>>
>>>>>
>>>> Setting 'from_domain' on the endpoint will do it.  Are you having
>>>> issues with an internal address being used in the "From" header?
>>>>
>>>>
>>>>
>>>>>
>>>>>> Best Regards,
>>>>>> Madushan
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>> --
>>>>> Sent from Gmail Mobile
>>>>>
>>>>> --
>>>>> _
>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>>
>>>>> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>>>>>   http://www.asterisk.org/community/astricon-user-conference
>>>>>
>>>>> New to Asterisk? Start here:
>>>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>>>
>>>>> asterisk-users mailing list
>>>>> To UNSUBSCRIBE or update options visit:
>>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> George Joseph
>>>> Digium, Inc. | Software Developer
>>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>>>> Check us out at: www.digium.com & www.asterisk.org
>>>>
>>>>
>>>> --
>>>> _
>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>
>>>> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>>>>   http://www.asterisk.org/community/astricon-user-conference
>>>>
>>>> New to Asterisk? Start here:
>>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>>
>>>> asterisk-users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] Asterisk 13 externip

[George Joseph]

On Thu, Sep 15, 2016 at 7:17 AM, Faheem Muhammad 
wrote:

>
>
> On Wednesday, 14 September 2016, Madushan Geethanga <
> mgliyanage...@gmail.com> wrote:
>
>> Hi,
>>
>> What is the equal option for externip in asterisk 13 with pjsip. I have
>> tried
>>
>> external_media_address=XX.XX.XX.XX
>> external_signaling_address=XX.XX.XX.XX
>>
>> but asterisk 13 writes local ip to the from header. any suggestions?
>>
>
Setting 'from_domain' on the endpoint will do it.  Are you having issues
with an internal address being used in the "From" header?



>
>> Best Regards,
>> Madushan
>>
>>
>>
>
> --
> Sent from Gmail Mobile
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>   http://www.asterisk.org/community/astricon-user-conference
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users