Re: [asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.

[Pavel]

23.04.2019 0:27, Joshua C. Colp wrote:

On Mon, Apr 22, 2019, at 2:10 PM, Pavel wrote:




Tried already.

"line" is good, but not perfect.

Every time I restart asterisk, it will generate new random string for ";line=".

So, every time I restart asterisk, registrar (Server1) will save one
more contact in it's database.

Some will remove obsolete contacts, but some will not.

For example, FreePBX will not remove obsolete contacts, if max_contacts
specified (FreePBX will set rewrite_contact=no in this case).

So, after a number of Asterisk restarts, FreePBX will reject new
registrations, as max_contacts is reached.

It should specify remove_existing to remove old ones to make room for the new 
ones. That would be a FreePBX thing, though.


FreePBX is an example, where it can be a critical problem.

3cx will work, but if you will restart asterisk 10 times - you will see 
10 times more contacts in 3cx.


When you will make call from 3cx - it will make 10 calls (10 contacts), 
untill they will obsolete...




Unfortunately, "line" does not save random between restarts.

It's also unable to specify "random" value in pjsip.conf.


I'm thinking to patch res_pjsip_outbound_registration to add this feature.

Am I wrong and there is another way ?

I don't see any reason why this couldn't be an option.


For flexibility.

Not to register new fake contacts in peer PBX.


It's also a security hole, as anybody can generate INVITE with
";line=random" from any IP address !

You can use an ACL to limit the endpoint to certain source IP addresses.


5+ !

Thank you, ACL is a good idea !



res_pjsip_outbound_registration will only match "line", but will not
take care about source IP, ...



Is there any more clear way to identify incoming INVITE/OPTIONS packets ?

Not very familliar with SIP, not sure, how should it be done.

There is no real defined mechanism within SIP to do this. Phones employ 
different mechanisms to differentiate. Some may use a similar mechanism to the 
line option. Some run multiple SIP transports on different ports for each 
account so they can differentiate based on where it came in on. Some look at 
the request URI coming in. Some just don't care.

Sniffered some time ago how it's done in phonerlite, jitsi, linksys, ...

Some use different port, some use ";rinstance=", the same like ";line=" 
in asterisk.


Was not sure it's a right way to go.


I will probably extend "line" a bit to specify it's value in pjsip.conf .

It will be less than 10 lines of code.


Thank you very much !

Your help will simplify my life a lot :-)



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Re: [asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.

[Joshua C. Colp]

On Mon, Apr 22, 2019, at 2:10 PM, Pavel wrote:



> Tried already.
> 
> "line" is good, but not perfect.
> 
> Every time I restart asterisk, it will generate new random string for 
> ";line=".
> 
> So, every time I restart asterisk, registrar (Server1) will save one 
> more contact in it's database.
> 
> Some will remove obsolete contacts, but some will not.
> 
> For example, FreePBX will not remove obsolete contacts, if max_contacts 
> specified (FreePBX will set rewrite_contact=no in this case).
> 
> So, after a number of Asterisk restarts, FreePBX will reject new 
> registrations, as max_contacts is reached.

It should specify remove_existing to remove old ones to make room for the new 
ones. That would be a FreePBX thing, though.
 
> Unfortunately, "line" does not save random between restarts.
> 
> It's also unable to specify "random" value in pjsip.conf.
> 
> 
> I'm thinking to patch res_pjsip_outbound_registration to add this feature.
> 
> Am I wrong and there is another way ?

I don't see any reason why this couldn't be an option.
 
> 
> It's also a security hole, as anybody can generate INVITE with 
> ";line=random" from any IP address !

You can use an ACL to limit the endpoint to certain source IP addresses.

> 
> res_pjsip_outbound_registration will only match "line", but will not 
> take care about source IP, ...
> 
> 
> 
> Is there any more clear way to identify incoming INVITE/OPTIONS packets ?
> 
> Not very familliar with SIP, not sure, how should it be done.

There is no real defined mechanism within SIP to do this. Phones employ 
different mechanisms to differentiate. Some may use a similar mechanism to the 
line option. Some run multiple SIP transports on different ports for each 
account so they can differentiate based on where it came in on. Some look at 
the request URI coming in. Some just don't care.

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior 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] Incoming SIP call, outgoing SIP registration. PJSIP.

[Pavel]

Hi,

Thank for your answer.

22.04.2019 23:47, Joshua C. Colp пишет:

On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote:

Hi,

Got problems with incoming SIP calls.

Scenario:

Server1: 3cx or any other server

Server2: Asterisk 16.2.1 . PJPROJECT 2.8

Server2 registers on Server1 with SIP ID 1121.

Registration is OK.

Server2 outgoing calls are OK.

INVITE, unauthorized, INVITE with password, OK, RINGING,...

Troubles with incoming calls / incoming INVITE's .

I can not identify endpoint by IP, I have multiple registrations on the
same Server1.

As far as I understood, res_pjsip_endpoint_identifier_user match
endpoint by "From" header, so it will not match also.

match_headers also seems useless (not able to match "INVITE" string,
just headers like "TO:").

Is there any way to match incoming INVITE calls ? (also OPTIONS, NOTIFY,
... packets)

It should be a typical scenario, but it does not work...

Is there any way to make it working ?

Outbound registration provides the line option[1] which can be used to 
differentiate traffic in regards to different outbound registrations. It 
requires the remote server to adhere to the SIP RFC and report back some data 
we give in our Contact, so you have to test it and see if it works.

[1] 
https://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/


Tried already.

"line" is good, but not perfect.

Every time I restart asterisk, it will generate new random string for 
";line=".


So, every time I restart asterisk, registrar (Server1) will save one 
more contact in it's database.


Some will remove obsolete contacts, but some will not.

For example, FreePBX will not remove obsolete contacts, if max_contacts 
specified (FreePBX will set rewrite_contact=no in this case).


So, after a number of Asterisk restarts, FreePBX will reject new 
registrations, as max_contacts is reached.


Unfortunately, "line" does not save random between restarts.

It's also unable to specify "random" value in pjsip.conf.


I'm thinking to patch res_pjsip_outbound_registration to add this feature.

Am I wrong and  there is another way ?

It's also a security hole, as anybody can generate INVITE with 
";line=random" from any IP address !


res_pjsip_outbound_registration will only match "line", but will not 
take care about source IP, ...



Is there any more clear way to identify incoming INVITE/OPTIONS packets ?

Not very familliar with SIP, not sure, how should it be done.

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Re: [asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.

[Joshua C. Colp]

On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote:
> Hi,
> 
> Got problems with incoming SIP calls.
> 
> Scenario:
> 
> Server1: 3cx or any other server
> 
> Server2: Asterisk 16.2.1 . PJPROJECT 2.8
> 
> Server2 registers on Server1 with SIP ID 1121.
> 
> Registration is OK.
> 
> Server2 outgoing calls are OK.
> 
> INVITE, unauthorized, INVITE with password, OK, RINGING,...
> 
> Troubles with incoming calls / incoming INVITE's .
> 
> I can not identify endpoint by IP, I have multiple registrations on the 
> same Server1.
> 
> As far as I understood, res_pjsip_endpoint_identifier_user match 
> endpoint by "From" header, so it will not match also.
> 
> match_headers also seems useless (not able to match "INVITE" string, 
> just headers like "TO:").
> 
> Is there any way to match incoming INVITE calls ? (also OPTIONS, NOTIFY, 
> ... packets)
> 
> It should be a typical scenario, but it does not work...
> 
> Is there any way to make it working ?

Outbound registration provides the line option[1] which can be used to 
differentiate traffic in regards to different outbound registrations. It 
requires the remote server to adhere to the SIP RFC and report back some data 
we give in our Contact, so you have to test it and see if it works.

[1] 
https://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior 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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[asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.

[Pavel]

Hi,

Got problems with incoming SIP calls.

Scenario:

Server1: 3cx or any other server

Server2: Asterisk 16.2.1 . PJPROJECT 2.8

Server2 registers on Server1 with SIP ID 1121.

Registration is OK.

Server2 outgoing calls are OK.

INVITE, unauthorized, INVITE with password, OK, RINGING,...

Troubles with incoming calls / incoming INVITE's .

I can not identify endpoint by IP, I have multiple registrations on the 
same Server1.


As far as I understood, res_pjsip_endpoint_identifier_user match 
endpoint by "From" header, so it will not match also.


match_headers also seems useless (not able to match "INVITE" string, 
just headers like "TO:").


Is there any way to match incoming INVITE calls ? (also OPTIONS, NOTIFY, 
... packets)


It should be a typical scenario, but it does not work...

Is there any way to make it working ?


[0.0.0.0-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060

[endpoint0](!)
type=endpoint
transport=0.0.0.0-udp
disallow=all
allow=alaw
allow=ulaw
t38_udptl=no
t38_udptl_ec=none
fax_detect=no
t38_udptl_nat=no
dtmf_mode=auto
direct_media=yes
from_domain=172.16.25.23
timers_sess_expires=1800
tone_zone=ru
language=ru
rewrite_contact=yes
rtp_symmetric=yes
force_rport=yes

[registration0](!)
type=registration
transport=0.0.0.0-udp
retry_interval=60
max_retries=10
expiration=3600
auth_rejection_permanent=yes
server_uri=sip:172.16.25.23


[fxs17](endpoint0)
context=from-sip-fxs
aors=fxs17
outbound_auth=fxs17
from_user=1121
set_var=DAHDICHAN=17

[fxs17]
type=aor
qualify_frequency=60
contact=sip:1121@172.16.25.23

[fxs17]
type=auth
auth_type=userpass
password=11
username=1121

[fxs17](registration0)
outbound_auth=fxs17
client_uri=sip:1121@172.16.25.23
contact_user=fxs17
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Re: [asterisk-users] incoming call label

[Julian Beach]

Hello Thelma,

Friday, February 16, 2018, 2:16:02 AM, you wrote:

> Contact: "sip:pstn-"

> And it found in sip.conf only:
> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060

> Is perhaps the name effected by the special character "-" (dash) that is
> why it only matches "pstn" and take the first one it found.  Will it
> make a difference if I rename the port to pstn_ in configuration files.

If the type=friend then it matches on IP Address and Port Number, not
the user name. It will then use the first entry in the sip.conf - it
does not take any notice of the name. If you change the order that the
two entries appear, all the calls will appear to come from [pstn-9998]
even if they come from [pstn-].

I  used  to  set user=peer, which solved the problem for me, but I now
direct  all  the calls to a single context in extensions.conf and then
send them to their own contexts based on the DNID.


-- 
Best regards,
 Julianmailto:jb_s...@trink.co.uk


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Re: [asterisk-users] incoming call label

[Jean Aunis]

Le 16/02/2018 à 05:30, the...@sys-concept.com a écrit :

On 02/15/2018 04:49 PM, Joshua Colp wrote:

On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:




Thanks again for the hint.
Here is the output from asterisk.

The call is coming on Audocodes gateway from: pstn-

But asterisk display:
Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060

Why not loolking up "pstn-" in sip.conf?

It found pstn- using 10.10.0.8:5060 - if the request always comes from the 
same IP address and port it has no other way built in to differentiate between 
the two except by matching based on username in the 'From' header.

It didn't find "pstn- using 10.10.0.8:5060"
The call came IN from PSTN line on audiocodes equipment to FXO port that
is labelled "pstn-"  so asterisk reported as such.
And I think asterisk suppose to lookup this label in sip.conf to the
registered entry but instead selected pstn-9998 entry; I don't know why.

If the call came IN on pstn-
and sip.conf has two entries:
[pstn-]
[pstn-9998]

Why it can not distinguish between the two of them correctly?

--
Thelma


If your device supports SIP authentication, you can try to turn on the 
"match_auth_username" parameter in sip.conf. It is said to be 
experimental but has always worked well for me.



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Re: [asterisk-users] incoming call label

[thelma]

On 02/15/2018 04:49 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:
> 
> 
> 
>>
>> Thanks again for the hint.
>> Here is the output from asterisk.
>>
>> The call is coming on Audocodes gateway from: pstn-
>>
>> But asterisk display:
>> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060
>>
>> Why not loolking up "pstn-" in sip.conf?
> 
> It found pstn- using 10.10.0.8:5060 - if the request always comes from 
> the same IP address and port it has no other way built in to differentiate 
> between the two except by matching based on username in the 'From' header.

It didn't find "pstn- using 10.10.0.8:5060"
The call came IN from PSTN line on audiocodes equipment to FXO port that
is labelled "pstn-"  so asterisk reported as such.
And I think asterisk suppose to lookup this label in sip.conf to the
registered entry but instead selected pstn-9998 entry; I don't know why.

If the call came IN on pstn-
and sip.conf has two entries:
[pstn-]
[pstn-9998]

Why it can not distinguish between the two of them correctly?

--
Thelma


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Re: [asterisk-users] incoming call label

[thelma]

Thelma
On 02/15/2018 07:16 PM, the...@sys-concept.com wrote:
> 
> On 02/15/2018 04:49 PM, Joshua Colp wrote:
>> On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:
>>
>> 
>>
>>>
>>> Thanks again for the hint.
>>> Here is the output from asterisk.
>>>
>>> The call is coming on Audocodes gateway from: pstn-
>>>
>>> But asterisk display:
>>> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060
>>>
>>> Why not loolking up "pstn-" in sip.conf?
>>
>> It found pstn- using 10.10.0.8:5060 - if the request always comes from 
>> the same IP address and port it has no other way built in to differentiate 
>> between the two except by matching based on username in the 'From' header.
>>
> 
> Call comes from same IP address always.
> To comes  form Audiocode:
> 
> <--- SIP read from UDP:10.10.0.8:5060 --->
> INVITE sip:4@10.10.0.4 SIP/2.0
> Via: SIP/2.0/UDP 10.10.0.8;branch=z9hG4bKac766808844
> Max-Forwards: 70
> From: "Z" ;tag=1c766802762
> To: 
> Call-ID: 7668022781522018162620@10.10.0.8
> CSeq: 1 INVITE
> Contact: 
> 
> Contact: "sip:pstn-"
> 
> And it found in sip.conf only:
> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060
> 
> Is perhaps the name effected by the special character "-" (dash) that is
> why it only matches "pstn" and take the first one it found.  Will it
> make a difference if I rename the port to pstn_ in configuration files.
> 
> --
> Thelma
 
sip show peers
Name/username HostDyn 
Forcerport ComediaACL Port Status  Description  
pstn-/voice-  10.10.0.8D  No
 No 5060 Unmonitored  
pstn-9998/fax-999810.10.0.8D  No
 No 5060 Unmonitored


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Re: [asterisk-users] incoming call label

[thelma]

On 02/15/2018 04:49 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:
> 
> 
> 
>>
>> Thanks again for the hint.
>> Here is the output from asterisk.
>>
>> The call is coming on Audocodes gateway from: pstn-
>>
>> But asterisk display:
>> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060
>>
>> Why not loolking up "pstn-" in sip.conf?
> 
> It found pstn- using 10.10.0.8:5060 - if the request always comes from 
> the same IP address and port it has no other way built in to differentiate 
> between the two except by matching based on username in the 'From' header.
> 

Call comes from same IP address always.
To comes  form Audiocode:

<--- SIP read from UDP:10.10.0.8:5060 --->
INVITE sip:4@10.10.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.8;branch=z9hG4bKac766808844
Max-Forwards: 70
From: "Z" ;tag=1c766802762
To: 
Call-ID: 7668022781522018162620@10.10.0.8
CSeq: 1 INVITE
Contact: 

Contact: "sip:pstn-"

And it found in sip.conf only:
Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060

Is perhaps the name effected by the special character "-" (dash) that is
why it only matches "pstn" and take the first one it found.  Will it
make a difference if I rename the port to pstn_ in configuration files.

--
Thelma

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Re: [asterisk-users] incoming call label

[Joshua Colp]

On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:



> 
> Thanks again for the hint.
> Here is the output from asterisk.
> 
> The call is coming on Audocodes gateway from: pstn-
> 
> But asterisk display:
> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060
> 
> Why not loolking up "pstn-" in sip.conf?

It found pstn- using 10.10.0.8:5060 - if the request always comes from the 
same IP address and port it has no other way built in to differentiate between 
the two except by matching based on username in the 'From' header.

-- 
Joshua Colp
Digium, Inc. | Senior 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] incoming call label

[thelma]

On 02/15/2018 04:08 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 7:03 PM, the...@sys-concept.com wrote:
>> On 02/15/2018 03:44 PM, Joshua Colp wrote:
>>> On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
 I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports

 IN audocodes setting I have:
 "EndPoint Phone Number"

 Channel: 3phone number: pstn-
 Channel: 4phone number: pstn-9998

 When I am calling " pstn-" the port number "Channel:3" lights up but
 asterisk is showing that the call is coming on "pstn-9998"

 -- Executing . Answer("SIP/pstn-9998

 Asterisk should be showing "pstn-" (not pstn-9998)
 Where is this label coming from?
>>>
>>> It is from the SIP entry in sip.conf that it was matched against.
>>>
>>
>> Thanks for the input.
>>
>> In sip.conf I have relevant entries.
>>
>> [pstn-] ; incoming/outgoing calls on FXO port
>> type=friend
>> secret=spa354
>> username=voice-
>> mailbox=622 ; just for audiocodes error complain
>> host=dynamic
>> canreinvite=no ; (dtmf not wroking correctly without this one)
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> nat=no
>> context=incoming
>> callgroup=1
>> pickupgroup=1
>> insecure=invite
>>
>> [pstn-9998]
>> type=friend
>> secret=158567
>> username=fax-9998
>> insecure=invite
>> mailbox=622  ; just for audiocodes error complain
>> host=dynamic
>> canreinvite=no  ; (dtmf not wroking correctly without this one)
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> nat=no
>> context=incoming
>> callgroup=1
>> pickupgroup=
>>
>> My asterisk registration is correct as well:
>> sip show users
>> Username   Secret   Accountcode  Def.Context
>>  ACL  Forcerport
>> pstn-9998  158567   incoming
>> No   No
>> pstn-  spa354 incoming
>>   No   No
>>
>> Caller display ID from PSTN on FXO ports are working OK.
>> The [pstn-]  is channel: 4
>> The [pstn-9998] is channel: 3
>>
>> If the call on Audocode is lighting UP "channel:3" the sip.conf should
>> associate that call with  [pstn-] (and not [pstn-9998])
> 
> Not necessarily. You appear to be doing IP+port based matching. If requests 
> always come from the same source IP address and port, then it would match 
> only one. Turning on sip debug using "sip set debug on" and verbosity using 
> "core set debug 9" would give you more information about each packet 
> (including where it is from) and what was actually matched based on it.

Thanks again for the hint.
Here is the output from asterisk.

The call is coming on Audocodes gateway from: pstn-

But asterisk display:
Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060

Why not loolking up "pstn-" in sip.conf?

<--- SIP read from UDP:10.10.0.8:5060 --->
INVITE sip:4@10.10.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.8;branch=z9hG4bKac766808844
Max-Forwards: 70
From: "Z" ;tag=1c766802762
To: 
Call-ID: 7668022781522018162620@10.10.0.8
CSeq: 1 INVITE
Contact: 
Supported:
em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 249

v=0
o=AudiocodesGW 766797875 766797759 IN IP4 10.10.0.8
s=Phone-Call
c=IN IP4 10.10.0.8
t=0 0
m=audio 6000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<->
--- (14 headers 12 lines) ---
Sending to 10.10.0.8:5060 (no NAT)
Sending to 10.10.0.8:5060 (no NAT)
Using INVITE request as basis request - 7668022781522018162620@10.10.0.8
Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer -
audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.10.0.8:6000
Looking for 4 in incoming (domain 10.10.0.4)
list_route: hop: 

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Thelma

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Re: [asterisk-users] incoming call label

[Joshua Colp]

On Thu, Feb 15, 2018, at 7:03 PM, the...@sys-concept.com wrote:
> On 02/15/2018 03:44 PM, Joshua Colp wrote:
> > On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
> >> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
> >>
> >> IN audocodes setting I have:
> >> "EndPoint Phone Number"
> >>
> >> Channel: 3phone number: pstn-
> >> Channel: 4phone number: pstn-9998
> >>
> >> When I am calling " pstn-" the port number "Channel:3" lights up but
> >> asterisk is showing that the call is coming on "pstn-9998"
> >>
> >> -- Executing . Answer("SIP/pstn-9998
> >>
> >> Asterisk should be showing "pstn-" (not pstn-9998)
> >> Where is this label coming from?
> > 
> > It is from the SIP entry in sip.conf that it was matched against.
> > 
> 
> Thanks for the input.
> 
> In sip.conf I have relevant entries.
> 
> [pstn-] ; incoming/outgoing calls on FXO port
> type=friend
> secret=spa354
> username=voice-
> mailbox=622 ; just for audiocodes error complain
> host=dynamic
> canreinvite=no ; (dtmf not wroking correctly without this one)
> disallow=all
> allow=ulaw
> allow=alaw
> nat=no
> context=incoming
> callgroup=1
> pickupgroup=1
> insecure=invite
> 
> [pstn-9998]
> type=friend
> secret=158567
> username=fax-9998
> insecure=invite
> mailbox=622  ; just for audiocodes error complain
> host=dynamic
> canreinvite=no  ; (dtmf not wroking correctly without this one)
> disallow=all
> allow=ulaw
> allow=alaw
> nat=no
> context=incoming
> callgroup=1
> pickupgroup=
> 
> My asterisk registration is correct as well:
> sip show users
> Username   Secret   Accountcode  Def.Context
>  ACL  Forcerport
> pstn-9998  158567   incoming
> No   No
> pstn-  spa354 incoming
>   No   No
> 
> Caller display ID from PSTN on FXO ports are working OK.
> The [pstn-]  is channel: 4
> The [pstn-9998] is channel: 3
> 
> If the call on Audocode is lighting UP "channel:3" the sip.conf should
> associate that call with  [pstn-] (and not [pstn-9998])

Not necessarily. You appear to be doing IP+port based matching. If requests 
always come from the same source IP address and port, then it would match only 
one. Turning on sip debug using "sip set debug on" and verbosity using "core 
set debug 9" would give you more information about each packet (including where 
it is from) and what was actually matched based on it.

-- 
Joshua Colp
Digium, Inc. | Senior 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] incoming call label

[thelma]

On 02/15/2018 03:44 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
>> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
>>
>> IN audocodes setting I have:
>> "EndPoint Phone Number"
>>
>> Channel: 3phone number: pstn-
>> Channel: 4phone number: pstn-9998
>>
>> When I am calling " pstn-" the port number "Channel:3" lights up but
>> asterisk is showing that the call is coming on "pstn-9998"
>>
>> -- Executing . Answer("SIP/pstn-9998
>>
>> Asterisk should be showing "pstn-" (not pstn-9998)
>> Where is this label coming from?
> 
> It is from the SIP entry in sip.conf that it was matched against.
> 

Thanks for the input.

In sip.conf I have relevant entries.

[pstn-] ; incoming/outgoing calls on FXO port
type=friend
secret=spa354
username=voice-
mailbox=622 ; just for audiocodes error complain
host=dynamic
canreinvite=no ; (dtmf not wroking correctly without this one)
disallow=all
allow=ulaw
allow=alaw
nat=no
context=incoming
callgroup=1
pickupgroup=1
insecure=invite

[pstn-9998]
type=friend
secret=158567
username=fax-9998
insecure=invite
mailbox=622  ; just for audiocodes error complain
host=dynamic
canreinvite=no  ; (dtmf not wroking correctly without this one)
disallow=all
allow=ulaw
allow=alaw
nat=no
context=incoming
callgroup=1
pickupgroup=

My asterisk registration is correct as well:
sip show users
Username   Secret   Accountcode  Def.Context
 ACL  Forcerport
pstn-9998  158567   incoming
No   No
pstn-  spa354 incoming
  No   No

Caller display ID from PSTN on FXO ports are working OK.
The [pstn-]  is channel: 4
The [pstn-9998] is channel: 3

If the call on Audocode is lighting UP "channel:3" the sip.conf should
associate that call with  [pstn-] (and not [pstn-9998])

--
Thelma

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Re: [asterisk-users] incoming call label

[Joshua Colp]

On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
> 
> IN audocodes setting I have:
> "EndPoint Phone Number"
> 
> Channel: 3phone number: pstn-
> Channel: 4phone number: pstn-9998
> 
> When I am calling " pstn-" the port number "Channel:3" lights up but
> asterisk is showing that the call is coming on "pstn-9998"
> 
> -- Executing . Answer("SIP/pstn-9998
> 
> Asterisk should be showing "pstn-" (not pstn-9998)
> Where is this label coming from?

It is from the SIP entry in sip.conf that it was matched against.

-- 
Joshua Colp
Digium, Inc. | Senior 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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[asterisk-users] incoming call label

[thelma]

I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports

IN audocodes setting I have:
"EndPoint Phone Number"

Channel: 3phone number: pstn-
Channel: 4phone number: pstn-9998

When I am calling " pstn-" the port number "Channel:3" lights up but
asterisk is showing that the call is coming on "pstn-9998"

-- Executing . Answer("SIP/pstn-9998

Asterisk should be showing "pstn-" (not pstn-9998)
Where is this label coming from?

-- 
Thelma

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Re: [asterisk-users] Incoming Call by DID

[A J Stiles]

On Wednesday 26 Oct 2016, KyD wrote:
> Hi,
> 
> My sip provider gave me 2 numbers for the incoming call via pstn.
> 
> nro1 = 12341234
> nro2 = 45674567
> 
> I have a dialplan for each.
> if i put this on my dialplan:
> 
> exten => s,1,Dial(SIP/1001)
> exten => Hangup()
> 
> Works!
> 
> But if i put one of them:
> 
> exten => 12341234,1,Dial(SIP/1001)
> exten => _1234,1,Dial(SIP/1001)
> exten => 45674567,1,Dial(SIP/1001)
> exten => _4567,1,Dial(SIP/1001)
> 
> incoming calls do not arrive.
> 
> Any ideas?

The incoming call must be arriving with ${EXTEN} containing something that 
doesn't match  12341234, _1234, 45674567 or _4567, so it is 
not triggering any of the extensions in your dialplan.  Maybe it still has the 
STD code or even the IDD code prepended.  (Been caught this way once before 
.  our old ISDN-30 provider used to send just the local number, then we 
moved to a new ISDN-30 provider who send the number with STD code but no 
initial 0.  Cue frantic editing of dialplan before rest of staff arrived .)
 
So try this;
 
exten => s,1,NoOp(Incoming call for '${EXTEN}')
exten => s,n,Dial(SIP/1001)
exten => s,n,Hangup()
 
Run `# asterisk -vvvr`, dial one of your DDI numbers from 
a mobile phone and watch the messages scrolling past.
 
Now you will be seeing exactly what ${EXTEN} contains when a call comes in, so 
you should be able to work out what is going on, and craft your extension 
expressions to suit.  If in doubt, post an excerpt from your CLI output.

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] Incoming Call by DID

[David Duffett]

It seems like your SIP provider is not sending and DID information, or that
the information is not being sent in the same format you are using in your
dialplan.

You can check this by looking at the SIP debug information for the inbound
calls and/or by checking with your SIP provider (that they are sending the
DID number and what format it is in).

All the best,
David

On 27 Oct 2016 5:21 am, "KyD"  wrote:

Hi,

My sip provider gave me 2 numbers for the incoming call via pstn.

nro1 = 12341234
nro2 = 45674567

I have a dialplan for each.
if i put this on my dialplan:

exten => s,1,Dial(SIP/1001)
exten => Hangup()

Works!

But if i put one of them:

exten => 12341234,1,Dial(SIP/1001)
exten => _1234,1,Dial(SIP/1001)
exten => 45674567,1,Dial(SIP/1001)
exten => _4567,1,Dial(SIP/1001)

incoming calls do not arrive.

Any ideas?
--
KyD
GNU/Linux SysAdmin
Quanto mais você sabe, mais você percebe que você não sabe nada.

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[asterisk-users] Incoming Call by DID

[KyD]

Hi,

My sip provider gave me 2 numbers for the incoming call via pstn.

nro1 = 12341234
nro2 = 45674567

I have a dialplan for each.
if i put this on my dialplan:

exten => s,1,Dial(SIP/1001)
exten => Hangup()

Works!

But if i put one of them:

exten => 12341234,1,Dial(SIP/1001)
exten => _1234,1,Dial(SIP/1001)
exten => 45674567,1,Dial(SIP/1001)
exten => _4567,1,Dial(SIP/1001)

incoming calls do not arrive.

Any ideas?
-- 
KyD
GNU/Linux SysAdmin
Quanto mais você sabe, mais você percebe que você não sabe nada.

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Re: [asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate

[Julian Beach]

Hello Phil,

On Saturday, April 23, 2016, 11:11:29 PM, you wrote:

> Actually, this is now sorted. It turns out the latest recommended
> configs on the A wiki had peer vs. user confusion. On correcting
> this, all was well.

I'm glad you found it. It look me a while to track down that problem
when I had it.

The one that was hardest for me to track down was a slight mis-match
between the RTP ports in Asterisk and the corresponding ports open on
a firewall, which resulted in about 1 in 10 calls having no audio!
Doh!

-- 
Best regards,
 Julianmailto:jb_s...@trink.co.uk


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Re: [asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate

[Phil Reynolds]

On Sat, 23 Apr 2016 22:45:32 +0100
Julian Beach  wrote:

> Hello Phil,
> 
> I have a couple of lines with A, and I have not been having any
> problems recently. When I have had similar problems in the past, it
> has been an issue with the SIP config. I originally had a number of
> contexts set up in sip.conf to handle the lines coming in (such as
> [aa-line1], [aa-line2]) each with their own username and password
> settings. The type=user setting was critical, because all the calls
> came from the same IP address, and using type=peer caused matching
> problems which resulted in authentication failures. This got too
> complex to manage once I added in all the IP addresses A calls might
> come in from. so I simplified the setup.
> 
> I now have just one context in sip.conf to handle incoming A calls,
> with the same username for all lines, and type=peer. Calls are then
> sent to extensions.conf, where the calls are directed to the correct
> call-handler for the line based on the CID. Here is the setup in
> sip.conf for A calls:

Actually, this is now sorted. It turns out the latest recommended
configs on the A wiki had peer vs. user confusion. On correcting
this, all was well.

-- 
Phil Reynolds
mail: phil-aster...@tinsleyviaduct.com
Web: http://phil.tinsleyviaduct.com/

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Re: [asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate

[Julian Beach]

Hello Phil,

On Saturday, April 23, 2016, 12:19:15 PM, you wrote:

> I have checked that the username and password in my config agree both
> ends, and have even tried changing them.

> The bulk of my calls come in on A, so I am obviously trying to find
> out what has gone wrong. No-one else is seeing any problem. What do I
> need to do to track this down?

I have a couple of lines with A, and I have not been having any
problems recently. When I have had similar problems in the past, it
has been an issue with the SIP config. I originally had a number of
contexts set up in sip.conf to handle the lines coming in (such as
[aa-line1], [aa-line2]) each with their own username and password
settings. The type=user setting was critical, because all the calls
came from the same IP address, and using type=peer caused matching
problems which resulted in authentication failures. This got too
complex to manage once I added in all the IP addresses A calls might
come in from. so I simplified the setup.

I now have just one context in sip.conf to handle incoming A calls,
with the same username for all lines, and type=peer. Calls are then
sent to extensions.conf, where the calls are directed to the correct
call-handler for the line based on the CID. Here is the setup in
sip.conf for A calls:

---
sip.conf

[aa-incoming](!)
type=peer
context=aa-incoming
insecure=invite
transport=udp
disallow=all
allow=alaw
trustrpid=yes
sendrpid=yes

; IPv4 hostnames
[voiceless-1](aa-incoming)
host=a4.voiceless.aa.net.uk
[voiceless-2](aa-incoming)
host=b4.voiceless.aa.net.uk
[voiceless-3](aa-incoming)
host=c4.voiceless.aa.net.uk
[voiceless-4](aa-incoming)
host=d4.voiceless.aa.net.uk
[voiceless-5](aa-incoming)
host=e4.voiceless.aa.net.uk
[voiceless-6](aa-incoming)
host=f4.voiceless.aa.net.uk
[voiceless-7](aa-incoming)
host=g4.voiceless.aa.net.uk
[voiceless-8](aa-incoming)
host=h4.voiceless.aa.net.uk
[voiceless-9](aa-incoming)
host=i4.voiceless.aa.net.uk
[voiceless-10](aa-incoming)
host=j4.voiceless.aa.net.uk
---

The trustrpid and sendrpid settings were important.

---
extensions.conf  (DNIDs changed)
===
[aa-incoming]
exten => 4401,1,Goto(from-aa-line1,s,1)
exten => 4402,1,Goto(from-aa-line2,s,1)
exten => 4403,1,Goto(from-aa-line3,s,1)
---

Hope this helps.

Julian





-- 
Best regards,
 Julianmailto:jb_s...@trink.co.uk


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[asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate

[Phil Reynolds]

I have service with both VoIPtalk.org and Andrews & Arnold (aa.net.uk).
VoIPtalk calls are unauthenticated and reach me fine, but Andrews &
Arnold calls are authenticated. The last call I successfully received
was on Tuesday afternoon. Initially, A were for some odd reason not
sending calls to my server, but that has been resolved. The problem now
is that the calls fail to authenticate, and are therefore rejected -
error 403 is presented to them, and I see this in Asterisk's console: 

[Apr 23 11:53:19] NOTICE[27398][C-0004]: chan_sip.c:25535
handle_request_invite: Failed to authenticate device "X XX"
;tag=201604231153191

I have checked that the username and password in my config agree both
ends, and have even tried changing them.

The bulk of my calls come in on A, so I am obviously trying to find
out what has gone wrong. No-one else is seeing any problem. What do I
need to do to track this down?

-- 
Phil Reynolds
mail: phil-aster...@tinsleyviaduct.com
Web: http://phil.tinsleyviaduct.com/

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Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

[Steve Edwards]

On Sun, 20 Mar 2016, Trey Hilyard wrote:


On Mar 18, 2016 8:27 PM, "Steve Edwards"  wrote:
>>
>> On Fri, 18 Mar 2016, Trey Hilyard wrote:
>>
>>> I thought this would be as easy as
>>> exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10})
>
>
> How about something like:
>
> [parse-lrn]
>         exten = _x.,1,                  verbose(1,[${EXTEN}@${CONTEXT}])
>         same = n,                       set(DID=${CUT(EXTEN,\;,1)})
>         same = n,                       set(LRN=${CUT(EXTEN,\;,2):3:12})
>         same = n,                       execif($["${LRN:0:1}" = 
"+"]?set(LRN=${LRN:1}))
>         same = n,                       execif($["${LRN:0:1}" = 
"1"]?set(LRN=${LRN:1}))
>         same = n,                       goto(${LRN},${DID},1)
>         same = n,                       hangup()

That's a good one. One thing it doesn't do is actually validate that the 
LRN is mine, but that shouldn't be tough to add now the the LRN is in 
its own variable. Thanks for the help!


If the LRN is not yours, you will not have a matching context so the 
goto() will run the invalid handler (the 'i' extension). You could play an 
appropriate message there.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281-- 
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Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

[Trey Hilyard]

On Mar 18, 2016 8:27 PM, "Steve Edwards"  wrote:
>>
>> On Fri, 18 Mar 2016, Trey Hilyard wrote:
>>
>>> I thought this would be as easy as
>>> exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10})
>
>
> How about something like:
>
> [parse-lrn]
> exten = _x.,1,  verbose(1,[${EXTEN}@${CONTEXT}])
> same = n,   set(DID=${CUT(EXTEN,\;,1)})
> same = n,   set(LRN=${CUT(EXTEN,\;,2):3:12})
> same = n,   execif($["${LRN:0:1}" =
"+"]?set(LRN=${LRN:1}))
> same = n,   execif($["${LRN:0:1}" =
"1"]?set(LRN=${LRN:1}))
> same = n,   goto(${LRN},${DID},1)
> same = n,   hangup()

That's a good one. One thing it doesn't do is actually validate that the
LRN is mine, but that shouldn't be tough to add now the the LRN is in its
own variable. Thanks for the help!

>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> https://www.linkedin.com/in/steve-edwards-4244281
>
> --
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Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

[Steve Edwards]

On Fri, 18 Mar 2016, Trey Hilyard wrote:


I thought this would be as easy as
exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10})


Have you tried the '_!.' pattern?

--
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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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[asterisk-users] Incoming INVITE with Portability Info and LRN

[Trey Hilyard]

I am trying to set up my Asterisk server so that it will recognize an
incoming call to the Asterisk's own Location Routing Number (LRN),
validating the "rn" in the INVITE and then using the Called Number from the
INVITE as the extension in the dialplan.

The INVITE R-URI looks like:
INVITE 
sip:+19135041291;rn=+1913663;npdi@12.4.240.200:5060;user=phone;transport=udp
SIP/2.0

The +1913663000 is the LRN of the Asterisk box, so I would want to have the
dialplan validate that the "rn" is that number. The +19136631291 is the
extension within the system that they are trying to reach, that extension
will vary, and will have an exten defined in the dialplan.

I assume that this is just going to require that I do some matching and
substring-type variable replacement to hit a context with just the Called
Number part of the request, but I wondered if anyone had a working example
of this before I started putting too much effort into it.

As a PBX, Asterisk doesn't have to worry about portability, but I am using
it to simulate a full-blown Class 5 switch, so I have to have an LRN
assigned to it to allow users to port to that switch.

-Trey
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Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

[Trey Hilyard]

On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI 
wrote:

> Le 18/03/2016 16:20, Trey Hilyard a écrit :
> > I am trying to set up my Asterisk server so that it will recognize an
> > incoming call to the Asterisk's own Location Routing Number (LRN),
> > validating the "rn" in the INVITE and then using the Called Number from
> > the INVITE as the extension in the dialplan.
> >
> > The INVITE R-URI looks like:
> > INVITE
> > sip:+19135041291;rn=+1913663;npdi@12.4.240.200
> :5060;user=phone;transport=udp
> > SIP/2.0
> >
> > The +1913663000 is the LRN of the Asterisk box, so I would want to have
> > the dialplan validate that the "rn" is that number. The +19136631291 is
> > the extension within the system that they are trying to reach, that
> > extension will vary, and will have an exten defined in the dialplan.
> >
> > I assume that this is just going to require that I do some matching and
> > substring-type variable replacement to hit a context with just the
> > Called Number part of the request, but I wondered if anyone had a
> > working example of this before I started putting too much effort into it.
>
> Use the SIP_HEADER function
>
> http://www.voip-info.org/wiki/view/Asterisk+func+sip_header


I am not sure that this is needed here. The Request URI has all of the
values that I need. I agree that I might need to CUT part of the R-URI, but
I don't need access to any other header to find the info I need.

When the call arrives at the Asterisk right now, this is the exten/context
that it is hitting, so it already has the info I need:
Executing [9135041291;rn=+1913663;npdi@from_pstn:1]

As far as I can tell, I think that I just need to figure out how to make an
extension entry that matches on the "rn=+1913663\;npdi" and then moves
to another context (or same one) with ${EXTEN,0,10}.

I just can't get that first extension to match on the RN value.



>
>
> --
> Daniel
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

[Administrator TOOTAI]

Le 18/03/2016 16:20, Trey Hilyard a écrit :

I am trying to set up my Asterisk server so that it will recognize an
incoming call to the Asterisk's own Location Routing Number (LRN),
validating the "rn" in the INVITE and then using the Called Number from
the INVITE as the extension in the dialplan.

The INVITE R-URI looks like:
INVITE
sip:+19135041291;rn=+1913663;npdi@12.4.240.200:5060;user=phone;transport=udp
SIP/2.0

The +1913663000 is the LRN of the Asterisk box, so I would want to have
the dialplan validate that the "rn" is that number. The +19136631291 is
the extension within the system that they are trying to reach, that
extension will vary, and will have an exten defined in the dialplan.

I assume that this is just going to require that I do some matching and
substring-type variable replacement to hit a context with just the
Called Number part of the request, but I wondered if anyone had a
working example of this before I started putting too much effort into it.


Use the SIP_HEADER function

http://www.voip-info.org/wiki/view/Asterisk+func+sip_header

--
Daniel

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Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

[Trey Hilyard]

I thought this would be as easy as

exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10})

But it appears that the pattern match doesn't work once I get to the "r" in
"rn". I am assuming that the pattern match doesn't like dealing with
characters without taking the entire URI.

I am working on a plan using a lot more CUTs than I think I should need,
but we'll see if it works.

On Fri, Mar 18, 2016 at 10:58 AM Trey Hilyard  wrote:

> On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI 
> wrote:
>
>> Le 18/03/2016 16:20, Trey Hilyard a écrit :
>> > I am trying to set up my Asterisk server so that it will recognize an
>> > incoming call to the Asterisk's own Location Routing Number (LRN),
>> > validating the "rn" in the INVITE and then using the Called Number from
>> > the INVITE as the extension in the dialplan.
>> >
>> > The INVITE R-URI looks like:
>> > INVITE
>> > sip:+19135041291;rn=+1913663;npdi@12.4.240.200
>> :5060;user=phone;transport=udp
>> > SIP/2.0
>> >
>> > The +1913663000 is the LRN of the Asterisk box, so I would want to have
>> > the dialplan validate that the "rn" is that number. The +19136631291 is
>> > the extension within the system that they are trying to reach, that
>> > extension will vary, and will have an exten defined in the dialplan.
>> >
>> > I assume that this is just going to require that I do some matching and
>> > substring-type variable replacement to hit a context with just the
>> > Called Number part of the request, but I wondered if anyone had a
>> > working example of this before I started putting too much effort into
>> it.
>>
>> Use the SIP_HEADER function
>>
>> http://www.voip-info.org/wiki/view/Asterisk+func+sip_header
>
>
> I am not sure that this is needed here. The Request URI has all of the
> values that I need. I agree that I might need to CUT part of the R-URI, but
> I don't need access to any other header to find the info I need.
>
> When the call arrives at the Asterisk right now, this is the exten/context
> that it is hitting, so it already has the info I need:
> Executing [9135041291;rn=+1913663;npdi@from_pstn:1]
>
> As far as I can tell, I think that I just need to figure out how to make
> an extension entry that matches on the "rn=+1913663\;npdi" and then
> moves to another context (or same one) with ${EXTEN,0,10}.
>
> I just can't get that first extension to match on the RN value.
>
>
>
>>
>>
>> --
>> Daniel
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

[Steve Edwards]

On Fri, 18 Mar 2016, Steve Edwards wrote:


Have you tried the '_!.' pattern?


The '_x.' pattern works fine.

--
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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

[Steve Edwards]

On Fri, 18 Mar 2016, Trey Hilyard wrote:


I thought this would be as easy as
exten => _XX\;rn=+1913663,1,Goto(from_pstn,${EXTEN:0:10})


How about something like:

[parse-lrn]
exten = _x.,1,  verbose(1,[${EXTEN}@${CONTEXT}])
same = n,   set(DID=${CUT(EXTEN,\;,1)})
same = n,   set(LRN=${CUT(EXTEN,\;,2):3:12})
same = n,   execif($["${LRN:0:1}" = 
"+"]?set(LRN=${LRN:1}))
same = n,   execif($["${LRN:0:1}" = 
"1"]?set(LRN=${LRN:1}))
same = n,   goto(${LRN},${DID},1)
same = n,   hangup()

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

[Clemens Leu]

Larry Moore  omninet.net.au> writes:

> 
> sip.conf
> 
> [general]
> faxdetect=t38
> 
> [sipcall.ch]
> directmedia=no
> 
> In extensions.conf change Wait(2) to Wait(5), if your VSP sends you a 
> T.38 re-invite this will trigger the switch to the Fax extension.
> 
> If this proves successful you can work on removing the Wait() from your 
> dialplan as Asterisk will remain in the audio path and should 
> successfully switch to the fax extension if extension 200 or 201 answer 
> a call that happens to be a fax.
> 
> Larry.
> 


Hi to all

Sorry to bump this old thread. Well, I found a while ago finally the reason
why T.38 doesn't work in conjunction with Swiss VoIP provider sipcall.

Despite T.38 is stated as "supported", that provider does NOT support T.38.
Their T.38 gateway has some fundamental negotiation problems, - it "exceeds
the T4 timer of the T.30 protocol". Therefore, T.38 faxing does not work.
http://wiki.innovaphone.com/index.php?title=Howto:Sipcall_business_-_SIP_Provider_Compatibility_Test

Sipcall has confirmed me that they work now on a solution. Will see...

Kind regards,

Clemens


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[asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome

[Alex Villací­s Lasso]

I am having trouble getting Google Chrome to accept a WebRTC call coming from 
Asterisk, even though Firefox can (now) accept the same call without issue.

My setup is as follows:

Server:
CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146
asterisk-11.21.0 patched to work around 
https://issues.asterisk.org/jira/browse/ASTERISK-25659
openssl-1.0.1e-51.el7_2.2.x86_64
[root@elx4 ~]# openssl version
OpenSSL 1.0.1e-fips 11 Feb 2013
[root@elx4 ~]# openssl ecparam -list_curves
  secp384r1 : NIST/SECG curve over a 384 bit prime field
  secp521r1 : NIST/SECG curve over a 521 bit prime field
  prime256v1: X9.62/SECG curve over a 256 bit prime field

Client:
Fedora 23 x86_64
Linphone (linphone-3.6.1-10.fc23.x86_64)
Firefox 43 (firefox-43.0.3-1.fc23.x86_64)
Google Chrome (google-chrome-stable-47.0.2526.111-1.x86_64)
SIP.js 0.7.2

I set up my SIP configuration to have two SIP accounts. Account 1000 is the 
Linphone and 1001 is the webrtc:

[general]
faxdetect=no
vmexten=*97
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-2.11.0(11.20.0)
disallow=all
allow=g723
allow=ulaw
allow=gsm
allow=alaw
allow=g729
allow=speex
allow=g722
allow=h264
allow=h263p
allow=h263
allow=h261
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscipher=ALL
tlsclientmethod=tlsv1
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
callevents=no
jbenable=no
videosupport=yes
allowguest=no
srvlookup=no
defaultexpiry=120
minexpiry=60
maxexpiry=3600
registerattempts=0
registertimeout=20
g726nonstandard=no
maxcallbitrate=384
canreinvite=no
rtptimeout=30
rtpholdtimeout=300
rtpkeepalive=0
checkmwi=10
notifyringing=yes
notifyhold=yes
nat=yes

[1000]
deny=0.0.0.0/0.0.0.0
secret=6ff108122cce3b0b45e0abf374c14ef4
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=yes
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
icesupport=no
dtlsenable=no
dtlsverify=no
dtlssetup=actpass
encryption=no
callgroup=
pickupgroup=
dial=SIP/1000
mailbox=1000@device
permit=0.0.0.0/0.0.0.0
callerid=Usuario 1 elx4 <1000>
callcounter=yes
faxdetect=no

[1001]
deny=0.0.0.0/0.0.0.0
secret=ce93963b0751ed9a88ec1badbc073fce
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=yes
port=5060
qualify=yes
qualifyfreq=60
transport=wss,ws,udp,tcp,tls
avpf=yes
icesupport=yes
dtlsenable=yes
dtlsverify=no
dtlssetup=actpass
dtlscertfile=/var/lib/asterisk/keys/localhost.crt
dtlsprivatekey=/var/lib/asterisk/keys/localhost.key
encryption=yes
callgroup=
pickupgroup=
dial=SIP/1001
mailbox=1001@device
permit=0.0.0.0/0.0.0.0
callerid=Usuario Alex <1001>
callcounter=yes
faxdetect=no


With this setup, I can make calls using the SIP softphone as usual, both into and out of asterisk. After approving the certificate exceptions, I can also use the webrtc account to generate a call from either Firefox or Google Chrome into asterisk, out to 
the SIP softphone.


The problem arises when I try to make asterisk send a call into the browser. When using Firefox 43 I can receive the call normally (this required patching around ASTERISK-25659) and all is well. However, in Google Chrome, the call is rejected with a 
message of "Failed to set remote video description send parameters.." as shown in this SIP trace in the browser console:




Wed Jan 20 2016 13:54:53 GMT-0500 (ECT) | sip.transport | received WebSocket 
text message:

INVITE sip:8dgpkoa2@192.0.2.210;transport=wss SIP/2.0
Via: SIP/2.0/WS 10.1.0.4:5060;branch=z9hG4bK4f80b96d;rport
Max-Forwards: 70
From: "Anonymous" ;tag=as37a33245
To: 
Contact: 
Call-ID: 61c6be5b44d587c80b56f98e756037ab@10.1.0.4:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.20.0)
Date: Wed, 20 Jan 2016 18:54:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1799

v=0
o=root 469858785 469858785 IN IP4 10.1.0.4
s=Asterisk PBX 11.21.0
c=IN IP4 10.1.0.4
b=CT:384
t=0 0
m=audio 14814 UDP/TLS/RTP/SAVPF 4 0 3 8 18 110 9 101
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:0e746cd50c88ce6e383ff3882acebb80
a=ice-pwd:1a9a09862254ae253f06a0bb184fd1b5
a=candidate:Ha010004 1 UDP 2130706431 10.1.0.4 14814 typ host
a=candidate:Hc0a80592 1 UDP 2130706431 192.168.5.146 14814 typ host
a=candidate:Ha010004 2 UDP 2130706430 10.1.0.4 14815 typ host
a=candidate:Hc0a80592 2 UDP 2130706430 192.168.5.146 14815 typ host
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 

Re: [asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome

[Alex Villací­s Lasso]

El 20/01/16 a las 16:25, Alex Villací­s Lasso escribió:

I am having trouble getting Google Chrome to accept a WebRTC call coming from 
Asterisk, even though Firefox can (now) accept the same call without issue.

My setup is as follows:

Server:
CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146
asterisk-11.21.0 patched to work around 
https://issues.asterisk.org/jira/browse/ASTERISK-25659
openssl-1.0.1e-51.el7_2.2.x86_64
[root@elx4 ~]# openssl version
OpenSSL 1.0.1e-fips 11 Feb 2013
[root@elx4 ~]# openssl ecparam -list_curves
  secp384r1 : NIST/SECG curve over a 384 bit prime field
  secp521r1 : NIST/SECG curve over a 521 bit prime field
  prime256v1: X9.62/SECG curve over a 256 bit prime field

Client:
Fedora 23 x86_64
Linphone (linphone-3.6.1-10.fc23.x86_64)
Firefox 43 (firefox-43.0.3-1.fc23.x86_64)
Google Chrome (google-chrome-stable-47.0.2526.111-1.x86_64)
SIP.js 0.7.2

I set up my SIP configuration to have two SIP accounts. Account 1000 is the 
Linphone and 1001 is the webrtc:

[general]
faxdetect=no
vmexten=*97
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-2.11.0(11.20.0)
disallow=all
allow=g723
allow=ulaw
allow=gsm
allow=alaw
allow=g729
allow=speex
allow=g722
allow=h264
allow=h263p
allow=h263
allow=h261
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscipher=ALL
tlsclientmethod=tlsv1
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
callevents=no
jbenable=no
videosupport=yes
allowguest=no
srvlookup=no
defaultexpiry=120
minexpiry=60
maxexpiry=3600
registerattempts=0
registertimeout=20
g726nonstandard=no
maxcallbitrate=384
canreinvite=no
rtptimeout=30
rtpholdtimeout=300
rtpkeepalive=0
checkmwi=10
notifyringing=yes
notifyhold=yes
nat=yes

[1000]
deny=0.0.0.0/0.0.0.0
secret=6ff108122cce3b0b45e0abf374c14ef4
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=yes
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
icesupport=no
dtlsenable=no
dtlsverify=no
dtlssetup=actpass
encryption=no
callgroup=
pickupgroup=
dial=SIP/1000
mailbox=1000@device
permit=0.0.0.0/0.0.0.0
callerid=Usuario 1 elx4 <1000>
callcounter=yes
faxdetect=no

[1001]
deny=0.0.0.0/0.0.0.0
secret=ce93963b0751ed9a88ec1badbc073fce
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=yes
port=5060
qualify=yes
qualifyfreq=60
transport=wss,ws,udp,tcp,tls
avpf=yes
icesupport=yes
dtlsenable=yes
dtlsverify=no
dtlssetup=actpass
dtlscertfile=/var/lib/asterisk/keys/localhost.crt
dtlsprivatekey=/var/lib/asterisk/keys/localhost.key
encryption=yes
callgroup=
pickupgroup=
dial=SIP/1001
mailbox=1001@device
permit=0.0.0.0/0.0.0.0
callerid=Usuario Alex <1001>
callcounter=yes
faxdetect=no


With this setup, I can make calls using the SIP softphone as usual, both into and out of asterisk. After approving the certificate exceptions, I can also use the webrtc account to generate a call from either Firefox or Google Chrome into asterisk, out to 
the SIP softphone.


The problem arises when I try to make asterisk send a call into the browser. When using Firefox 43 I can receive the call normally (this required patching around ASTERISK-25659) and all is well. However, in Google Chrome, the call is rejected with a 
message of "Failed to set remote video description send parameters.." as shown in this SIP trace in the browser console:




Wed Jan 20 2016 13:54:53 GMT-0500 (ECT) | sip.transport | received WebSocket 
text message:

INVITE sip:8dgpkoa2@192.0.2.210;transport=wss SIP/2.0
Via: SIP/2.0/WS 10.1.0.4:5060;branch=z9hG4bK4f80b96d;rport
Max-Forwards: 70
From: "Anonymous" ;tag=as37a33245
To: 
Contact: 
Call-ID: 61c6be5b44d587c80b56f98e756037ab@10.1.0.4:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.20.0)
Date: Wed, 20 Jan 2016 18:54:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1799

v=0
o=root 469858785 469858785 IN IP4 10.1.0.4
s=Asterisk PBX 11.21.0
c=IN IP4 10.1.0.4
b=CT:384
t=0 0
m=audio 14814 UDP/TLS/RTP/SAVPF 4 0 3 8 18 110 9 101
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:0e746cd50c88ce6e383ff3882acebb80
a=ice-pwd:1a9a09862254ae253f06a0bb184fd1b5
a=candidate:Ha010004 1 UDP 2130706431 10.1.0.4 14814 typ host
a=candidate:Hc0a80592 1 UDP 2130706431 192.168.5.146 14814 typ host
a=candidate:Ha010004 2 UDP 2130706430 10.1.0.4 14815 typ host
a=candidate:Hc0a80592 2 UDP 2130706430 192.168.5.146 14815 typ host
a=connection:new
a=setup:actpass

Re: [asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome

[Alex Villací­s Lasso]

El 20/01/16 a las 18:33, Alex Villací­s Lasso escribió:

El 20/01/16 a las 16:25, Alex Villací­s Lasso escribió:

Partial fix: Google Chrome accepts the call if videosupport is set to "no". 
This is the SDP of the successful INVITE that Chrome accepts:

INVITE sip:8cj802p8@192.0.2.240;transport=wss SIP/2.0
Via: SIP/2.0/WS 10.1.0.4:5060;branch=z9hG4bK65071dc5;rport
Max-Forwards: 70
From: "Anonymous" ;tag=as474012b5
To: 
Contact: 
Call-ID: 73b82a5b6fbaab50741cd99424b1f31a@10.1.0.4:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.20.0)
Date: Wed, 20 Jan 2016 23:27:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 937

v=0
o=root 2094440180 2094440180 IN IP4 10.1.0.4
s=Asterisk PBX 11.21.0
c=IN IP4 10.1.0.4
t=0 0
m=audio 18758 UDP/TLS/RTP/SAVPF 4 0 3 8 18 110 9 101
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:0033e2a20fe4becd1c34b13f5efcf1e3
a=ice-pwd:65693b30588f061710baa3584253eaba
a=candidate:Ha010004 1 UDP 2130706431 10.1.0.4 18758 typ host
a=candidate:Hc0a80592 1 UDP 2130706431 192.168.5.146 18758 typ host
a=candidate:Ha010004 2 UDP 2130706430 10.1.0.4 18759 typ host
a=candidate:Hc0a80592 2 UDP 2130706430 192.168.5.146 18759 typ host
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 
A1:9E:D0:11:26:AA:30:00:AF:06:87:9C:A7:C2:70:4F:A3:3F:89:B5:7D:5C:FA:90:89:7E:D0:8A:F0:72:F5:2A
a=sendrecv

However, I want to enable full video passthrough. Is this some kind of video 
codec incompatibility?



If I enable allow=vp8 to the set of allowed codecs, Chrome accepts the video 
call, but now I get no sound with the demo-congrats command.

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Re: [asterisk-users] Incoming calls get 488 error

[Andres]

On 8/21/15 6:45 PM, Technical Support wrote:
I got a new SNOM M65 which works fine for outgoing calls, but incoming 
calls never ring at the handset.  I captured the SIP traffic and see 
that my M65 is replying with an 488 not acceptable here. From what I 
read this is usually codec related but both asterisk and the M65 are 
set for ulaw as first choice.
Looks like the SNOM does not accept the video call.  Maybe you should 
look into why the Asterisk is trying to use video in the first place.


I have a SIP trace below.  Can someone suggest why the 488 is being 
generated?


---

Received from udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (1198 bytes)

INVITE sip:290006@192.168.253.20;line=14994 SIP/2.0
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
Max-Forwards: 70
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994
Contact: sip:230@192.168.253.4:5060
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.10.2)
Date: Fri, 21 Aug 2015 22:37:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE

Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 606

v=0
o=root 1678280845 1678280845 IN IP4 192.168.253.4
s=Asterisk PBX 11.10.2
c=IN IP4 192.168.253.4
b=CT:384
t=0 0
m=audio 18090 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 12226 RTP/AVP 99 98 34 31
a=rtpmap:99 H264/9
a=fmtp:99 
redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0

a=rtpmap:98 H263-1998/9
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:34 H263/9
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:31 H261/9
a=sendrecv


Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (280 bytes)

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
Content-Length: 0



Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (441 bytes)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
Max-Forwards: 70
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994;tag=ld65q
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
Contact: sip:290006@192.168.253.20;line=14994
User-Agent: snomM700/03.24.0007 (MAC=0004136103FB; SER= 0; HW=255)
Content-Length: 0






--
Technical Support
http://www.cellroute.net


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[asterisk-users] Incoming calls get 488 error

[Technical Support]

I got a new SNOM M65 which works fine for outgoing calls, but incoming 
calls never ring at the handset.  I captured the SIP traffic and see 
that my M65 is replying with an 488 not acceptable here.  From what I 
read this is usually codec related but both asterisk and the M65 are set 
for ulaw as first choice.


I have a SIP trace below.  Can someone suggest why the 488 is being 
generated?


---

Received from udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (1198 bytes)

INVITE sip:290006@192.168.253.20;line=14994 SIP/2.0
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
Max-Forwards: 70
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994
Contact: sip:230@192.168.253.4:5060
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.10.2)
Date: Fri, 21 Aug 2015 22:37:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE

Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 606

v=0
o=root 1678280845 1678280845 IN IP4 192.168.253.4
s=Asterisk PBX 11.10.2
c=IN IP4 192.168.253.4
b=CT:384
t=0 0
m=audio 18090 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 12226 RTP/AVP 99 98 34 31
a=rtpmap:99 H264/9
a=fmtp:99 
redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0

a=rtpmap:98 H263-1998/9
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:34 H263/9
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:31 H261/9
a=sendrecv


Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (280 bytes)

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
Content-Length: 0



Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (441 bytes)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
Max-Forwards: 70
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994;tag=ld65q
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
Contact: sip:290006@192.168.253.20;line=14994
User-Agent: snomM700/03.24.0007 (MAC=0004136103FB; SER= 0; HW=255)
Content-Length: 0



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Re: [asterisk-users] Incoming calls get 488 error

[Rafael Prado Rocchi]

Hi,
By the sip trace is very difficult to tell because the SIP messages are fine. 
Try to enable all codec, and if possible copy and paste your asterisk sip 
configuration for this peer.



Enviado do meu telefone Android usando o Symantec TouchDown (www.symantec.com)


-Original Message-
From: Technical Support [supp...@telium.ca]
Received: sexta-feira, 21 ago 2015, 19:46
To: asterisk-users@lists.digium.com [asterisk-users@lists.digium.com]
Subject: [asterisk-users] Incoming calls get 488 error


I got a new SNOM M65 which works fine for outgoing calls, but incoming
calls never ring at the handset.  I captured the SIP traffic and see
that my M65 is replying with an 488 not acceptable here.  From what I
read this is usually codec related but both asterisk and the M65 are set
for ulaw as first choice.

I have a SIP trace below.  Can someone suggest why the 488 is being
generated?

---

Received from udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (1198 bytes)

INVITE sip:290006@192.168.253.20;line=14994 SIP/2.0
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
Max-Forwards: 70
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994
Contact: sip:230@192.168.253.4:5060
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.10.2)
Date: Fri, 21 Aug 2015 22:37:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 606

v=0
o=root 1678280845 1678280845 IN IP4 192.168.253.4
s=Asterisk PBX 11.10.2
c=IN IP4 192.168.253.4
b=CT:384
t=0 0
m=audio 18090 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 12226 RTP/AVP 99 98 34 31
a=rtpmap:99 H264/9
a=fmtp:99
redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=rtpmap:98 H263-1998/9
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:34 H263/9
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:31 H261/9
a=sendrecv


Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (280 bytes)

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
Content-Length: 0



Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (441 bytes)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
Max-Forwards: 70
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994;tag=ld65q
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
Contact: sip:290006@192.168.253.20;line=14994
User-Agent: snomM700/03.24.0007 (MAC=0004136103FB; SER= 0; HW=255)
Content-Length: 0



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[asterisk-users] Incoming calls to a GSM gateway SIP/2.0 401 Unauthorized response when dial 7777 to Asterisk

[Luis Eduardo Cortes]

Hello:

I'm newbie in asterisk, please help me.

My context is as follows:

192.168.4.2 -- Asterisk 11.13.1 complied from source

192.168.4.4 -- Yeastar NeoGate TG100 GSM gateway

When I call from a GSM cell phone, my TG100 GSM gateway answers and
dials extension  (configured as a hotline on TG100) to asterisk
server, but asterisk server sends me SIP/2.0 401 Unauthorized
response, I think it's a matter of contexts but I don't find the
problem.

Attached are sip.conf, extensions.conf and debug from 192.168.4.4
(TG100 GSM gateway).

Thanks in advance.
--- SIP read from UDP:192.168.4.4:5060 ---
INVITE sip:@192.168.4.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.4:5060;branch=z9hG4bK0b7c4652;rport
Max-Forwards: 70
From: 9 sip:9@192.168.4.4;tag=as67354416
To: sip:@192.168.4.2:5060
Contact: sip:9@192.168.4.4
Call-ID: 12663beb04ae514f10c4b3a145368d5c@192.168.4.4
CSeq: 102 INVITE
User-Agent: TG100
Date: Wed, 12 Nov 2014 10:13:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 1426707418 1426707418 IN IP4 192.168.4.4
s=Asterisk PBX 1.6.2.6
c=IN IP4 192.168.4.4
t=0 0
m=audio 10048 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
-
--- (14 headers 13 lines) ---
Sending to 192.168.4.4:5060 (no NAT)
Sending to 192.168.4.4:5060 (no NAT)
Using INVITE request as basis request - 12663beb04ae514f10c4b3a145368d5c@192.168.4.4
Found peer '5' for '9' from 192.168.4.4:5060

--- Reliably Transmitting (no NAT) to 192.168.4.4:5060 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.4.4:5060;branch=z9hG4bK0b7c4652;received=192.168.4.4;rport=5060
From: 9 sip:9@192.168.4.4;tag=as67354416
To: sip:@192.168.4.2:5060;tag=as16de6e5c
Call-ID: 12663beb04ae514f10c4b3a145368d5c@192.168.4.4
CSeq: 102 INVITE
Server: Asterisk PBX 11.13.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=72011a6b
Content-Length: 0



Scheduling destruction of SIP dialog '12663beb04ae514f10c4b3a145368d5c@192.168.4.4' in 6400 ms (Method: INVITE)

--- SIP read from UDP:192.168.4.4:5060 ---
ACK sip:@192.168.4.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.4:5060;branch=z9hG4bK0b7c4652;rport
Max-Forwards: 70
From: 9 sip:9@192.168.4.4;tag=as67354416
To: sip:@192.168.4.2:5060;tag=as16de6e5c
Contact: sip:9@192.168.4.4
Call-ID: 12663beb04ae514f10c4b3a145368d5c@192.168.4.4
CSeq: 102 ACK
User-Agent: TG100
Content-Length: 0

-
--- (10 headers 0 lines) ---
Really destroying SIP dialog '6e9eab843b74d0860b108ed13a5d22c9@192.168.4.4' Method: OPTIONS
Really destroying SIP dialog '12663beb04ae514f10c4b3a145368d5c@192.168.4.4' Method: ACK
uc*CLI 



extensions.conf
Description: Binary data


sip.conf
Description: Binary data
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Re: [asterisk-users] incoming calls fall into echo test mode

[A J Stiles]

On Saturday 19 Jul 2014, Norman Molhant wrote:

 I tried many things on our FreePBX box and found out
 the problem seems somehow linked with the customer's
 extension (or phone number), not his inbound route
 (changing the latter has no effect on the problem).
 
 Creating a new extension with another phone number
 would solve the problem (I tried it and it works),
 but this customer wants to keep his current phone
 number and when I tried deleting his extension then
 creating a new one with his current phone number,
 the new extension presented the same problem as the
 previous one...
 
 Anyone knows what could cause such a problem and/or
 how to solve it ?

You really have supplied incomplete information here, by neglecting to mention 
the actual extension number which is causing the problems.  That would have 
had somebody onto it like a shot.  What follows is an educated guess based on 
the most likely scenario according to the available information:

Somewhere in your dialplan, probably in a section that has already been 
helpfully configured for you by FreePBX, the extension number you assigned to 
your customer has been appropriated for an echotest.

I suggest to grep for  (firstly)  the extension number in question, and  (if 
that does not work, perhaps because the echotest is a wildcard match aot a 
literal one)  then search instead for 'exten[ ]*='  (afraid that one will 
give you many more hits .  you'll have to look through them yourself)  
under /etc/asterisk.  Use the -R option to search subfolders as well.


-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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[asterisk-users] incoming calls fall into echo test mode

[Norman Molhant]

Hello all,

Weird trouble here:
we have 60-some happy subscribers on a FreePBX box,
each with its own phone number, with no problem at all,
except for one (and only one) subscriber who has this
problem: his outgoing calls are ok, but when someone
dials his phone number (be it from our network or from
any other place in the world), the caller ears the
standard message signalling he has entered the echo
test mode and must dial # to exit that mode.

Most callers don't understand what's going on, then
give up and hang up without dialling #.  Very few
dial # one or more times, then those few get our
customer's phone ringing and are then able to reach
our customer.

I went through all the docs, wikis and discussions
I found on the web, without finding any data on how
to solve that problem.

I tried many things on our FreePBX box and found out
the problem seems somehow linked with the customer's
extension (or phone number), not his inbound route
(changing the latter has no effect on the problem).

Creating a new extension with another phone number
would solve the problem (I tried it and it works),
but this customer wants to keep his current phone
number and when I tried deleting his extension then
creating a new one with his current phone number,
the new extension presented the same problem as the
previous one...

Anyone knows what could cause such a problem and/or
how to solve it ?

Thanks,
Norman.
ad...@csur.ca






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Re: [asterisk-users] incoming calls fall into echo test mode

[Michelle Dupuis]

You might get a better response on the FreePBX forum.  (FreePBX adds pre-built 
dialplan elements onto standard asterisk.  This forum is more for Asterisk)

But some suggestions:

SSH to your PBX
enter the Asterisk CLI
set verbose to 10
Call into the problematic number
...and watch where the call is being misrouted in the dialplan



From: asterisk-users-boun...@lists.digium.com 
asterisk-users-boun...@lists.digium.com on behalf of Norman Molhant 
ad...@csur.ca
Sent: Saturday, July 19, 2014 10:43 AM
To: Asterisk Users List
Subject: [asterisk-users] incoming calls fall into echo test mode

Hello all,

Weird trouble here:
we have 60-some happy subscribers on a FreePBX box,
each with its own phone number, with no problem at all,
except for one (and only one) subscriber who has this
problem: his outgoing calls are ok, but when someone
dials his phone number (be it from our network or from
any other place in the world), the caller ears the
standard message signalling he has entered the echo
test mode and must dial # to exit that mode.

Most callers don't understand what's going on, then
give up and hang up without dialling #.  Very few
dial # one or more times, then those few get our
customer's phone ringing and are then able to reach
our customer.

I went through all the docs, wikis and discussions
I found on the web, without finding any data on how
to solve that problem.

I tried many things on our FreePBX box and found out
the problem seems somehow linked with the customer's
extension (or phone number), not his inbound route
(changing the latter has no effect on the problem).

Creating a new extension with another phone number
would solve the problem (I tried it and it works),
but this customer wants to keep his current phone
number and when I tried deleting his extension then
creating a new one with his current phone number,
the new extension presented the same problem as the
previous one...

Anyone knows what could cause such a problem and/or
how to solve it ?

Thanks,
Norman.
ad...@csur.ca






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Re: [asterisk-users] incoming calls fall into echo test mode

[Pat Collins]

Perhaps assigned as a test number somewhere along the line?
Are these ISDN, SIP, IAX calls?
There are MANY smart people on this list. 
Maybe sharing the relevant configs and traces is a good place to start???

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Norman Molhant
Sent: Saturday, July 19, 2014 10:43 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] incoming calls fall into echo test mode

Hello all,

Weird trouble here:
we have 60-some happy subscribers on a FreePBX box, each with its own phone
number, with no problem at all, except for one (and only one) subscriber who
has this
problem: his outgoing calls are ok, but when someone dials his phone number
(be it from our network or from any other place in the world), the caller
ears the standard message signalling he has entered the echo test mode and
must dial # to exit that mode.

Most callers don't understand what's going on, then give up and hang up
without dialling #.  Very few dial # one or more times, then those few get
our customer's phone ringing and are then able to reach our customer.

I went through all the docs, wikis and discussions I found on the web,
without finding any data on how to solve that problem.

I tried many things on our FreePBX box and found out the problem seems
somehow linked with the customer's extension (or phone number), not his
inbound route (changing the latter has no effect on the problem).

Creating a new extension with another phone number would solve the problem
(I tried it and it works), but this customer wants to keep his current phone
number and when I tried deleting his extension then creating a new one with
his current phone number, the new extension presented the same problem as
the previous one...

Anyone knows what could cause such a problem and/or how to solve it ?

Thanks,
Norman.
ad...@csur.ca






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Re: [asterisk-users] incoming calls fall into echo test mode

[covici]

check your logs /var/log/asterisk/full -- make sure your verbosity is
set high enough to do you good and you wll probably find the answer.

Pat Collins drdialt...@optonline.net wrote:

 Perhaps assigned as a test number somewhere along the line?
 Are these ISDN, SIP, IAX calls?
 There are MANY smart people on this list. 
 Maybe sharing the relevant configs and traces is a good place to start???
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Norman Molhant
 Sent: Saturday, July 19, 2014 10:43 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] incoming calls fall into echo test mode
 
 Hello all,
 
 Weird trouble here:
 we have 60-some happy subscribers on a FreePBX box, each with its own phone
 number, with no problem at all, except for one (and only one) subscriber who
 has this
 problem: his outgoing calls are ok, but when someone dials his phone number
 (be it from our network or from any other place in the world), the caller
 ears the standard message signalling he has entered the echo test mode and
 must dial # to exit that mode.
 
 Most callers don't understand what's going on, then give up and hang up
 without dialling #.  Very few dial # one or more times, then those few get
 our customer's phone ringing and are then able to reach our customer.
 
 I went through all the docs, wikis and discussions I found on the web,
 without finding any data on how to solve that problem.
 
 I tried many things on our FreePBX box and found out the problem seems
 somehow linked with the customer's extension (or phone number), not his
 inbound route (changing the latter has no effect on the problem).
 
 Creating a new extension with another phone number would solve the problem
 (I tried it and it works), but this customer wants to keep his current phone
 number and when I tried deleting his extension then creating a new one with
 his current phone number, the new extension presented the same problem as
 the previous one...
 
 Anyone knows what could cause such a problem and/or how to solve it ?
 
 Thanks,
 Norman.
 ad...@csur.ca
 
 
 
 
 
 
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 -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
 Asterisk? Join us for a live introductory webinar every Thurs:
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 asterisk-users mailing list
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How do
you spend it?

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 cov...@ccs.covici.com

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Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

[Larry Moore]

On 3/02/2014 3:34 PM, Jakob-Matthias Böttger wrote:
.
.
.


[sipcall.ch]
type=peer
insecure=invite
defaultuser=123456789
fromuser=123456789
fromdomain=voipdomain.com
secret=gueswhat
host=voipdomain.com
qualify=yes
context=from-sip
dtmfmode=rfc2833
callbackextension=123456789



add

directmedia=no
setvar=FAXOPT(gateway)=no

change
insecure=port,invite




[fax-rx]

exten = receive,1,NoOp( FAX RECEIVE )
exten = receive,n,Set(GLOBAL(FAXCOUNT)=$[${GLOBAL(FAXCOUNT)} + 1])
exten = receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})


Do you want to keep your received faxes or is it OK to overwrite them 
the next time asterisk is re-started!?




udptl.conf
[general]
udptlstart=4000
udptlend=4999
udptlfecentries = 3
udptlfecspan = 3
use_even_ports = no



You may want to change

use_even_ports=yes

You will need to restart Asterisk for this change.

Some other suggestion if the above doesn't help are;

faxdetect=cng
t38pt_udptl=no

Larry.

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Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

[Jakob-Matthias Böttger]

Hi, changing

faxdetect=cng
and
t38pt_udptl=no

helped making it work.

Thanks


Am 03.02.2014 11:57, schrieb Larry Moore:

On 3/02/2014 3:34 PM, Jakob-Matthias Böttger wrote:
.
.
.


[sipcall.ch]
type=peer
insecure=invite
defaultuser=123456789
fromuser=123456789
fromdomain=voipdomain.com
secret=gueswhat
host=voipdomain.com
qualify=yes
context=from-sip
dtmfmode=rfc2833
callbackextension=123456789



add

directmedia=no
setvar=FAXOPT(gateway)=no

change
insecure=port,invite




[fax-rx]

exten = receive,1,NoOp( FAX RECEIVE )
exten = receive,n,Set(GLOBAL(FAXCOUNT)=$[${GLOBAL(FAXCOUNT)} + 1])
exten = receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})


Do you want to keep your received faxes or is it OK to overwrite them 
the next time asterisk is re-started!?




udptl.conf
[general]
udptlstart=4000
udptlend=4999
udptlfecentries = 3
udptlfecspan = 3
use_even_ports = no



You may want to change

use_even_ports=yes

You will need to restart Asterisk for this change.

Some other suggestion if the above doesn't help are;

faxdetect=cng
t38pt_udptl=no

Larry.




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Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

[Larry Moore]

On 3/02/2014 7:15 PM, Jakob-Matthias Böttger wrote:

Hi, changing

faxdetect=cng
and
t38pt_udptl=no

helped making it work.



Hmm, the fax will be received as an audio call rather than T.38, setting 
t38pt_udptl=no has turned off T.38.


Do you know if your upstream provider supports T.38?

Larry.

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Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

[Jakob-Matthias Böttger]

as He is describing it he should actually provide t.38. but i don't know 
why it is not working thus im now getting


Feb  3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 
process_sdp: Failed to initialize UDPTL, declining image stream
[Feb  3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497 
process_sdp: Insufficient information in SDP (c=)...

and then the fax session starts recording data

Am 03.02.2014 12:34, schrieb Larry Moore:

On 3/02/2014 7:15 PM, Jakob-Matthias Böttger wrote:

Hi, changing

faxdetect=cng
and
t38pt_udptl=no

helped making it work.



Hmm, the fax will be received as an audio call rather than T.38, 
setting t38pt_udptl=no has turned off T.38.


Do you know if your upstream provider supports T.38?

Larry.




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Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

[Larry Moore]

On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote:

as He is describing it he should actually provide t.38. but i don't know
why it is not working thus im now getting

Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp:
Failed to initialize UDPTL, declining image stream
[Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497
process_sdp: Insufficient information in SDP (c=)...
and then the fax session starts recording data



In udptl.conf set use_even_ports=yes and then issue a reload.

You can confirm the settings have been applied by performing udptl show 
config.


Change the the t38 line to read as;

t38pt_udptl=yes,redundancy,maxdatagram=400

Reload sip and test.

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Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

[Jakob-Matthias Böttger]

Am 03.02.2014 12:56, schrieb Larry Moore:

On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote:

as He is describing it he should actually provide t.38. but i don't know
why it is not working thus im now getting

Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp:
Failed to initialize UDPTL, declining image stream
[Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497
process_sdp: Insufficient information in SDP (c=)...
and then the fax session starts recording data



In udptl.conf set use_even_ports=yes and then issue a reload.

You can confirm the settings have been applied by performing udptl 
show config.


Change the the t38 line to read as;

t38pt_udptl=yes,redundancy,maxdatagram=400

Reload sip and test.


after that i started udptl debug as well and now i'm getting lots of

UDPTL (SIP/sipcall.ch-0007): packet to 212.117.203.76:24492 (seq 
152, len 11)


and in between

[Feb  3 13:18:30] WARNING[26313][C-0007]: res_rtp_asterisk.c:3548 
ast_rtp_read: RTP Read too short


and in the end

[Feb  3 13:18:37] WARNING[9942]: chan_sip.c:4409 __sip_autodestruct: 
Autodestruct on dialog '24d15e0d-28df847a-9fae13c-7...@sip.iforb.com~1o' 
with owner SIP/sipcall.ch-0007 in place (Method: BYE). Rescheduling 
destruction for 1 ms
[Feb  3 13:18:41] ERROR[26313][C-0007]: res_fax.c:1535 
generic_fax_exec: channel 'SIP/sipcall.ch-0007' FAX session '7' 
failure, reason: 'fax session timed-out' (TIMEOUT)
  == Spawn extension (fax-rx, receive, 11) exited non-zero on 
'SIP/sipcall.ch-0007'



Thx, Jakob

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Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

[Jakob-Matthias Böttger]

Am 03.02.2014 13:20, schrieb Jakob-Matthias Böttger:

Am 03.02.2014 12:56, schrieb Larry Moore:

On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote:
as He is describing it he should actually provide t.38. but i don't 
know

why it is not working thus im now getting

Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 
process_sdp:

Failed to initialize UDPTL, declining image stream
[Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497
process_sdp: Insufficient information in SDP (c=)...
and then the fax session starts recording data



In udptl.conf set use_even_ports=yes and then issue a reload.

You can confirm the settings have been applied by performing udptl 
show config.


Change the the t38 line to read as;

t38pt_udptl=yes,redundancy,maxdatagram=400

Reload sip and test.


after that i started udptl debug as well and now i'm getting lots of

UDPTL (SIP/sipcall.ch-0007): packet to 212.117.203.76:24492 (seq 
152, len 11)


and in between

[Feb  3 13:18:30] WARNING[26313][C-0007]: res_rtp_asterisk.c:3548 
ast_rtp_read: RTP Read too short


and in the end

[Feb  3 13:18:37] WARNING[9942]: chan_sip.c:4409 __sip_autodestruct: 
Autodestruct on dialog 
'24d15e0d-28df847a-9fae13c-7...@sip.iforb.com~1o' with owner 
SIP/sipcall.ch-0007 in place (Method: BYE). Rescheduling 
destruction for 1 ms
[Feb  3 13:18:41] ERROR[26313][C-0007]: res_fax.c:1535 
generic_fax_exec: channel 'SIP/sipcall.ch-0007' FAX session '7' 
failure, reason: 'fax session timed-out' (TIMEOUT)
  == Spawn extension (fax-rx, receive, 11) exited non-zero on 
'SIP/sipcall.ch-0007'



Thx, Jakob

may do i have to open more ports then udp 1:2 (RTP), udp 
4000:4999 (UDPTL) and tcp 5060,5061(SIP/TLS)


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Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

[Larry Moore]

On 3/02/2014 8:42 PM, Jakob-Matthias Böttger wrote:

Am 03.02.2014 13:20, schrieb Jakob-Matthias Böttger:

Am 03.02.2014 12:56, schrieb Larry Moore:

On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote:

as He is describing it he should actually provide t.38. but i don't
know
why it is not working thus im now getting

Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353
process_sdp:
Failed to initialize UDPTL, declining image stream
[Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10497
process_sdp: Insufficient information in SDP (c=)...
and then the fax session starts recording data



In udptl.conf set use_even_ports=yes and then issue a reload.

You can confirm the settings have been applied by performing udptl
show config.

Change the the t38 line to read as;

t38pt_udptl=yes,redundancy,maxdatagram=400

Reload sip and test.


after that i started udptl debug as well and now i'm getting lots of

UDPTL (SIP/sipcall.ch-0007): packet to 212.117.203.76:24492 (seq
152, len 11)

and in between

[Feb 3 13:18:30] WARNING[26313][C-0007]: res_rtp_asterisk.c:3548
ast_rtp_read: RTP Read too short

and in the end

[Feb 3 13:18:37] WARNING[9942]: chan_sip.c:4409 __sip_autodestruct:
Autodestruct on dialog
'24d15e0d-28df847a-9fae13c-7...@sip.iforb.com~1o' with owner
SIP/sipcall.ch-0007 in place (Method: BYE). Rescheduling
destruction for 1 ms
[Feb 3 13:18:41] ERROR[26313][C-0007]: res_fax.c:1535
generic_fax_exec: channel 'SIP/sipcall.ch-0007' FAX session '7'
failure, reason: 'fax session timed-out' (TIMEOUT)
== Spawn extension (fax-rx, receive, 11) exited non-zero on
'SIP/sipcall.ch-0007'


Thx, Jakob


may do i have to open more ports then udp 1:2 (RTP), udp
4000:4999 (UDPTL) and tcp 5060,5061(SIP/TLS)



The T.38 connection will be attempted when ReceiveFax is called.

The port number to use should be in the SDP information, yes, allow udp 
ports 4000-4999 in and out. If your firewall can be so configured you 
could set it to allow traffic in and out based upon the user ID Asterisk 
is running as, assuming it is using a unique unprivileged id.


You may like to try the following to see if your SIP provider will 
initiate a T.38 re-invite.


sip.conf

[general]
faxdetect=t38

[sipcall.ch]
directmedia=no


In extensions.conf change Wait(2) to Wait(5), if your VSP sends you a 
T.38 re-invite this will trigger the switch to the Fax extension.


If this proves successful you can work on removing the Wait() from your 
dialplan as Asterisk will remain in the audio path and should 
successfully switch to the fax extension if extension 200 or 201 answer 
a call that happens to be a fax.


Larry.

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[asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

[Jakob-Matthias Böttger]

Hi, im using a Asterisk Server which is not behind NAT.
First i had problems with the fax detection. But this is now solved 
after adding a wait(2) at the correct place. But i'm still unable to 
receive a fax due to res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too 
short after the Fax session has started.



My sip.conf includes

[general]
allowguest=no
alwaysauthreject=yes

sendrpid=rpid
trustrpid=yes
language=de
videosupport=yes
callevents=yes
qualify=yes
nat=force_rport,comedia
faxdetect=yes
t38pt_udptl=yes,redundancy,maxdatagram=400
directrtpsetup=yes
disallow=all
allow=ulaw
allow=alaw

and the corresponding Peer

[sipcall.ch]
type=peer
insecure=invite
defaultuser=123456789
fromuser=123456789
fromdomain=voipdomain.com
secret=gueswhat
host=voipdomain.com
qualify=yes
context=from-sip
dtmfmode=rfc2833
callbackextension=123456789

the Dialplan

[inbound]
exten = _X.,1,Answer()
exten = _X.,n,Set(DB(lastcaller/number)=${CALLERID(num)})
exten = _X.,n,GotoIf(${BLACKLIST()}?black,1)
exten = _X.,n,Wait(2)
exten = _X.,n,Dial(SIP/200SIP/201,60,tToxX)
exten = _X.,n,Goto(ausser-zeit,_X.,3)
exten = _X.,n,Hangup()

exten = fax,1,NoOp( FAX DETECTED )
exten = fax,n,Goto(fax-rx,receive,1)

[fax-rx]

exten = receive,1,NoOp( FAX RECEIVE )
exten = receive,n,Set(GLOBAL(FAXCOUNT)=$[${GLOBAL(FAXCOUNT)} + 1])
exten = receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
exten = receive,n,Set(FAXFILE=fax-${FAXCOUNT}-rx.tif)
exten = receive,n,Set(GLOBAL(LASTFAXCALLERoNUM)=${CALLERID(num)})
exten = receive,n,Set(GLOBAL(LASTFAXCALLERNAME)=${CALLERID(name)})
exten = receive,n,NoOp( SETTING FAXOPT )
exten = receive,n,Set(FAXOPT(ecm)=yes)
exten = receive,n,Set(FAXOPT(headerinfo)=MYFAX RX)
exten = receive,n,Set(FAXOPT(localstationid)=1234567890)
exten = receive,n,Set(FAXOPT(maxrate)=14400)
exten = receive,n,Set(FAXOPT(minrate)=2400)
exten = receive,n,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)})
exten = receive,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)})
exten = receive,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)})
exten = receive,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)})
exten = receive,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)})
exten = receive,n,NoOp( RECEIVING FAX : ${FAXFILE} )
exten = receive,n,ReceiveFAX(/var/spool/asterisk/faxin/${FAXFILE},dfs)
exten = h,1,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)})
exten = h,n,NoOp(FAXOPT(filename) : ${FAXOPT(filename)})
exten = h,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)})
exten = h,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)})
exten = h,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)})
exten = h,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)})
exten = h,n,NoOp(FAXOPT(pages) : ${FAXOPT(pages)})
exten = h,n,NoOp(FAXOPT(rate) : ${FAXOPT(rate)})
exten = h,n,NoOp(FAXOPT(remotestationid) : ${FAXOPT(remotestationid)})
exten = h,n,NoOp(FAXOPT(resolution) : ${FAXOPT(resolution)})
exten = h,n,NoOp(FAXOPT(status) : ${FAXOPT(status)})
exten = h,n,NoOp(FAXOPT(statusstr) : ${FAXOPT(statusstr)})
exten = h,n,NoOp(FAXOPT(error) : ${FAXOPT(error)})

udptl.conf
[general]
udptlstart=4000
udptlend=4999
udptlfecentries = 3
udptlfecspan = 3
use_even_ports = no

rtp.conf
[general]
rtpstart=1
rtpend=2

res_fax.conf
[general]
maxrate=14400
minrate=2400
statusevents=yes
modems=v17,v27,v29
ecm=yes


mail*CLI core set verbose 6
Set remote console verbosity to 6
  == Using SIP RTP CoS mark 5
-- Executing [41325122774@from-sip:1] 
Answer(SIP/sipcall.ch-008d, ) in new stack
0x7f3964080f30 -- Probation passed - setting RTP source 
address to 123.456.789.123:20600
Got  RTP packet from123.456.789.123:20600 (type 00, seq 042281, ts 
1387619622, len 000160)
-- Executing [41325122774@from-sip:2] 
Set(SIP/sipcall.ch-008d, DB(lastcaller/number)=987654321) in new 
stack
-- Executing [41325122774@from-sip:3] 
GotoIf(SIP/sipcall.ch-008d, 0?black,1) in new stack
-- Executing [41325122774@from-sip:4] 
Wait(SIP/sipcall.ch-008d, 2) in new stack
Got  RTP packet from123.456.789.123:20600 (type 00, seq 042282, ts 
1387619782, len 000160)
Got  RTP packet from123.456.789.123:20600 (type 00, seq 042283, ts 
1387619942, len 000160)
Got  RTP packet from123.456.789.123:20600 (type 00, seq 042284, ts 
1387620102, len 000160)
Got  RTP packet from123.456.789.123:20600 (type 00, seq 042285, ts 
1387620262, len 000160)
Got  RTP packet from123.456.789.123:20600 (type 00, seq 042286, ts 
1387620422, len 000160)
Got  RTP packet from123.456.789.123:20600 (type 00, seq 042287, ts 
1387620582, len 000160)
Got  RTP packet from123.456.789.123:20600 (type 00, seq 042288, ts 
1387620742, len 000160)
Got  RTP packet from123.456.789.123:20600 (type 00, seq 042289, ts 
1387620902, len 000160)
Got  RTP packet from123.456.789.123:20600 (type 00, seq 042290, ts 
1387621062, len 000160)
Got  RTP packet from123.456.789.123:20600 (type 00, seq 042291, ts 
1387621222, len 000160)
Got  RTP packet from123.456.789.123:20600 (type 00, seq 

[asterisk-users] incoming DAHDI Channel explained

[Thorsten Göllner]

Hi,

I use a Sangoma A104d-Card (with 4 x germany E1). I process some calls 
via an AGI-Script. When parsing the AGI-Variables I can see one that 
look like that:


[agi_channel] = DAHDI/i3/211123456-89c

What hat do the values mean in detail, please?

DAHDI : this is clear
i3 : does it mean, that the call comes in via E1-Port 3?
211123456 : Incoming-Call Caller-ID
-89c : ?

WANPIPE Release: 7.0.1
DAHDI Version: 2.6.2 Echo Canceller: HWEC
libpri version: 1.4.12

Best regards
-Thorsten-


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Re: [asterisk-users] incoming DAHDI Channel explained

[jg]

Sangoma's tech support is probably the better source of information.

DAHDI: obviously DAHDI channel
i: incoming call
3: span 3 (not the port)
211123456: CLID, probably subject to filtering (see 
national/international prefix settings)

89c: internal counter (i.e. 2204 calls so far)

jg

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Re: [asterisk-users] incoming DAHDI Channel explained

[Richard Mudgett]

 Sangoma's tech support is probably the better source of information.
 
 DAHDI: obviously DAHDI channel
 i: incoming call

The 'i' is for ISDN not incoming call since it will be this way for outgoing 
calls as well.

 3: span 3 (not the port)
 211123456: CLID, probably subject to filtering (see
 national/international prefix settings)
 89c: internal counter (i.e. 2204 calls so far)

The other fields are pretty much as described by jg.

Richard

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Re: [asterisk-users] incoming DAHDI Channel explained

[jg]

Yes, my assumption was wrong and to make things worse, my CDR data 
clearly show that i cannot denote incoming calls.


Maybe it's time that I learn the rules as well:

Analog channels  do not seem to have a special identifier. The 1st call 
for analog channel 13 would be s.th. like DAHDI/13-1.


Outside calls via an ISDN connection with s.th. like 
DIAL(DAHDI/r2/08932168,..) would dial the number using DAHDI group 2 in 
a round robin fashion, but internally the channel would be s.th. like 
DAHDI/iX/08932168-abcd. The span X is not related to the dial group and 
depends on the configuration.


For a BRI device a single span has 2 channels, a PRI device up to 30. As 
far as channel variables go the actual channel does not seem to get 
reported, but this is not really necessary.


jg

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Re: [asterisk-users] incoming DAHDI Channel explained

[Mordechay Kaganer]

B.H.

Hi!

On Wed, Jun 5, 2013 at 7:26 PM, jg webaccou...@jgoettgens.de wrote:

For a BRI device a single span has 2 channels, a PRI device up to 30. As
 far as channel variables go the actual channel does not seem to get
 reported, but this is not really necessary.


AFAIK, at least for AMI listeners, the real channel/span is reported
by DAHDIChannelEvent attributes:

'dahdichannel' reports the actual DAHDI channel number
'dahdispan' is a span number




 jg



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[asterisk-users] Incoming Fax to Recipient using OCR

[Roy Abshire]

I have fax working but since most people and services don't know how to 
Fax to Extensions,

I installed tesseract to convert the Fax to Text.

I really only need the First Page converted and will tell Faxers to make 
sure they put To: Name on the cover page.


tesseract is converting the entire fax fine but its unnecessary and 
extra time to convert the entire fax.


I searched and can't find anything on how to tell it just to do the 
first page.  Does anyone have any ideas?


I created a perl script I borrowed but I don't know PERL.

I know PHP so can someone show me how to use REGEX in Perl to search the 
output.txt file for the to: name or TO: NAME or To: Name


Then I want to do something like:

Switch($to) {
Case: Roy - Email u...@gmail.com
Case: Jeff - Email u...@yahoo.com
Default:
Email ad...@domain.com
}

--
Co-op Vacation Rentals
www.coopvr.com
15218 Summit Ave
Suite #300-354
Fontana, CA 92336
Phone/Fax (855) 760-COOP (2667)


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[asterisk-users] Incoming Fax to Recipient using OCR

[Roy Abshire]

I have fax working but since most people and services don't know how to 
Fax to Extensions,

I installed tesseract to convert the Fax to Text.

I really only need the First Page converted and will tell Faxers to make 
sure they put To: Name on the cover page.


tesseract is converting the entire fax fine but its unnecessary and 
extra time to convert the entire fax.


I searched and can't find anything on how to tell it just to do the 
first page.  Does anyone have any ideas?


I created a perl script I borrowed but I don't know PERL.

I know PHP so can someone show me how to use REGEX in Perl to search the 
output.txt file for the to: name or TO: NAME or To: Name


Then I want to do something like:

Switch($to) {
Case: Roy - Email u...@gmail.com
Case: Jeff - Email u...@yahoo.com
Default:
Email ad...@domain.com
}

--
Co-op Vacation Rentals
www.coopvr.com
15218 Summit Ave
Suite #300-354
Fontana, CA 92336
Phone/Fax (855) 760-COOP (2667)


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Re: [asterisk-users] Incoming Fax to Recipient using OCR

[Danny Nicholas]

If ($_ =~ /[Tt][Oo]\:.[Nn]ame/) {
Is the way I do it. 
If ($_ =~ /[Tt][Oo]..[Nn]ame/) {
Would also work


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roy Abshire
Sent: Tuesday, November 06, 2012 1:51 PM
To: Asterisk Users
Subject: [asterisk-users] Incoming Fax to Recipient using OCR

I have fax working but since most people and services don't know how to Fax
to Extensions, I installed tesseract to convert the Fax to Text.

I really only need the First Page converted and will tell Faxers to make
sure they put To: Name on the cover page.

tesseract is converting the entire fax fine but its unnecessary and extra
time to convert the entire fax.

I searched and can't find anything on how to tell it just to do the first
page.  Does anyone have any ideas?

I created a perl script I borrowed but I don't know PERL.

I know PHP so can someone show me how to use REGEX in Perl to search the
output.txt file for the to: name or TO: NAME or To: Name

Then I want to do something like:

Switch($to) {
Case: Roy - Email u...@gmail.com
Case: Jeff - Email u...@yahoo.com
Default:
 Email ad...@domain.com
}

--
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www.coopvr.com
15218 Summit Ave
Suite #300-354
Fontana, CA 92336
Phone/Fax (855) 760-COOP (2667)


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Re: [asterisk-users] Incoming Fax to Recipient using OCR

[Christopher Harrington]

On Tue, Nov 6, 2012 at 1:50 PM, Roy Abshire r...@coopvr.com wrote:

 I have fax working but since most people and services don't know how to
 Fax to Extensions,
 I installed tesseract to convert the Fax to Text.

 I really only need the First Page converted and will tell Faxers to make
 sure they put To: Name on the cover page.

 tesseract is converting the entire fax fine but its unnecessary and extra
 time to convert the entire fax.

 I searched and can't find anything on how to tell it just to do the first
 page.  Does anyone have any ideas?



If you're passing a TIFF file to tesseract, you can pass it through
imagemagick first to pop off the first page. This really seems off-topic
for Asterisk.

-- 
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ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248
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[asterisk-users] Incoming SIP call is rejected always.

[Yaroslav Panych]

Hi

Have an asterisk. Setup a couple of friends.
Sip.conf - http://pastebin.com/zUgiYbBi
Trying to make incoming call, and have such error(cli output)
http://pastebin.com/zFfgYcNR

NOTICE[4994]: chan_sip.c:23316 handle_request_invite: Call from
'RMT20' (192.168.8.1:5062) to extension '4001020' rejected because
extension not found in context 'rmt-context'.
But, as you see, there is such extension.

What I'm doing wrong?

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Re: [asterisk-users] Incoming SIP call is rejected always.

[Danny Nicholas]

Maybe it needs to be _4001020?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yaroslav
Panych
Sent: Tuesday, April 17, 2012 7:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Incoming SIP call is rejected always.

Hi

Have an asterisk. Setup a couple of friends.
Sip.conf - http://pastebin.com/zUgiYbBi
Trying to make incoming call, and have such error(cli output)
http://pastebin.com/zFfgYcNR

NOTICE[4994]: chan_sip.c:23316 handle_request_invite: Call from 'RMT20'
(192.168.8.1:5062) to extension '4001020' rejected because extension not
found in context 'rmt-context'.
But, as you see, there is such extension.

What I'm doing wrong?

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Re: [asterisk-users] Incoming SIP call is rejected always.

[Yaroslav Panych]

2012/4/17 Danny Nicholas da...@debsinc.com:
 Maybe it needs to be _4001020?


Not, it doesn't. Actually I have traced this incoming call step by
step. Real reason it refuses - wrong domain. But why it wrong - have
not any idea.

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Re: [asterisk-users] Incoming SIP call is rejected always.

[Matthew Jordan]

Without knowing the URI the INVITE request was addressed to, its
difficult to say what might be the actual cause of this.  However,
in your SIP configuration you have set allowexternaldomains to no.
That implies that if the domain of the URI does not match any
of the allowed domains you have set, that the INVITE request will
be rejected.

I imagine that this is the case, as ASTERISK-19601 noted that
when this situation occurs, the NOTICE message indicates that
there is a failure to match the extension, as opposed to a failure
to match an allowed domain.

Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org

- Original Message -
 From: Yaroslav Panych panyc...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Tuesday, April 17, 2012 4:58:14 PM
 Subject: Re: [asterisk-users] Incoming SIP call is rejected always.
 
 2012/4/17 Danny Nicholas da...@debsinc.com:
  Maybe it needs to be _4001020?
 
 
 Not, it doesn't. Actually I have traced this incoming call step by
 step. Real reason it refuses - wrong domain. But why it wrong - have
 not any idea.
 
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Re: [asterisk-users] Incoming SIP call is rejected always.

[Yaroslav Panych]

2012/4/18 Matthew  Jordan mjor...@digium.com:
 I imagine that this is the case, as ASTERISK-19601 noted that
 when this situation occurs, the NOTICE message indicates that
 there is a failure to match the extension, as opposed to a failure
 to match an allowed domain.

Yes, it was hell to detect real error cause(I was forced to learn how
to debug in KDevelop in less than four hours). Yes, it looks like
ASTERISK-19601. But still I cannot understand why asterisk extracts
wrong domain from request.
 However, in your SIP configuration you have set allowexternaldomains to no.
Yes, it is intended.

 Without knowing the URI the INVITE request was addressed to, its
 difficult to say what might be the actual cause of this.
I first letter I have provided CLI log which contains full request
packets(Authless and authed INVITE included).

Probably I do not understand how to configure Asterisk:
I have one asterisk. It serves SIP domain example.com. This asterisk
must be able to establish session with registered client of this
account and also must be able to accept incoming sessions. No sessions
with 3rd-party accounts on 3rd-party domains allowed to established.
How I should setup this asterisk?

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Re: [asterisk-users] Incoming SIP call is rejected always.

[Matthew Jordan]

- Original Message -
 From: Yaroslav Panych panyc...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Tuesday, April 17, 2012 6:56:17 PM
 Subject: Re: [asterisk-users] Incoming SIP call is rejected always.
 
 2012/4/18 Matthew  Jordan mjor...@digium.com:
  I imagine that this is the case, as ASTERISK-19601 noted that
  when this situation occurs, the NOTICE message indicates that
  there is a failure to match the extension, as opposed to a failure
  to match an allowed domain.
 
 Yes, it was hell to detect real error cause(I was forced to learn how
 to debug in KDevelop in less than four hours). Yes, it looks like
 ASTERISK-19601. But still I cannot understand why asterisk extracts
 wrong domain from request.
  However, in your SIP configuration you have set
  allowexternaldomains to no.
 Yes, it is intended.
 
  Without knowing the URI the INVITE request was addressed to, its
  difficult to say what might be the actual cause of this.
 I first letter I have provided CLI log which contains full request
 packets(Authless and authed INVITE included).
 
 Probably I do not understand how to configure Asterisk:
 I have one asterisk. It serves SIP domain example.com. This asterisk
 must be able to establish session with registered client of this
 account and also must be able to accept incoming sessions. No
 sessions
 with 3rd-party accounts on 3rd-party domains allowed to established.
 How I should setup this asterisk?

Well, I can't tell you how to configure your Asterisk server.  However,
I can tell you why Asterisk rejected the INVITE request.

The URI that the INVITE request was addressed to is
4001020@192.168.8.2:5060.  The domain portion of this URI is
192.168.8.2.  Hence, the allowed domains need to include that
particular IPv4 address.  Looking at the allowed domains you've
specified in sip.conf, we have:

domain=sop-korniychuk
domain=192.168.8.1
domain=192.168.8.1:5062

So, since the INVITE request does not match any of those three domains,
its rejected.

Note: I noticed that you have autodomain set to yes; I'm going to
assume that the IPv4 address 192.168.8.2 is not associated with the
server.

Matt

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[asterisk-users] Incoming Call Recording

[Rick Hall]

Longtime lurker, first time poster.  :)

A client of mine is in need of having Asterisk record every call that comes
in from a specific incoming route.  I've added the following lines to the
sip_additional.conf file, but no recordings are showing up in the
/var/spool/asterisk/monitor/ folder.

record_out=always
record_in=always

Another page I came across on Google (
http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor) suggested I add the
following line to my sip.conf file:

exten = 2060,3,Monitor(wav,myfilename)

I can see how this could work, but I'm not sure what to replace 2060 with,
as what I need setup is the record of all incoming calls across the board,
not just calls associated with a particular extension number (ie:  2060).

Your sure is appreciated!


-- 
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Senior Vice President
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Affordable Website  Reseller Hosting
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Re: [asterisk-users] Incoming Call Recording

[Danny Nicholas]

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rick Hall
Sent: Friday, June 10, 2011 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Incoming Call Recording

 

Longtime lurker, first time poster.  :)

A client of mine is in need of having Asterisk record every call that comes
in from a specific incoming route.  I've added the following lines to the
sip_additional.conf file, but no recordings are showing up in the
/var/spool/asterisk/monitor/ folder.

record_out=always
record_in=always

Another page I came across on Google
(http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor) suggested I add
the following line to my sip.conf file:

exten = 2060,3,Monitor(wav,myfilename) 

I can see how this could work, but I'm not sure what to replace 2060 with,
as what I need setup is the record of all incoming calls across the board,
not just calls associated with a particular extension number (ie:  2060).

Your sure is appreciated!



-- 

Rick Hall
Senior Vice President
ReadyWire Multimedia Solutions
 
Affordable Website  Reseller Hosting
http://www.readywire.com/
(312) 278-4446 x5446
 
Technical Support:
24 hours a day / 7 days a week
 
Customer Login...: https://secure.readywire.com/
Server Notices.: http://status.readywire.com/
Support Center: https://secure.readywire.com/
Twitter.: http://twitter.com/readywire
Blog: http://blog.readywire.com/

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disseminate, distribute or copy this e-mail. Please notify the sender
immediately by e-mail if you have received this e-mail by mistake and delete
this e-mail from your system. E-mail transmission cannot be guaranteed to be
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destroyed, arrive late or incomplete, or contain viruses. The sender
therefore does not accept liability for any errors or omissions in the
contents of this message, which arise as a result of e-mail transmission. If
verification is required please request a hard-copy version. ReadyWire
Multimedia Solutions. PO BOX 811061, Chicago, IL, USA, 60681.
www.readywire.com.

 

This will do the trick, but you should play the you are being recorded
file to cover your assets

exten = s,n,MixMonitor(Zap_${UNIQUEID}.wav|av(0}V(0))

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[asterisk-users] Incoming SRTP call not working with Bria iPhone Edition

[Alexis de BRUYN]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi Everybody,

I am experiencing some troubles with my Bria iPhone Edition (v. 1.2.8
build 5312, on iOS 4.2.1 iPhone 3G) and Asterisk 1.8.3.2 + TLS/SRTP on
LAN (without NAT).

With 2 computer clients (Blink, one on Mac, one on Windows/Linux),9i can
have a very fine secure conversation in both directions.

When I want to do the same with my iPhone, only outgoing calls are
working. If i try to call (from Blink Win/Mac) my iPhone, Bria is not
ringing. Asterisk logs only said that nobody has picked up :
{{{
  == Using SIP RTP CoS mark 5
-- Executing [400@local:1] Dial(SIP/500-0004, SIP/400,20) in
new stack
  == Using SIP RTP CoS mark 5
-- Called 400
SSL certificate ok
-- Nobody picked up in 2 ms
}}}

My config files are :

 * sip.conf :
{{{
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1 ;none of the others seem to work with Blink as the
client

[400]
type=peer
secret=400 ;note that this is NOT a secure password
host=dynamic
context=local
dtmfmode=rfc2833
disallow=all
allow=g722,gsm
transport=tls
encryption=yes
context=local

[500]
type=peer
secret=500 ;note that this is NOT a secure password
host=dynamic
context=local
dtmfmode=rfc2833
disallow=all
allow=g722,gsm
transport=tls
encryption=yes
context=local
}}}

 * extensions.conf :
{{{
exten = 400,1,Dial(SIP/400,20)
exten = 400,2,VoiceMail(u400@default)
exten = 400,VoiceMail(b400@default)
exten = 400,3,Hangup()
exten = 500,1,Dial(SIP/500,20)
exten = 500,2,VoiceMail(u500@default)
exten = 500,VoiceMail(b500@default)
exten = 500,3,Hangup()
}}}

If I try with a simple SIP (no TLS/SRTP) configuration, the iPhone is
ringing and I can pick up but there is no sound. It is working fine on
the other direction. Network Traversal Strategy is set to Server
Managed (I have tried the others with success).

I have already ask for CounterPath/Bria support, but I didn't have a
positive answer yet.

What is wrong with my settings? Thanks for your help.
- --
Alexis de BRUYN
email : ale...@de-bruyn.fr
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.10 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iEYEARECAAYFAk2VrlAACgkQNy3UyEOc6xUCDwCfVvGO2l80LAJZMn1T4+1UIzcj
ZN8AoJC4o7R6FkrN7jZ2q48hDAWca9nv
=y7JN
-END PGP SIGNATURE-

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[asterisk-users] incoming

[Thomas Perron]

Is it possible to have
Calls incoming to different DIDs?
I want an AA that handles 100s of businesses.

[Incoming-pizza]
Exten = 4045551212,1,Goto(pizza,s,1)

[Incoming-hvac]
Exten = 8085551212,1,Goto(hvac,s,1)

[Incoming-gutter]
Exten = 6175551212,1,Goto(gutter,s,1)

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Re: [asterisk-users] incoming

[Rick Hall]

Yes, I don't see why not.  You just need to setup an IVR for each business
and then assign each individual DID to the appropriate IVR.

This may help:

http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu

Cheers!

Rick

-- 
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Senior Vice President
ReadyWire Multimedia Solutions

Affordable Website  Reseller Hosting
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(312) 278-4446 x5446

Technical Support:
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verification is required please request a hard-copy version. ReadyWire
Multimedia Solutions. PO BOX 811061, Chicago, IL, USA, 60681.
www.readywire.com.



On Sun, Jan 2, 2011 at 11:50 AM, Thomas Perron thomas.per...@gmail.comwrote:

 Is it possible to have
 Calls incoming to different DIDs?
 I want an AA that handles 100s of businesses.

 [Incoming-pizza]
 Exten = 4045551212,1,Goto(pizza,s,1)

 [Incoming-hvac]
 Exten = 8085551212,1,Goto(hvac,s,1)

 [Incoming-gutter]
 Exten = 6175551212,1,Goto(gutter,s,1)

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Re: [asterisk-users] incoming

[Thomas Perron]

Cool.  So, one Asterisk machine handling up to 100 DID numbers, correct?
Yes. I will have unique IVR flows/plans for each.
I assume that the DID mumbers dialed would be the exaxt match needed
to start the respective context.  Correct?

On 1/3/11, Rick Hall r...@readywire.com wrote:
 Yes, I don't see why not.  You just need to setup an IVR for each business
 and then assign each individual DID to the appropriate IVR.

 This may help:

 http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu

 Cheers!

 Rick

 --
 Rick Hall
 Senior Vice President
 ReadyWire Multimedia Solutions

 Affordable Website  Reseller Hosting
 http://www.readywire.com/
 (312) 278-4446 x5446

 Technical Support:
 24 hours a day / 7 days a week

 Customer Login...: https://secure.readywire.com/
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 individual named. If you are not the named addressee you should not
 disseminate, distribute or copy this e-mail. Please notify the sender
 immediately by e-mail if you have received this e-mail by mistake and delete
 this e-mail from your system. E-mail transmission cannot be guaranteed to be
 secure or error-free as information could be intercepted, corrupted, lost,
 destroyed, arrive late or incomplete, or contain viruses. The sender
 therefore does not accept liability for any errors or omissions in the
 contents of this message, which arise as a result of e-mail transmission. If
 verification is required please request a hard-copy version. ReadyWire
 Multimedia Solutions. PO BOX 811061, Chicago, IL, USA, 60681.
 www.readywire.com.



 On Sun, Jan 2, 2011 at 11:50 AM, Thomas Perron
 thomas.per...@gmail.comwrote:

 Is it possible to have
 Calls incoming to different DIDs?
 I want an AA that handles 100s of businesses.

 [Incoming-pizza]
 Exten = 4045551212,1,Goto(pizza,s,1)

 [Incoming-hvac]
 Exten = 8085551212,1,Goto(hvac,s,1)

 [Incoming-gutter]
 Exten = 6175551212,1,Goto(gutter,s,1)

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Re: [asterisk-users] incoming

[Steve Edwards]

On Sun, 2 Jan 2011, Thomas Perron wrote:


Is it possible to have Calls incoming to different DIDs?


Yes*, depending on whether your provider 'provides' the DID in the call 
setup.



*) Better subjects attract more readers. More detail yields better 
answers.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] incoming

[Roger Burton West]

On Mon, Jan 03, 2011 at 02:41:36AM +0400, Thomas Perron wrote:
Cool.  So, one Asterisk machine handling up to 100 DID numbers, correct?

As many as you like, modulo memory and CPU requirements.

I assume that the DID mumbers dialed would be the exaxt match needed
to start the respective context.  Correct?

Depends on how they're presented to you by the DID provider.


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Re: [asterisk-users] incoming

[Steve Edwards]

On Mon, 3 Jan 2011, Thomas Perron wrote:


So, one Asterisk machine handling up to 100 DID numbers, correct?


The number of DIDs is not limited. You could handle a bazillion DIDs with 
a simple dial plan like:


exten = _!.,1,  verbose(1,[${ext...@${context}])
exten = _!.,n,  playback(demo-congrats)
exten = _!.,n,  hangup()


I assume that the DID mumbers dialed would be the exaxt match needed
to start the respective context. Correct?


The exten does not determine which context is started. The provider 
configuration does.


Matching is facilitated by patterns.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Incoming calls through SS7 for data modem transmissions - possible??

[Matt Watson]

Just out of curiosity, what country are you in?

I agree with the others in this thread, this seems very bizzare that the
telco requires you to do SS7 for dialup connections.  I would ask them for
specifics about the legal issues with what you are doing - it sounds to me
like they are just trying to upsell you on a more expensive product.

I am in Canada and we run exactly the configuration you are currently
doing... we still have dialup internet customers that dial into AS5300's via
PRI's.  Our telco has a PRI product gear specifically for this use... they
call it 'ISP-PRI' I'm not entirely sure what the restriction is on it I
have also just kind of assumed that it is inbound calls only, but I've never
tried making outbound calls on them.  I do know they 25-30% cheaper than our
regular voice PRIs though.

--
Matt


2010/11/24 José Pablo Méndez Soto aux...@gmail.com

 Hello,

 We are working on implementing a solution for a medium service provider.
 They were previously using a Cisco AS5300 gateway with some PRI trunks to
 receive modem calls, then route them out the Internet.

 The Telco they were buying the trunks to discovered this configuration and
 restricted them due to legal conventions, and stated that in order to
 continue doing this, they would have to talk SS7 directly.


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Re: [asterisk-users] Incoming calls through SS7 for data modem transmissions - possible??

[Robert Thomas]

Matt,

We are located on Costa Rica and so far there's just 1 TELCO running the
industrym with the CAFTA treatment the carrier had to open for
interconnection but they get to define the ground rules for the
interconnection.

They are arguing ISDN is and end customer circuit and you cannot use it to
resell service. You need an Interconnect circuit, to resell dial up
access, and they require you to support SS7 for Interconnect T1s

It's just the carrier raising the bar waiting for 90% of the competition to
drop off, and a little porcentage to make it.

We asked for quotes for multiple solutions, most of them ranging 15 000$,
and finally come to the conclusion Asterisk and SS7 would be the only viable
option for this project.

Cheers,

On Tue, Nov 30, 2010 at 9:21 AM, Matt Watson m...@mattgwatson.ca wrote:

 Just out of curiosity, what country are you in?

 I agree with the others in this thread, this seems very bizzare that the
 telco requires you to do SS7 for dialup connections.  I would ask them for
 specifics about the legal issues with what you are doing - it sounds to me
 like they are just trying to upsell you on a more expensive product.

 I am in Canada and we run exactly the configuration you are currently
 doing... we still have dialup internet customers that dial into AS5300's via
 PRI's.  Our telco has a PRI product gear specifically for this use... they
 call it 'ISP-PRI' I'm not entirely sure what the restriction is on it I
 have also just kind of assumed that it is inbound calls only, but I've never
 tried making outbound calls on them.  I do know they 25-30% cheaper than our
 regular voice PRIs though.

 --
 Matt


 2010/11/24 José Pablo Méndez Soto aux...@gmail.com

 Hello,

 We are working on implementing a solution for a medium service provider.
 They were previously using a Cisco AS5300 gateway with some PRI trunks to
 receive modem calls, then route them out the Internet.

 The Telco they were buying the trunks to discovered this configuration and
 restricted them due to legal conventions, and stated that in order to
 continue doing this, they would have to talk SS7 directly.


 --
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-- 
Robert
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[asterisk-users] Incoming calls through SS7 for data modem transmissions - possible??

[José Pablo Méndez Soto]

Hello,

We are working on implementing a solution for a medium service provider.
They were previously using a Cisco AS5300 gateway with some PRI trunks to
receive modem calls, then route them out the Internet.

The Telco they were buying the trunks to discovered this configuration and
restricted them due to legal conventions, and stated that in order to
continue doing this, they would have to talk SS7 directly.

We are planning on solving this by placing an Asterisk server with some
TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the
AS5300 for the dial-up to complete after authenticating against a RADIUS
server.

My questions is: can we use only Asterisk to complete/terminate the dial-up
connection, removing the AS5300 out of the picture?

Current topology to be set-up:
Telco -- SS7 -- TE410P-AsteriskServer -- ISDN -- AS5300 -- Internet

Ideal topology:
Telco -- SS7 -- TE410P-AsteriskServer -- Internet


Some posts talk about zapRAS being able to accomplish this, not quite sure
though

Sounds like possible:
http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.htmlasterisk-users@lists.digium.com

Sounds like not possible:
http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html


Thanks in advance,


*José Pablo Méndez
  *
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Re: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible??

[Cary Fitch]

I am not sure where you are and what legal conventions are involved.

 

Are you saying the Telco (and legal restrictions) say you can’t send calls
to the internet via the AS5300 but you can if Asterisk does it directly?
What is the “logic” in that?

 

Or are they saying your Telco to Asterisk trunks have to be SS7 controlled? 

 

Or are you concerned about Asterisk handling the TDM to IP conversion in an
adequate manner?

 

I am not sure/aware myself that Asterisk will do a modem to IP conversion.
I think in your example the AS5300 is doing that.

 

What is the Telco’s problem in doing what the customer was doing before?

 

Feel free to correspond directly if you want to.

 

Cary Fitch

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of José Pablo
Méndez Soto
Sent: Wednesday, November 24, 2010 7:31 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Incoming calls through SS7 for data
modemtransmissions - possible??

 

Hello,

We are working on implementing a solution for a medium service provider.
They were previously using a Cisco AS5300 gateway with some PRI trunks to
receive modem calls, then route them out the Internet.

The Telco they were buying the trunks to discovered this configuration and
restricted them due to legal conventions, and stated that in order to
continue doing this, they would have to talk SS7 directly.

We are planning on solving this by placing an Asterisk server with some
TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the
AS5300 for the dial-up to complete after authenticating against a RADIUS
server.

My questions is: can we use only Asterisk to complete/terminate the dial-up
connection, removing the AS5300 out of the picture?

Current topology to be set-up:
Telco -- SS7 -- TE410P-AsteriskServer -- ISDN -- AS5300 -- Internet

Ideal topology:
Telco -- SS7 -- TE410P-AsteriskServer -- Internet


Some posts talk about zapRAS being able to accomplish this, not quite sure
though

Sounds like possible:
http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.html
mailto:asterisk-users@lists.digium.com 

Sounds like not possible:
http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html


Thanks in advance,


José Pablo Méndez
  

-- 
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Re: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible??

[José Pablo Méndez Soto]

Thanks Cary,

What happens is, the Telco won't allow the small company to resell the ISDN
connections, meaning, they bought the trunks and DIDs, then sold dialing
plans to route incoming calls through the PRIs out the Internet. This is not
the issue though. We definitely have to migrate to an SS7 capable platform,
because that is the only way the Telco allows them to resell the dial-up
connections (not ISDN), and Asterisk is the current bet.

If we can get Asterisk to pick up those calls via SS7, then authenticate
them, send them out to the Internet, we would be achieving a %100 usage on
the Digium cards, because one of them wouldn't be used to talk to the AS.

Can Asterisk do this?


Thanks again,

*José Pablo Méndez
   *


On Wed, Nov 24, 2010 at 7:59 PM, Cary Fitch ca...@usawide.net wrote:

  I am not sure where you are and what legal conventions are involved.



 Are you saying the Telco (and legal restrictions) say you can’t send calls
 to the internet via the AS5300 but you can if Asterisk does it directly?
 What is the “logic” in that?



 Or are they saying your Telco to Asterisk trunks have to be SS7 controlled?




 Or are you concerned about Asterisk handling the TDM to IP conversion in an
 adequate manner?



 I am not sure/aware myself that Asterisk will do a modem to IP conversion.
 I think in your example the AS5300 is doing that.



 What is the Telco’s problem in doing what the customer was doing before?



 Feel free to correspond directly if you want to.



 Cary Fitch


  --

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *José Pablo Méndez
 Soto
 *Sent:* Wednesday, November 24, 2010 7:31 PM
 *To:* asterisk-users@lists.digium.com
 *Subject:* [asterisk-users] Incoming calls through SS7 for data
 modemtransmissions - possible??



 Hello,

 We are working on implementing a solution for a medium service provider.
 They were previously using a Cisco AS5300 gateway with some PRI trunks to
 receive modem calls, then route them out the Internet.

 The Telco they were buying the trunks to discovered this configuration and
 restricted them due to legal conventions, and stated that in order to
 continue doing this, they would have to talk SS7 directly.

 We are planning on solving this by placing an Asterisk server with some
 TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the
 AS5300 for the dial-up to complete after authenticating against a RADIUS
 server.

 My questions is: can we use only Asterisk to complete/terminate the dial-up
 connection, removing the AS5300 out of the picture?

 Current topology to be set-up:
 Telco -- SS7 -- TE410P-AsteriskServer -- ISDN -- AS5300 -- Internet

 Ideal topology:
 Telco -- SS7 -- TE410P-AsteriskServer -- Internet


 Some posts talk about zapRAS being able to accomplish this, not quite sure
 though

 Sounds like possible:
 http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.htmlasterisk-users@lists.digium.com

 Sounds like not possible:
 http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html


 Thanks in advance,


 *José Pablo Méndez**
   *

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 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] Incoming calls through SS7 for datamodemtransmissions - possible??

[José Pablo Méndez Soto]

Thanks Cary,

The first topology we are working on should be the best way then.

Asterisk will answer SS7 calls, route them to the ISDN channels to be
terminated by the AS5300 as they were doing before. I think TDM-2-TDM
shouldn't be that much of a problem eh? No further equipment needed?


*José Pablo Méndez
   *


2010/11/24 Cary Fitch ca...@usawide.net

  I understand the problem.  You can’t resell PRI connections.



 I don’t think Asterisk can convert TDM to IP.  It does convert TDM to SIP
 which is then sent out over IP.What you want to do is have it do the
 TDM/Modem conversion without the PRIs and Cisco Gear.



 There used to be a way to do this, and maybe still is but not just with
 Asterisk perhaps.



 I know that Ascend/Lucent TNTs (and I am sure some other equipment)  could
 take TDM trunks, which could be SS7 trunks, and convert them to IP.



 The point in this is that they are SS7 based.  You can take SS7 trunks from
 either the Asterisk box or direct from the Telco and convert them to IP.



 NO PRIs involved.  Yes, more “telco grade carrier equipment” but no PRIs.



 A lot of this equipment was available by the pound a few years back.



 Cary


  --

 *From:* José Pablo Méndez Soto [mailto:aux...@gmail.com]
 *Sent:* Wednesday, November 24, 2010 8:34 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Cc:* ca...@usawide.net
 *Subject:* Re: [asterisk-users] Incoming calls through SS7 for
 datamodemtransmissions - possible??



 Thanks Cary,

 What happens is, the Telco won't allow the small company to resell the ISDN
 connections, meaning, they bought the trunks and DIDs, then sold dialing
 plans to route incoming calls through the PRIs out the Internet. This is not
 the issue though. We definitely have to migrate to an SS7 capable platform,
 because that is the only way the Telco allows them to resell the dial-up
 connections (not ISDN), and Asterisk is the current bet.

 If we can get Asterisk to pick up those calls via SS7, then authenticate
 them, send them out to the Internet, we would be achieving a %100 usage on
 the Digium cards, because one of them wouldn't be used to talk to the AS.

 Can Asterisk do this?


 Thanks again,

 *José Pablo Méndez**
*

  On Wed, Nov 24, 2010 at 7:59 PM, Cary Fitch ca...@usawide.net wrote:

 I am not sure where you are and what legal conventions are involved.



 Are you saying the Telco (and legal restrictions) say you can’t send calls
 to the internet via the AS5300 but you can if Asterisk does it directly?
 What is the “logic” in that?



 Or are they saying your Telco to Asterisk trunks have to be SS7 controlled?




 Or are you concerned about Asterisk handling the TDM to IP conversion in an
 adequate manner?



 I am not sure/aware myself that Asterisk will do a modem to IP conversion.
 I think in your example the AS5300 is doing that.



 What is the Telco’s problem in doing what the customer was doing before?



 Feel free to correspond directly if you want to.



 Cary Fitch


  --

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *José Pablo Méndez
 Soto
 *Sent:* Wednesday, November 24, 2010 7:31 PM
 *To:* asterisk-users@lists.digium.com
 *Subject:* [asterisk-users] Incoming calls through SS7 for data
 modemtransmissions - possible??



 Hello,

 We are working on implementing a solution for a medium service provider.
 They were previously using a Cisco AS5300 gateway with some PRI trunks to
 receive modem calls, then route them out the Internet.

 The Telco they were buying the trunks to discovered this configuration and
 restricted them due to legal conventions, and stated that in order to
 continue doing this, they would have to talk SS7 directly.

 We are planning on solving this by placing an Asterisk server with some
 TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the
 AS5300 for the dial-up to complete after authenticating against a RADIUS
 server.

 My questions is: can we use only Asterisk to complete/terminate the dial-up
 connection, removing the AS5300 out of the picture?

 Current topology to be set-up:
 Telco -- SS7 -- TE410P-AsteriskServer -- ISDN -- AS5300 -- Internet

 Ideal topology:
 Telco -- SS7 -- TE410P-AsteriskServer -- Internet


 Some posts talk about zapRAS being able to accomplish this, not quite sure
 though

 Sounds like possible:
 http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.htmlasterisk-users@lists.digium.com

 Sounds like not possible:
 http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html


 Thanks in advance,


 *José Pablo Méndez**
   *


 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs

[asterisk-users] Incoming calls

[Flavio Miranda]

Hi all,
  I'd like that each analog trunk of my TDM410p was received in different 
extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each trunk in a 
different context and in my extensions.conf, under [default] I put such 
contexts and an especific estension to answer it. therefore, when I get  call, 
it always is ringing on the first extensions, dont matter trunk  . Anybody 
could teach me how can I organize that ?
 Att,

 

Flavio Roberto Miranda

MSN:flaviormira...@hotmail.com
Skype: flaviormiranda

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Re: [asterisk-users] Incoming calls

[Steve Edwards]

On Thu, 18 Nov 2010, Flavio Miranda wrote:

 I'd like that each analog trunk of my TDM410p was received in different 
 extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each 
 trunk in a different context and in my extensions.conf, under [default] 
 I put such contexts and an especific estension to answer it. therefore, 
 when I get call, it always is ringing on the first extensions, dont 
 matter trunk. Anybody could teach me how can I organize that ?

0) Use a subject that gives a clue what you're looking for. Almost 
everybody has had a question about an incomig call at some point in time.
Better bait = better fish.

1) It sounds like you have a clue about how to do it and are on the right 
track.

2) Including some details like the console output from:

zap show channel 1 (I'm a 1.2 Luddite.)
zap show channel 2
zap show channel 3
zap show channel 4

as well as the console log from a call coming in on each channel

will help in assisting you in resolving this issue.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Incoming calls

[Flavio Miranda]

Hi Steve,
thanks for the tips Better bait = better fish !

As you said, I  am in the right track.
Looking to dahdi show channles , I realized  that all the trunks was in the 
same context. So, I have changed  this and everything works!
thanks you !!
Att,

 

Flavio Roberto Miranda

MSN:flaviormira...@hotmail.com
Skype: flaviormiranda



 Date: Thu, 18 Nov 2010 11:53:26 -0800
 From: asterisk@sedwards.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Incoming calls
 
 On Thu, 18 Nov 2010, Flavio Miranda wrote:
 
  I'd like that each analog trunk of my TDM410p was received in different 
  extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each 
  trunk in a different context and in my extensions.conf, under [default] 
  I put such contexts and an especific estension to answer it. therefore, 
  when I get call, it always is ringing on the first extensions, dont 
  matter trunk. Anybody could teach me how can I organize that ?
 
 0) Use a subject that gives a clue what you're looking for. Almost 
 everybody has had a question about an incomig call at some point in time.
 Better bait = better fish.
 
 1) It sounds like you have a clue about how to do it and are on the right 
 track.
 
 2) Including some details like the console output from:
 
 zap show channel 1 (I'm a 1.2 Luddite.)
 zap show channel 2
 zap show channel 3
 zap show channel 4
 
 as well as the console log from a call coming in on each channel
 
 will help in assisting you in resolving this issue.
 
 -- 
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000
 
 -- 
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Incoming calls

[Steve Edwards]

On Thu, 18 Nov 2010, Flavio Miranda wrote:

Looking to dahdi show channles , I realized  that all the trunks was in 
the same context. So, I have changed  this and everything works!


That's why I prefer to work from what Asterisk parsed the file as, not 
what the poster thinks :)


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000-- 
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[asterisk-users] Incoming calls

[Flavio Miranda]

Hi all,

   After a lot of trouble with a TE110p working with mfcr2 , brazil variant, 
everything looks great,but I can not go out of my calls.
When I try I receive the following  log:
== Using SIP RTP CoS mark 5-- Executing [33220...@local:1] 
Dial(SIP/4804-001a, DAHDI/g11/33220567,,T) in new stack  == Everyone is 
busy/congested at this time (1:0/1/0)-- Auto fallthrough, channel 
'SIP/4804-001a' status is 
'CONGESTION'This
 is  my dahdi show status:Digium Wildcard TE110P T1/E1 Card 0  REC 0
  0  0  CAS HDB3 CRC4 0 db (CSU)/0-133 feet 
(DSX-1)thi´s
 my dahdi show channels:
asterisk*CLI dahdi show channels   Chan Extension  Context Language   
MOH InterpretBlockedState  pseudodefault
default In Service  1 4800   default
default In Service  2 4800   
defaultdefault In Service  3 
4805   defaultdefault In 
Service  4defaultdefault
 In Service  5defaultdefault
 In Service  6defaultdefault
 In Service  7default
default In Service  8default
default In Service  9default
default In Service 10
defaultdefault In Service 11
defaultdefault In Service   
  12defaultdefault In 
Service 13defaultdefault
 In Service 14defaultdefault
 In Service 15defaultdefault
 In Service 17default
default In Service 18default
default In Service 19default
default In Service 20
defaultdefault In Service 21
defaultdefault In Service   
  22defaultdefault In 
Service 23defaultdefault
 In Service 24defaultdefault
 In Service 25defaultdefault
 In Service 26default
default In Service 27default
default In Service 28default
default In Service 29
defaultdefault In Service 30
defaultdefault In Service   
  31defaultdefault In 
Service
*In
 my incoming call , the log is:
MFC/R2 call offered on chan 1. ANI = 1221341400, DNIS = 4804, Category = 
National SubscriberNew MFC/R2 call detected on chan 2.

and  don't ring nowhere!
Thanks for help!Att,

 

Flavio Roberto Miranda

MSN:flaviormira...@hotmail.com
Skype: flaviormiranda

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[asterisk-users] incoming call FXO

[Flavio Miranda]

Hi all,
  Recently I  have instaled one Digium TDM410 on my Asterisk. After instaled ,  
I can do outgoing calls but I  cant receive calls. I receive the following 
messages:
chan_dahdi.c: Got event 2 (Ring/Answered)...[Sep 14 11:24:44] NOTICE[2654] 
chan_dahdi.c: Got event 18 (Ring Begin)...[Sep 14 11:24:44] WARNING[2654] 
pbx.c: Channel 'DAHDI/4-1' sent into invalid extension 's' in context 
'default', but no invalid handler
I have not this 's' extension.
Anybody knows what happen?
Att,

 

Flavio Roberto Miranda

MSN:flaviormira...@hotmail.com
Skype: flaviormiranda

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Re: [asterisk-users] incoming call FXO

[Kevin P. Fleming]

On 09/15/2010 07:20 AM, Flavio Miranda wrote:

   Recently I  have instaled one Digium TDM410 on my Asterisk. After
 instaled ,  I can do outgoing calls but I  cant receive calls. I receive
 the following messages:
 
 chan_dahdi.c: Got event 2 (Ring/Answered)...
 [Sep 14 11:24:44] NOTICE[2654] chan_dahdi.c: Got event 18 (Ring Begin)...
 [Sep 14 11:24:44] WARNING[2654] pbx.c: Channel 'DAHDI/4-1' sent into
 invalid extension 's' in context 'default', but no invalid handler
 
 I have not this 's' extension.

Right, that's what the message is telling you. For incoming calls on
FXO, they can *only* be sent to the 's' extension in the target context,
since there is no target number passed over the FXO connection. You'll
have to create an 's' extension to handle incoming calls however you like.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] incoming call FXO

[Zeeshan Zakaria]

As Kevin said, you need to define an 's' extension where the calls will be
answered. Seems like you are using default configuration. Open file
'extensions.conf' in /etc/asterisk folder and look for context named
[default]. If it is not there, create one and add something under it, e.g.,

[default]
exten = s,1,Verbose( - - - Call received - - - )
exten = s,n,Playback(hello-world)
extent = s,n,HangUp()

Then do a 'core extensions reload' on Asterisk CLI. Now calling in on FXO
should play the message 'hello-world' (assuming this sound file exists in
the sound folder of asterisk), and you'll see the call activity on the CLI.

For the rest, you need to consult chapters 5 and 6 of 'Asterisk-The Future
of Telephony' book.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-09-15 8:59 AM, Kevin P. Fleming kpflem...@digium.com wrote:

On 09/15/2010 07:20 AM, Flavio Miranda wrote:

 Recently I have instaled one Digium TDM410 on my...
Right, that's what the message is telling you. For incoming calls on
FXO, they can *only* be sent to the 's' extension in the target context,
since there is no target number passed over the FXO connection. You'll
have to create an 's' extension to handle incoming calls however you like.

--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] incoming call FXO

[Flavio Miranda]

Ok. Problem solved . 
Thank you very much!!!

Att,

 

Flavio Roberto Miranda

MSN:flaviormira...@hotmail.com
Skype: flaviormiranda



Date: Wed, 15 Sep 2010 09:56:36 -0400
From: zisha...@gmail.com
To: kpflem...@digium.com; asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] incoming call FXO

As Kevin said, you need to define an 's' extension where the calls will be 
answered. Seems like you are using default configuration. Open file 
'extensions.conf' in /etc/asterisk folder and look for context named [default]. 
If it is not there, create one and add something under it, e.g.,


[default]

exten = s,1,Verbose( - - - Call received - - - )

exten = s,n,Playback(hello-world)

extent = s,n,HangUp()

Then do a 'core extensions reload' on Asterisk CLI. Now calling in on FXO 
should play the message 'hello-world' (assuming this sound file exists in the 
sound folder of asterisk), and you'll see the call activity on the CLI.


For the rest, you need to consult chapters 5 and 6 of 'Asterisk-The Future of 
Telephony' book.

Zeeshan A Zakaria

--

www.ilovetovoip.com


On 2010-09-15 8:59 AM, Kevin P. Fleming kpflem...@digium.com wrote:

On 09/15/2010 07:20 AM, Flavio Miranda wrote:


   Recently I  have instaled one Digium TDM410 on my...
Right, that's what the message is telling you. For incoming calls on

FXO, they can *only* be sent to the 's' extension in the target context,

since there is no target number passed over the FXO connection. You'll

have to create an 's' extension to handle incoming calls however you like.



--

Kevin P. Fleming

Digium, Inc. | Director of Software Technologies

445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

skype: kpfleming | jabber: kflem...@digium.com

Check us out at www.digium.com  www.asterisk.org



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[asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used

[Joe Wood]

Hello.

I have been beating my head over this problem for about 6 hours now.

I have a SIP peer, who I register to (successfully), who should be
directing all incoming calls at my [default] stanza in my
extensions.conf:

[ Context 'default' created by 'pbx_config' ]
  's' =1. Wait(1)[pbx_config]
2. Answer()   [pbx_config]
3. Background(welcome)[pbx_config]
4. Background(and)[pbx_config]
5. Background(thank-you-for-calling)  [pbx_config]
6. Background(conference-reservations)[pbx_config]
7. Waitfor()  [pbx_config]
8. Hangup()   [pbx_config]

Unfortunately, no matter how I configure extensions.conf or sip.conf,
the phone call always ends up saying: Extension is unavailable.
Please leave your message after the tone.

sip.conf:

[general]
register = NPANXX:passw...@service_provider_ip
registertimeout=29
registerattempts=0
defaultexpiry=60

[DID_NUMBER]
type=peer
context=default
host=SERVICE_PROVIDER_IP
authuser=DID_NUMBER
fromuser=DID_NUMBER
fromdomain=SERVICE_PROVIDER_REALM
remotesecret=SERVICE_PROVIDER_PASSWD
secret=SERVICE_PROVIDER_PASSWD
dtmfmode=rfc2833
disallow=all
allow=ulaw
qualify=yes

I am attempting just to get the starting point where I can direct
users through my asterisk box, but it won't direct users to the 's'
extention, only to some voicemail box. I've removed the voicemail
config.

My extensions.conf is tiny:

[globals]

[general]

[default]
exten = s,1,Wait(1)
exten = s,n,Answer()
exten = s,n,Background(welcome)
exten = s,n,Background(and)
exten = s,n,Background(thank-you-for-calling)
exten = s,n,Background(conference-reservations)
exten = s,n,Waitfor()
exten = s,n,Hangup()


What am I doing wrong here?



Thanks for any help you can give.


Joe

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Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used

[Warren Selby]

On Wed, Aug 4, 2010 at 8:52 PM, Joe Wood sch...@gmail.com wrote:

 Hello.

 I have been beating my head over this problem for about 6 hours now.

 I have a SIP peer, who I register to (successfully), who should be
 directing all incoming calls at my [default] stanza in my
 extensions.conf:

 [ Context 'default' created by 'pbx_config' ]
  's' =1. Wait(1)
  [pbx_config]
2. Answer()
 [pbx_config]
3. Background(welcome)
  [pbx_config]
4. Background(and)
  [pbx_config]
5. Background(thank-you-for-calling)
  [pbx_config]
6. Background(conference-reservations)
  [pbx_config]
7. Waitfor()
  [pbx_config]
8. Hangup()
 [pbx_config]

 Unfortunately, no matter how I configure extensions.conf or sip.conf,
 the phone call always ends up saying: Extension is unavailable.
 Please leave your message after the tone.

 sip.conf:

 [general]
 register = NPANXX:passw...@service_provider_ip
 registertimeout=29
 registerattempts=0
 defaultexpiry=60

 [DID_NUMBER]
 type=peer
 context=default
 host=SERVICE_PROVIDER_IP
 authuser=DID_NUMBER
 fromuser=DID_NUMBER
 fromdomain=SERVICE_PROVIDER_REALM
 remotesecret=SERVICE_PROVIDER_PASSWD
 secret=SERVICE_PROVIDER_PASSWD
 dtmfmode=rfc2833
 disallow=all
 allow=ulaw
 qualify=yes

 I am attempting just to get the starting point where I can direct
 users through my asterisk box, but it won't direct users to the 's'
 extention, only to some voicemail box. I've removed the voicemail
 config.

 My extensions.conf is tiny:

 [globals]

 [general]

 [default]
 exten = s,1,Wait(1)
 exten = s,n,Answer()
 exten = s,n,Background(welcome)
 exten = s,n,Background(and)
 exten = s,n,Background(thank-you-for-calling)
 exten = s,n,Background(conference-reservations)
 exten = s,n,Waitfor()
 exten = s,n,Hangup()


 What am I doing wrong here?



 Thanks for any help you can give.


 Joe


You don't have any extensions in your default context that match the
extension that your sip peer is dialing in on.  's' is not a default
extension for SIP...try using _X., and see what you get.  Bump up the CLI
(core set verbose 10) and then repost a failed called attempt.  Some SIP
providers also use a + symbol in front of their inbound calls, so you may
need to use _+X., instead.


-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used

[Joe Wood]

I don't see any

On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby wcse...@selbytech.com wrote:

 You don't have any extensions in your default context that match the
 extension that your sip peer is dialing in on.  's' is not a default
 extension for SIP...try using _X., and see what you get.  Bump up the CLI
 (core set verbose 10) and then repost a failed called attempt.  Some SIP
 providers also use a + symbol in front of their inbound calls, so you may
 need to use _+X., instead.

I don't see any call attempt/logs when I bump up the verbosity, and
when I check my verbose logs I show:

[Aug  4 19:16:42] VERBOSE[12287] pbx.c: -- Registered extension
context 'default' (0xb77980c0) in local table 0xb77960c0; registrar:
pbx_config
[Aug  4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.'
priority 1 to default (0xb77980c0)
[Aug  4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.'
priority 2 to default (0xb77980c0)
[Aug  4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.'
priority 3 to default (0xb77980c0)
[Aug  4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.'
priority 4 to default (0xb77980c0)
[Aug  4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.'
priority 5 to default (0xb77980c0)
[Aug  4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.'
priority 6 to default (0xb77980c0)
[Aug  4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.'
priority 7 to default (0xb77980c0)
[Aug  4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.'
priority 8 to default (0xb77980c0)
[Aug  4 19:16:42] VERBOSE[12287] pbx.c: -- Registered extension
context 'parkedcalls' (0xb7797ee0) in local table 0xb77960c0;
registrar: features
[Aug  4 19:16:42] VERBOSE[12287] pbx.c: -- merging
incls/swits/igpats from old(parkedcalls) to new(parkedcalls) context,
registrar = pbx_config
[Aug  4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '700'
priority 1 to parkedcalls (0xb7797ee0)
[Aug  4 19:16:42] VERBOSE[12287] pbx.c: -- Time to scan old
dialplan and merge leftovers back into the new: 0.89 sec
[Aug  4 19:16:42] VERBOSE[12287] pbx.c: -- Time to restore hints
and swap in new dialplan: 0.02 sec
[Aug  4 19:16:42] VERBOSE[12287] pbx.c: -- Time to delete the old
dialplan: 0.11 sec
[Aug  4 19:16:42] VERBOSE[12287] pbx.c: -- Total time
merge_contexts_delete: 0.000102 sec
[Aug  4 19:17:04] VERBOSE[12255] netsock.c:   == Using SIP RTP CoS mark 5
[Aug  4 19:19:04] VERBOSE[12255] netsock.c:   == Using SIP RTP CoS mark 5
[Aug  4 19:21:39] VERBOSE[12255] netsock.c:   == Using SIP RTP CoS mark 5

I get the same error. Same random voicemail when no voicemail is configured.

I was under the impressing that s was the catchall for all incoming
trunks. What has changed?

Joe

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Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used

[Warren Selby]

On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood sch...@gmail.com wrote:

 I don't see any

 On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby wcse...@selbytech.com
 wrote:
 
  You don't have any extensions in your default context that match the
  extension that your sip peer is dialing in on.  's' is not a default
  extension for SIP...try using _X., and see what you get.  Bump up the CLI
  (core set verbose 10) and then repost a failed called attempt.  Some SIP
  providers also use a + symbol in front of their inbound calls, so you may
  need to use _+X., instead.

 I don't see any call attempt/logs when I bump up the verbosity, and
 when I check my verbose logs I show:


The next step would be to enable sip debug on the peer you're trying to
receive calls from (sip set debug peer PEERNAME or sip set debug ip
IPADDRESS).  Then send another call inbound and see what's happening.  As
far as the 's' extension, that's the start extension, it's used when no
other extension information is presented.  Pretty much every SIP peer I've
ever seen presents an extension when entering a context, and thus the 's'
extension doesn't catch it.  I've typically only seen 's' used in Macros and
with inbound analog lines.

-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used

[Joe Wood]

On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby wcse...@selbytech.com wrote:
 On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood sch...@gmail.com wrote:

 I don't see any

 On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby wcse...@selbytech.com
 wrote:
 
  You don't have any extensions in your default context that match the
  extension that your sip peer is dialing in on.  's' is not a default
  extension for SIP...try using _X., and see what you get.  Bump up the
  CLI
  (core set verbose 10) and then repost a failed called attempt.  Some SIP
  providers also use a + symbol in front of their inbound calls, so you
  may
  need to use _+X., instead.

 I don't see any call attempt/logs when I bump up the verbosity, and
 when I check my verbose logs I show:


 The next step would be to enable sip debug on the peer you're trying to
 receive calls from (sip set debug peer PEERNAME or sip set debug ip
 IPADDRESS).  Then send another call inbound and see what's happening.  As
 far as the 's' extension, that's the start extension, it's used when no
 other extension information is presented.  Pretty much every SIP peer I've
 ever seen presents an extension when entering a context, and thus the 's'
 extension doesn't catch it.  I've typically only seen 's' used in Macros and
 with inbound analog lines.


My experience with Asterisk in the past has been with inbound analog
lines so that would make sense :)

See if you spot anything weird here:

--- SIP read from UDP:209.221.186.98:5060 ---
INVITE sip:s...@209.221.186.50 SIP/2.0
Record-Route: sip:209.221.186.98;ftag=f7093e2d7e16a927d0816f6f5ed7aba4;lr
Via: SIP/2.0/UDP
209.221.186.98;branch=z9hG4bK6b95.ae84c9620e19761029084a0f85255c29.0
Via: SIP/2.0/UDP
209.221.186.98:5071;branch=z9hG4bKf0a046579186bf058ba7783ed98a5a66;rport=5071
Max-Forwards: 16
From: 2538544199
sip:2538544...@209.221.186.98;tag=f7093e2d7e16a927d0816f6f5ed7aba4
To: sip:2063161...@209.221.186.98
Call-ID: ba711218-1ae3-122e-41bd-18a905721e58-b2b_1
CSeq: 200 INVITE
Contact: Anonymous sip:2538544...@209.221.186.98:5071
Expires: 300
User-Agent: Sippy Softswitch v2.0.80
cisco-GUID: 1225641884-3786690633-966044271-4144140181
h323-conf-id: 1225641884-3786690633-966044271-4144140181
Content-Length: 321
Content-Type: application/sdp

v=0
o=- 1280279699622 1280279699622 IN IP4 209.221.186.98
s=-
c=IN IP4 209.221.186.98
t=0 0
m=audio 60304 RTP/AVP 0
a=fmtp:4 bitrate=6300;annexa=no
a=rtpmap:96 iLBC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=fmtp:18 annexb=no
a=rtpmap:98 telephone-event/8000
a=oldmediaip:208.76.155.20
a=nortpproxy:yes

-
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:  Header
 0 [ 35]: INVITE sip:s...@209.221.186.50 SIP/2.0
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:  Header
 1 [ 75]: Record-Route:
sip:209.221.186.98;ftag=f7093e2d7e16a927d0816f6f5ed7aba4;lr
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:  Header
 2 [ 85]: Via: SIP/2.0/UDP
209.221.186.98;branch=z9hG4bK6b95.ae84c9620e19761029084a0f85255c29.0
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:  Header
 3 [ 94]: Via: SIP/2.0/UDP
209.221.186.98:5071;branch=z9hG4bKf0a046579186bf058ba7783ed98a5a66;rport=5071
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:  Header
 4 [ 16]: Max-Forwards: 16
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:  Header
 5 [ 85]: From: 2538544199
sip:2538544...@209.221.186.98;tag=f7093e2d7e16a927d0816f6f5ed7aba4
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:  Header
 6 [ 35]: To: sip:2063161...@209.221.186.98
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:  Header
 7 [ 51]: Call-ID: ba711218-1ae3-122e-41bd-18a905721e58-b2b_1
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:  Header
 8 [ 16]: CSeq: 200 INVITE
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:  Header
 9 [ 55]: Contact: Anonymous sip:2538544...@209.221.186.98:5071
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:  Header
10 [ 12]: Expires: 300
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:  Header
11 [ 36]: User-Agent: Sippy Softswitch v2.0.80
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:  Header
12 [ 54]: cisco-GUID: 1225641884-3786690633-966044271-4144140181
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:  Header
13 [ 56]: h323-conf-id: 1225641884-3786690633-966044271-4144140181
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:  Header
14 [ 19]: Content-Length: 321
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:  Header
15 [ 29]: Content-Type: application/sdp
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:  Header
16 [  0]:
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body
 0 [  3]: v=0
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body
 1 [ 53]: o=- 1280279699622 1280279699622 IN IP4 209.221.186.98
[Aug  4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:   

Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used

[Warren Selby]

On Wed, Aug 4, 2010 at 10:25 PM, Joe Wood sch...@gmail.com wrote:

 On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby wcse...@selbytech.com
 wrote:
  On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood sch...@gmail.com wrote:
 

 My experience with Asterisk in the past has been with inbound analog
 lines so that would make sense :)

 See if you spot anything weird here:


Try adding insecure=invite to the DID_NUMBER peer, reload SIP and try your
call again.  By the way, it looks like your SIP provider has a built-in
auto-failover to voicemail setup.  You may want to get them to disable that
once you get everything working on your end.

-- 
Thanks,
--Warren Selby
http://www.selbytech.com
-- 
_
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Re: [asterisk-users] Incoming call doesn't finish when internal phone hangs up

[Harel Cohen]

Hi Elder,
I would first check the behaviour of your PSTN lines (i.e. nothing to do with 
Asterisk). In many places PSTN companies allow between 30 to 90 seconds of 
connection to remain open even if the -called- party, NOT the calling party, 
has hung-up. Normally this is to allow putting down the phone in one room and 
picking up in another room without disconnecting the line. Make a small test to 
verify this and if this is the case you will need to discuss this with your 
PSTN provider.
Harel

Date: Thu, 8 Jul 2010 12:01:40 -0500
From: Daniel - Asterisk earohua...@gmail.com
Subject: [asterisk-users] Incoming call doesn't finish when internal
phone   hangs up
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
aanlktikxafnxhbsws0ov4u5ht3yjbeevuh26vehrg...@mail.gmail.com
Content-Type: text/plain; charset=ISO-8859-1

Hello guys,

I have this problem when a call is received in my PBX:

(Caller) -- (Redirecting Service) -- (E1 PRI) -- (Asterisk PBX) -- 
(Internal Phone)

Reception works fine, but when conversation finishes and the agent at internal 
phone hangs up, the call at caller's side is still alive for many seconds until 
it hangs up.

The problem is that Telephone Company is billing me acording Caller's duration 
which is bigger than Asterisk's CDR. The same issue happens when Caller dials 
E1 PRI directly. In every case Asterisk finishes normally the call as CDR and 
CLI register correctly.

I'm using Asterisk 1.4.21.2 and OpenVox DE210P card. Configuration files follow:

zaptel.conf:
span=1,1,1,ccs,hdb3
bchan=1-15,17-31
dchan=16

# Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2
span=2,2,1,ccs,hdb3
bchan=32-46,48-62
dchan=47

# Global data
loadzone= us
defaultzone = us


zapata.conf:
[channels]
language=es
context=default
rxwink=300
usecallerid=yes
hidecallerid=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
busydetect=yes
busycount=yes
busypattern=500,500
answeronpolarityswitch=yes
hanguponpolarityswitch=yes

;PRI RDSI - SPAN 1
group = 1
context = incoming-1
inmediate=no
switchtype=euroisdn
signalling=pri_cpe
channel = 1-15,17-31

;PRI RDSI - SPAN 2
group = 1
context = incoming-2
inmediate=no
switchtype=euroisdn
signalling=pri_cpe
channel = 32-46,48-62
...

Thanks in advance,

Elder Arohuanca Lagos
Phone: +51 1 991696900
Lima - Peru



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