Re: [asterisk-users] Question on the RTP packet header

[Mark Murawski]

Hi Dan,

Your best bet for looking at RTP media specifics is the standards that 
define RTP.


Wikipedia has some really good resources on RTP and a list of the 
various RFC standards that relate:

https://en.wikipedia.org/wiki/Real-time_Transport_Protocol



On 8/28/23 11:16, Dan Cropp wrote:


I am working on a project that uses Asterisk ARI ExternalMedia request 
to stream the RTP audio from Asterisk to an UDP/RTP receiver project.


Using slin16 format.

1) I believe I am seeing is a 12-byte header followed by 640 bytes of 
data.  Is this correct?


2) Is there some place I can find a description of the 12-byte packet 
header fields?


Dan


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Re: [asterisk-users] Question about Sip Trunks who support Stir Shaken

[Dirk-Willem van Gulik]

> On 18 Aug 2023, at 04:50, Federico  wrote:
> 
> I am looking for a decent provider of SIP Trunks but it has to pass the Stir 
> Shaken token to the next carrier. Does anybody know about any? Sipstation 
> from Sangoma, does not support Stir Shaken. ( Case #01466843 / 
> 001300G8PLG / MAIN / Open [ ref:_00D306mPe._5004U1BlBLF:ref ])
> 
> 
> 
I’d try Telnyx - where this works for me. 

And their online SIP debugging tool is second to none. Absolutely excels at 
finding issues with this sort of stuff. 

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Re: [asterisk-users] Question about Sip Trunks who support Stir Shaken

[Federico]

Thanks. I have accounts with both companies and both have issues.

From: asterisk-users  On Behalf Of 
Dovid Bender
Sent: Friday, August 18, 2023 11:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Question about Sip Trunks who support Stir Shaken

 

Telnyx, 382com, voicetel and as others mentioned BandWidth. I have contacts at 
382 and voicetel if you want an intro.

 

 

On Thu, Aug 17, 2023 at 11:50 PM Federico mailto:feder...@digitalipvoice.com> > wrote:

I am looking for a decent provider of SIP Trunks but it has to pass the Stir 
Shaken token to the next carrier. Does anybody know about any? Sipstation from 
Sangoma, does not support Stir Shaken. ( Case #01466843 / 001300G8PLG / 
MAIN / Open [ ref:_00D306mPe._5004U1BlBLF:ref ])

Although it’s mandatory, somehow they think it’s ok. Go figure.

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Re: [asterisk-users] Question about Sip Trunks who support Stir Shaken

[Dovid Bender]

Telnyx, 382com, voicetel and as others mentioned BandWidth. I have contacts
at 382 and voicetel if you want an intro.


On Thu, Aug 17, 2023 at 11:50 PM Federico 
wrote:

> I am looking for a decent provider of SIP Trunks but it has to pass the
> Stir Shaken token to the next carrier. Does anybody know about any?
> Sipstation from Sangoma, does not support Stir Shaken. ( Case #01466843 /
> 001300G8PLG / MAIN / Open [ ref:_00D306mPe._5004U1BlBLF:ref ])
>
> Although it’s mandatory, somehow they think it’s ok. Go figure.
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> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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Re: [asterisk-users] Question about Sip Trunks who support Stir Shaken

[Jeff LaCoursiere]

Bandwidth.com, although there are minimums to meet.

Cheers,

Jeff LaCoursiere
StratusTalk, Inc.

On Fri, Aug 18, 2023 at 7:52 AM TTT  wrote:

> Check out Twilio
>
>
>
>
>
> *From:* asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] *On
> Behalf Of *Federico
> *Sent:* Thursday, August 17, 2023 11:49 PM
> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' <
> asterisk-users@lists.digium.com>
> *Subject:* [asterisk-users] Question about Sip Trunks who support Stir
> Shaken
>
>
>
> I am looking for a decent provider of SIP Trunks but it has to pass the
> Stir Shaken token to the next carrier. Does anybody know about any?
> Sipstation from Sangoma, does not support Stir Shaken. ( Case #01466843 /
> 001300G8PLG / MAIN / Open [ ref:_00D306mPe._5004U1BlBLF:ref ])
>
> Although it’s mandatory, somehow they think it’s ok. Go figure.
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> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> 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] Question about Sip Trunks who support Stir Shaken

[TTT]

Check out Twilio

 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On
Behalf Of Federico
Sent: Thursday, August 17, 2023 11:49 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Subject: [asterisk-users] Question about Sip Trunks who support Stir Shaken

 

I am looking for a decent provider of SIP Trunks but it has to pass the Stir
Shaken token to the next carrier. Does anybody know about any? Sipstation
from Sangoma, does not support Stir Shaken. ( Case #01466843 /
001300G8PLG / MAIN / Open [ ref:_00D306mPe._5004U1BlBLF:ref ])

Although it's mandatory, somehow they think it's ok. Go figure.

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Re: [asterisk-users] Question on ring count on incoming circuits

[asterisk]

On 5/29/2023 4:12 PM, Steve Matzura wrote:

On 5/28/2023 2:27 PM, Naveen Albert wrote:
However, you can also pass audio without supervising (early media). 
You typically need to Progress() first to allow this, e.g. for SIP, 
or audio won't pass at all.


...



If you want it to ring once and do something else, you could simply do:

exten => s,1,Wait(6) ; 1 ring cycle is 6 seconds
    same => n,Answer(); answer, and do something else



Just as you said at the top of this reply, no audio of any kind gets 
passed, so all the Wait(6) did was provide six seconds of dead-air 
silence before the outgoing message played. Oh well. Customers can't 
have everything. ;-)
Well, yes, that's what you wanted, right? Or maybe I misunderstood. If 
you want people to hear *something* but not have it answer immediately, 
for those 6 seconds, amend that to:


exten => s,1,Progress()
   same => n,Playback(foobar,noanswer)
   same => n,Answer()
   same => n,DoSomething()

For example, this is common for playing an outgoing message or voicemail 
greeting, without supervising immediately, so if the caller hangs up 
before leaving a message, s/he is not charged for the call. Are you 
trying to do something like that?



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Re: [asterisk-users] Question on ring count on incoming circuits

[Steve Matzura]

On 5/28/2023 2:27 PM, Naveen Albert wrote:
However, you can also pass audio without supervising (early media). 
You typically need to Progress() first to allow this, e.g. for SIP, or 
audio won't pass at all.


...



If you want it to ring once and do something else, you could simply do:

exten => s,1,Wait(6) ; 1 ring cycle is 6 seconds
    same => n,Answer(); answer, and do something else



Just as you said at the top of this reply, no audio of any kind gets 
passed, so all the Wait(6) did was provide six seconds of dead-air 
silence before the outgoing message played. Oh well. Customers can't 
have everything. ;-)



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Re: [asterisk-users] Question on ring count on incoming circuits

[Doug Lytle]

On 5/28/23 14:20, Steve Matzura wrote:
Who controls how many times an incoming call from an external (DID) 
provider will ring before Asterisk picks up the call and handles it 
internally


Asterisk and this is defined with your timeout on the dial command, mine 
is 26 seconds so around 5 rings.




Doug

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Re: [asterisk-users] Question on ARI externalMedia

[Dan Cropp]

Please disregard, I figured out what I was doing wrong.

Dan


From: Dan Cropp
Sent: Friday, January 20, 2023 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Question on ARI externalMedia

A couple years ago, I know I had ARI externalMedia working.  Trying to figure 
out what I'm doing wrong today.


https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI

My ari.conf

[general]
enabled = yes
pretty = no
allowed_origins = *

[MyApp]
type = user
read_only = no
password_format = plain
password = Password

I send this curl -v -u MyApp:Password -X POST 
"http://localhost:8088/ari/channels/externalMedia?channelId=1234abcd5678&app=MyApp&external_host=192.168.33.32%3A1053&format=slin16";

I can make other ARI commands work, so it must be something specific to my 
externalMedia command and the parameters.

The output is the following...

{"id":"1234abcd5678","name":"UnicastRTP/192.168.33.32:1053-0x7fcffc020300","state":"Down","protocol_id":"","caller":{"name":"","number":""},"connected":{"name":"","number":""},"accountcode":"","dialplan":{"context":"default","exten":"s","priority":1,"app_name":"AppDial2","app_data":"(Outgoing
 
Line)"},"creationtime":"2023-01-20T10:59:24.569-0600","language":"en","channelvars":{"UNICASTRTP_LOCAL_PORT":"19194","UNICASTRTP_LOCAL_ADDRESS":"192.168.33.31"}}

Dan
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Re: [asterisk-users] Question on resources

[Jerry Geis]

On Mon, Sep 5, 2022 at 9:16 AM Mark Murawski 
wrote:

> On 8/4/22 20:32, Jerry Geis wrote:
> > I am running Asterisk 13.30.0
> > 40 core CPU (VM) VMware.
> > CentOS 7
> > 32 G ram
> > 10G vmx network
> >
> > Should be plenty of room for anything...
> >
> > Yes asterisk is running 270% CPU...
> > Is it not taking advantage of the 40 cores ?
> > I am bring around 300 SIP endpoints in a muted audio conference (so
> > one way) and this spikes up the CPU to 270%.
> >
> > Is there something I dont have set right to take advantage to
> > the resourses?
> > Thanks
> >
> > Jerry
> >
>
> Hi Jerry,
>
> If I recall correctly, there was a talk at an AstriCon or a web page
> somewhere that I came across at one point (I'm having a hard time
> finding it now) that dove in fairly deep into Asterisk performance
> related to multiple cores.
>
> And if I recall correctly, the conclusion was that the drop-off was
> around 8-12 cores -- and beyond that the extra cores aren't doing much
> other than helping schedule work and you can't really get more
> concurrent calls by adding more cores.
>
> Someone who is a bit more well-versed in large-machine performance with
> Asterisk can certainly chime in here, but from what I gather, throwing
> 40 cores at a single Asterisk instance is not the magic bullet to
> support a massive number of calls.
>
>
> Thanks Mark,

Jerry
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Re: [asterisk-users] Question on resources

[Mark Murawski]

On 8/4/22 20:32, Jerry Geis wrote:

I am running Asterisk 13.30.0
40 core CPU (VM) VMware.
CentOS 7
32 G ram
10G vmx network

Should be plenty of room for anything...

Yes asterisk is running 270% CPU...
Is it not taking advantage of the 40 cores ?
I am bring around 300 SIP endpoints in a muted audio conference (so 
one way) and this spikes up the CPU to 270%.


Is there something I dont have set right to take advantage to 
the resourses?

Thanks

Jerry



Hi Jerry,

If I recall correctly, there was a talk at an AstriCon or a web page 
somewhere that I came across at one point (I'm having a hard time 
finding it now) that dove in fairly deep into Asterisk performance 
related to multiple cores.


And if I recall correctly, the conclusion was that the drop-off was 
around 8-12 cores -- and beyond that the extra cores aren't doing much 
other than helping schedule work and you can't really get more 
concurrent calls by adding more cores.


Someone who is a bit more well-versed in large-machine performance with 
Asterisk can certainly chime in here, but from what I gather, throwing 
40 cores at a single Asterisk instance is not the magic bullet to 
support a massive number of calls.




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Re: [asterisk-users] Question on Originate with EarlyMedia

[Joshua C. Colp]

On Thu, Sep 1, 2022 at 1:32 PM Dan Cropp  wrote:

> Using AMI, we send an Originate with EarlyMedia: true setting
>
>
>
> If the other end sends a 183, Asterisk
>
> When the 183 is received, Asterisk indicates the ChannelState: 6 and
> ChannelStateDesc: Up values.
>
> All is fine up to this point.
>
>
>
> It may take the caller several seconds before the called party answers.
>
> When the called party answers (200 OK received), in the debugging I see
> Asterisk processing this and debugging show TSX State: Terminated  Inv
> State: EARLY
>
> At this point, the call is truly connected.
>
>
>
> Is there a configuration setting to indicate whether Asterisk should send
> an event indicating when the early media ends and the call is really Up?
>

There is no option. The Originate code makes the channel appear as answered
when the 183 arrives, everything reflects that afterwards. Even if a second
answer occurs it gets ignored. The log message you refer to is internal
state information to do with the SIP side.

-- 
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] Question on resources

[Karsten Wemheuer]

Hi,

Am Donnerstag, dem 04.08.2022 um 20:32 -0400 schrieb Jerry Geis:
> I am running Asterisk 13.30.0
> 40 core CPU (VM) VMware.
> CentOS 7
> 32 G ram
> 10G vmx network
> 
> Should be plenty of room for anything...
> 
> Yes asterisk is running 270% CPU...
> Is it not taking advantage of the 40 cores ? 
> I am bring around 300 SIP endpoints in a muted audio conference (so
> one way) and this spikes up the CPU to 270%.

What type of conference? Is it meetme or confbridge?

AFAIK meetme is working on a single thread...

> 
> Is there something I dont have set right to take advantage to
> the resourses?
> Thanks
> 
> Jerry

HTH,

Karsten



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Re: [asterisk-users] Question on resources

[dk]

Doesn’t that mean, effectively that you are using the equivalent of 100% of 2.7 
CPUs?

 

  --Don

 

 

From: asterisk-users  On Behalf Of 
Jerry Geis
Sent: Thursday, August 4, 2022 7:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: [asterisk-users] Question on resources

 

I am running Asterisk 13.30.0

40 core CPU (VM) VMware.

CentOS 7

32 G ram

10G vmx network

 

Should be plenty of room for anything...

 

Yes asterisk is running 270% CPU...

Is it not taking advantage of the 40 cores ? 

I am bring around 300 SIP endpoints in a muted audio conference (so one way) 
and this spikes up the CPU to 270%.

 

Is there something I dont have set right to take advantage to the resourses?

Thanks

 

Jerry

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

[George Joseph]

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

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

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

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

A real scenario to test!!!  Thanks!

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

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

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

Actually it would have a template specifying replacement channel variables.

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

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

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

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

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

Sounds more complicated than it needs to be.

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

[George Joseph]

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

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

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



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

[Dan Cropp]

Thanks Joshua

From: asterisk-users  On Behalf Of 
Joshua C. Colp
Sent: Friday, February 14, 2020 1:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Question on pjsip.conf and aors

On Fri, Feb 14, 2020 at 3:04 PM Dan Cropp 
mailto:d...@amtelco.com>> wrote:
I have the following configuration…

[aor3]
type = aor
max_contacts = 1
remove_existing = yes

[auth3]
type = auth
username = 1004
password = SuperSecretProbation

[1004]
type = endpoint
context = IS
transport = transport1
auth = auth3
aors = aor3
accountcode = 3
dtmf_mode = rfc4733
device_state_busy_at = 2
force_rport = no
moh_passthrough = yes
disallow = all
allow = ulaw
acl = acl1


When a register attempt is received, asterisk outputs…
[02/14 12:53:29.870] WARNING[7883] res_pjsip_registrar.c: AOR '1004' not found 
for endpoint '1004'

If I change the aor3 to be 1004, everything works.  As in [aor3] becomes [1004] 
and in the endpoint change aors = aor3 to be aors = 1004
Is there a setting I’m missing to allow the endpoint named 1004 to use an auth 
that doesn’t have the same 1004 name?

There isn't a configuration option. AOR is a SIP concept, and in fact when you 
send a REGISTER you state which AOR you are registering to. Your REGISTER is 
therefore saying "add me to AOR 1004". Since it's not saying "add me to aor3" 
it doesn't work. Some devices allow you to specify while others just assume 
that everything uses your username.

--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com<http://www.sangoma.com> and 
www.asterisk.org<http://www.asterisk.org>
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Re: [asterisk-users] Question on pjsip.conf and aors

[Joshua C. Colp]

On Fri, Feb 14, 2020 at 3:04 PM Dan Cropp  wrote:

> I have the following configuration…
>
>
>
> [aor3]
>
> type = aor
>
> max_contacts = 1
>
> remove_existing = yes
>
>
>
> [auth3]
>
> type = auth
>
> username = 1004
>
> password = SuperSecretProbation
>
>
>
> [1004]
>
> type = endpoint
>
> context = IS
>
> transport = transport1
>
> auth = auth3
>
> aors = aor3
>
> accountcode = 3
>
> dtmf_mode = rfc4733
>
> device_state_busy_at = 2
>
> force_rport = no
>
> moh_passthrough = yes
>
> disallow = all
>
> allow = ulaw
>
> acl = acl1
>
>
>
>
>
> When a register attempt is received, asterisk outputs…
>
> [02/14 12:53:29.870] WARNING[7883] res_pjsip_registrar.c: AOR '1004' not
> found for endpoint '1004'
>
>
>
> If I change the aor3 to be 1004, everything works.  As in [aor3] becomes
> [1004] and in the endpoint change aors = aor3 to be aors = 1004
> Is there a setting I’m missing to allow the endpoint named 1004 to use an
> auth that doesn’t have the same 1004 name?
>

There isn't a configuration option. AOR is a SIP concept, and in fact when
you send a REGISTER you state which AOR you are registering to. Your
REGISTER is therefore saying "add me to AOR 1004". Since it's not saying
"add me to aor3" it doesn't work. Some devices allow you to specify while
others just assume that everything uses your username.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] Question on WebRTC configuration

[Olivier]

Hello,

Reading this old thread, isn't there also an error in [1] as It also
mentions a tlscafile setting.

Cheers

[1]
https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone

Le ven. 7 déc. 2018 à 16:41, Kevin Harwell  a écrit :

> On Fri, Dec 7, 2018 at 9:11 AM Dan Cropp  wrote:
>
>> In the asterisk wiki instructions for Configuring Asterisk for WebRTC
>> clients…
>>
>>
>>
>>
>> https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients
>>
>>
>>
>> “To communicate with websocket clients, Asterisk uses its built-in HTTP
>> daemon.  Configure */etc/asterisk/http.conf* as follows:
>>
>>
>>
>> [general]
>>
>> enabled=yes
>>
>> bindaddr=0.0.0.0
>>
>> bindport=8088
>>
>> tlsenable=yes
>>
>> tlsbindaddr=0.0.0.0:8089
>>
>> tlscertfile=
>>
>> tlsprivatekey=
>>
>> tlscafile=”
>>
>>
>>
>> What is the tlscafile setting?
>>
>>
>>
>> When I look at the http.conf samples it doesn’t mention the tlscafile
>> setting.
>>
>> I see there is a tlscafile setting in sip.conf, but I don’t find this
>> anywhere else.
>>
>>
>>
>> Is the wiki web page mistaken or is this an actual http.conf setting that
>> is undocumented?
>>
>
> The page is mistaken. It should not be there. the 'tlscafile' option is
> not supported by the Asterisk http server. I've removed it from the wiki.
> Thanks for catching that!
>
>
>>
>>
>> Have a great day!
>>
>
> You too!
>
>
>> Dan
>> --
>>
>
> --
> Kevin Harwell
> Digium - A Sangoma Company | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: https://digium.com & https://asterisk.org
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
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Re: [asterisk-users] Question on calculating PJSIP md5 authentication with NEC

[Dan Cropp]

Thank you Joshua.

We are confident the problem is with NEC.

One day, Asterisk PJSIP REGISTER response (md5) was being rejected by NEC.
Next morning, it's suddenly working with no changes to asterisk.  Same exact 
configuration settings.
Suddenly, last Friday NEC starts rejecting the REGISTER again with no changes 
to asterisk configuration file.

NEC has since responded that they have a separate PJSIP setting for 
REGISTRATION.  We are trying to find out more information from them.
NEC claim doesn't explain why it worked for several hours and suddenly stopped 
working.  This really feels like they have been modifying settings on their end 
without informing us.

Dan

-Original Message-
From: asterisk-users  On Behalf Of 
Joshua C. Colp
Sent: Monday, July 15, 2019 1:31 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Question on calculating PJSIP md5 authentication 
with NEC

On Fri, Jul 12, 2019, at 5:10 PM, Dan Cropp wrote:
>  
> Just tracked down the code for the chan_sip MD5 REGISTER and have been 
> able to verify that chan_sip is calculating the HA1 same as I am 
> calculating the md5_cred for PJSIP
> 
> 3016:a...@xyz.com:3016
> 
> 
> Both chan_sip and PJSIP REGISTER traces show the uri as 
> sip:10.100.102.82
> 
> So the HA2 for both as MD5(method:uri) (which in this case is 
> REGISTER:sip:10.100.102.82).
> 
> 
> chan_sip response formula with qop detected (sent by NEC) is
> 
> ha1:nonce:noncecount:cnonce:auth:h2
> 
> Using this formula I am able to match it with my Asterisk chan_sip trace. 
> 
> NEC accepts this REGISTER
> 
> 
> Can anyone point me to the area where Asterisk PJSIP would be doing 
> the response for the REGISTER 401 reply?
> 
> NEC does not like the way PJSIP calculates the response value for the 
> REGISTER reply.
> 
> I tried to calculate the response exactly like chan_sip does (using 
> the values from the trace and the HA1/HA2) but it’s not matching what 
> the sip trace shows Asterisk sending for the response.
> 
> 
> Does Asterisk PJSIP support handle this or is it all done by PJSIP? 

The value is given to PJSIP[1] and it does the rest. You could use the password 
option and trace how PJSIP is calculating it in that scenario, and then switch 
to MD5 cred after you determine that and make it match.

[1] 
https://github.com/asterisk/asterisk/blob/master/res/res_pjsip_authenticator_digest.c#L180

--
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] Question on calculating PJSIP md5 authentication with NEC

[Joshua C. Colp]

On Fri, Jul 12, 2019, at 5:10 PM, Dan Cropp wrote:
>  
> Just tracked down the code for the chan_sip MD5 REGISTER and have been 
> able to verify that chan_sip is calculating the HA1 same as I am 
> calculating the md5_cred for PJSIP
> 
> 3016:a...@xyz.com:3016
> 
> 
> Both chan_sip and PJSIP REGISTER traces show the uri as sip:10.100.102.82
> 
> So the HA2 for both as MD5(method:uri) (which in this case is 
> REGISTER:sip:10.100.102.82).
> 
> 
> chan_sip response formula with qop detected (sent by NEC) is
> 
> ha1:nonce:noncecount:cnonce:auth:h2
> 
> Using this formula I am able to match it with my Asterisk chan_sip trace. 
> 
> NEC accepts this REGISTER
> 
> 
> Can anyone point me to the area where Asterisk PJSIP would be doing the 
> response for the REGISTER 401 reply?
> 
> NEC does not like the way PJSIP calculates the response value for the 
> REGISTER reply.
> 
> I tried to calculate the response exactly like chan_sip does (using the 
> values from the trace and the HA1/HA2) but it’s not matching what the 
> sip trace shows Asterisk sending for the response.
> 
> 
> Does Asterisk PJSIP support handle this or is it all done by PJSIP? 

The value is given to PJSIP[1] and it does the rest. You could use the password 
option and trace how PJSIP is calculating it in that scenario, and then switch 
to MD5 cred after you determine that and make it match.

[1] 
https://github.com/asterisk/asterisk/blob/master/res/res_pjsip_authenticator_digest.c#L180

-- 
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] Question on calculating PJSIP md5 authentication with NEC

[Dan Cropp]

Just tracked down the code for the chan_sip MD5 REGISTER and have been able to 
verify that chan_sip is calculating the HA1 same as I am calculating the 
md5_cred for PJSIP
3016:a...@xyz.com:3016

Both chan_sip and PJSIP REGISTER traces show the uri as sip:10.100.102.82
So the HA2 for both as MD5(method:uri)  (which in this case is 
REGISTER:sip:10.100.102.82).

chan_sip response formula with qop detected (sent by NEC) is
ha1:nonce:noncecount:cnonce:auth:h2
Using this formula I am able to match it with my Asterisk chan_sip trace.
NEC accepts this REGISTER

Can anyone point me to the area where Asterisk PJSIP would be doing the 
response for the REGISTER 401 reply?
NEC does not like the way PJSIP calculates the response value for the REGISTER 
reply.
I tried to calculate the response exactly like chan_sip does (using the values 
from the trace and the HA1/HA2) but it's not matching what the sip trace shows 
Asterisk sending for the response.

Does Asterisk PJSIP support handle this or is it all done by PJSIP?

Dan


From: asterisk-users  On Behalf Of Dan 
Cropp
Sent: Friday, July 12, 2019 2:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Question on calculating PJSIP md5 authentication 
with NEC

I have done additional testing and I haven't been able to figure out why it's 
failing.

Since my original testing we now set the realm on the authentication section to 
match what we receive from NEC.  It's of the format 
a...@xyz.com<mailto:a...@xyz.com>
I have verified the md5_cred several times and it matches the 
user:realm:password formula 3016:ins...@something0a64.com:3016 where 
username is 3016 and password is 3016

We suspect it has something to do with the format of the realm that NEC is 
sending where it may not be working correctly supported by the Asterisk PJSIP 
code.

Is there anyone who has used PJSIP outbound REGISTRATION using MD5 support that 
can provide some insight?
Or even anyone who know chan_sip's REGISTER and how it calculates it's HA1, HA2 
for the MD5 authentication?

Dan

From: asterisk-users 
mailto:asterisk-users-boun...@lists.digium.com>>
 On Behalf Of Dan Cropp
Sent: Wednesday, July 10, 2019 10:48 AM
To: asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>
Subject: [asterisk-users] Question on calculating the md5_sum

Using chan_sip, we are able to register with an NEC switch.  When I try to 
REGISTER with PJSIP, the authentication is being rejected.  Traces show it's 
using md5 authentication.
The packets looks almost identical.  The one area that I suspect is causing the 
problem is the md5_cred for my pjsip.conf registration.

I'm using a Poco MD5 utility to generate the MD5 passing username:realm:password
Where username is 3016
Realm is asterisk (default)
Password is 3016 which is the same as chan_sip's secret
The value I'm setting the md5_cred in auth section to is 
63e8aedc77335879c93123055d21211d

Would this value match what chan_sip would pass as the md5 credentials?


Our sip.conf looks like the following...
[general]
context = NECTEST
bindaddr = 0.0.0.0
bindport = 5060
websocket_enabled = false
srvlookup = no
allowguest = yes
debug = yes
sipdebug = yes
defaultexpiry = 480
deny = 0.0.0.0/24
permit = 10.100.102.0/24
permit = 192.168.9.0/24
canreinvite = yes
callcounter = yes
register = 3016:3016@10.100.102.82:5060/3016

[3016]
type = friend
qualify = no
nat = no
host = 10.100.102.82:5060
defaultuser = 3016
secret = 3016
incominglimit = 24
accountcode = 33
port = 5060
context = NECTEST
dtmfmode = auto
disallow = all
allow = ulaw
defaultexpiry = 480
insecure = invite
fromdomain = 10.100.102.82
acl = acl6

Have a great day!

Dan
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Re: [asterisk-users] Question on calculating PJSIP md5 authentication with NEC

[Dan Cropp]

I have done additional testing and I haven't been able to figure out why it's 
failing.

Since my original testing we now set the realm on the authentication section to 
match what we receive from NEC.  It's of the format a...@xyz.com
I have verified the md5_cred several times and it matches the 
user:realm:password formula 3016:ins...@something0a64.com:3016 where 
username is 3016 and password is 3016

We suspect it has something to do with the format of the realm that NEC is 
sending where it may not be working correctly supported by the Asterisk PJSIP 
code.

Is there anyone who has used PJSIP outbound REGISTRATION using MD5 support that 
can provide some insight?
Or even anyone who know chan_sip's REGISTER and how it calculates it's HA1, HA2 
for the MD5 authentication?

Dan

From: asterisk-users  On Behalf Of Dan 
Cropp
Sent: Wednesday, July 10, 2019 10:48 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Question on calculating the md5_sum

Using chan_sip, we are able to register with an NEC switch.  When I try to 
REGISTER with PJSIP, the authentication is being rejected.  Traces show it's 
using md5 authentication.
The packets looks almost identical.  The one area that I suspect is causing the 
problem is the md5_cred for my pjsip.conf registration.

I'm using a Poco MD5 utility to generate the MD5 passing username:realm:password
Where username is 3016
Realm is asterisk (default)
Password is 3016 which is the same as chan_sip's secret
The value I'm setting the md5_cred in auth section to is 
63e8aedc77335879c93123055d21211d

Would this value match what chan_sip would pass as the md5 credentials?


Our sip.conf looks like the following...
[general]
context = NECTEST
bindaddr = 0.0.0.0
bindport = 5060
websocket_enabled = false
srvlookup = no
allowguest = yes
debug = yes
sipdebug = yes
defaultexpiry = 480
deny = 0.0.0.0/24
permit = 10.100.102.0/24
permit = 192.168.9.0/24
canreinvite = yes
callcounter = yes
register = 3016:3016@10.100.102.82:5060/3016

[3016]
type = friend
qualify = no
nat = no
host = 10.100.102.82:5060
defaultuser = 3016
secret = 3016
incominglimit = 24
accountcode = 33
port = 5060
context = NECTEST
dtmfmode = auto
disallow = all
allow = ulaw
defaultexpiry = 480
insecure = invite
fromdomain = 10.100.102.82
acl = acl6

Have a great day!

Dan
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Re: [asterisk-users] Question about packet counts in voipmonitor

[Dovid Bender]

Mike,

Are you using port mirroring or is  VoipMonitor running on the same box? If
the latter I would run tcpdump and compare what VM says it has to what you
see in your wireshark dump. If you are sniffing via port mirroring your
switch maybe dropping packets (we had that when we tried to mirror 700 mbit
of traffic on a Juniper EX4200).


On Fri, Dec 21, 2018 at 8:13 PM Mike Diehl  wrote:

> Hi all,
>
>
>
> I'm not sure this is the place to ask, but here goes...
>
>
>
> I'm using voipmonitor to gather call statistics such as packet counts,
> average jitter, etc. Eventually, I want to use those stats to detect and
> alert on poor call quality.
>
>
>
> However, I'm finding that the packet counts for each leg of a given call
> can vary quite a bit.
>
>
>
> For example, I have a call that was connected for 84 seconds. At 50
> frames/sec, I expect to see about 4200 frames. However, on one side I see
> 4187 (which is good) and on the other side, I only see 2577 frames sent.
>
>
>
> Am I doing something wrong? Or is this approach simply doomed?
>
>
>
> Any thoughts would be welcome.
>
>
>
> --
>
> Mike Diehl
>
>
>
>
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> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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Re: [asterisk-users] Question on WebRTC configuration

[Kevin Harwell]

On Fri, Dec 7, 2018 at 9:11 AM Dan Cropp  wrote:

> In the asterisk wiki instructions for Configuring Asterisk for WebRTC
> clients…
>
>
>
>
> https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients
>
>
>
> “To communicate with websocket clients, Asterisk uses its built-in HTTP
> daemon.  Configure */etc/asterisk/http.conf* as follows:
>
>
>
> [general]
>
> enabled=yes
>
> bindaddr=0.0.0.0
>
> bindport=8088
>
> tlsenable=yes
>
> tlsbindaddr=0.0.0.0:8089
>
> tlscertfile=
>
> tlsprivatekey=
>
> tlscafile=”
>
>
>
> What is the tlscafile setting?
>
>
>
> When I look at the http.conf samples it doesn’t mention the tlscafile
> setting.
>
> I see there is a tlscafile setting in sip.conf, but I don’t find this
> anywhere else.
>
>
>
> Is the wiki web page mistaken or is this an actual http.conf setting that
> is undocumented?
>

The page is mistaken. It should not be there. the 'tlscafile' option is not
supported by the Asterisk http server. I've removed it from the wiki.
Thanks for catching that!


>
>
> Have a great day!
>

You too!


> Dan
> --
>

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Re: [asterisk-users] Question on PJSIP's endpoint section in wiki

[Joshua Colp]

On Fri, Apr 27, 2018, at 11:13 AM, Olivier wrote:
> Hello,
> 
> I don't know if this list is the best place to ask such question but here
> it is, anyway.
> 
> In page [1], I can read in PJSIP's endpoint section configuration reference:
> identify_by   username,location  Way(s) for Endpoint to be
> identified
> 
> Then clicking over identify_by text, you can read:
> identify_by   ... supported options are username, ... and auth_username
> 
> How do yopu read it ?
> I would expect the first line to written as:
> dentify_by   username,auth_username  Way(s) for Endpoint to be
> identified

The wiki documentation hasn't been regenerated lately (it's in queue to be 
fixed). "username,auth_username" would be correct. There's also others[1] 
depending on version.

[1] 
https://github.com/asterisk/asterisk/blob/13/configs/samples/pjsip.conf.sample#L633

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Re: [asterisk-users] Question on permit/deny

[Ishfaq Malik]

On 1 July 2015 at 04:03, Jerry Geis  wrote:

> I see in my log file this:
> Jun 30 21:44:26] NOTICE[42192][C-02f3] chan_sip.c: Call from '' (
> 5.189.144.120:5076) to extension '011972592675431' rejected because
> extension not found in context 'default'.
>
> which is great its rejected - however
> in my sip.conf file I have
>
> deny=0.0.0.0
> permit=x.y.z.z/255.255.255.255
> permit=a.b.c.d/255.255.255.255
>
> So I'm expecting to deny everything and only allow
> the two addresses I have listed of which the 5.189.144.120 is not one of?
>
> What is wrong with my permit/deny ?
>
> Thanks,
>
> Jerry
>
> --
> _
>
>
Check your sip.conf to see if allowguest is explicitly set to no.

;context=default ; Default context for incoming calls
;allowguest=no  ; Allow or reject guest calls (default is
yes)
; If your Asterisk is connected to the
Internet
; and you have allowguest=yes
; you want to check which services you
offer everyone
; out there, by enabling them in the
default context (see below).


Regards

Ish


-- 

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Company: Packnet Limited
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f: +44 (0)161 660 9825
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Re: [asterisk-users] Question about Warning message

[Fabian Borot]

thank you, we are using the same configuration files in 13, same setup, just 
different asterisk version. we just dont see the msgs in the console/logs, it 
is the same exact voice traffic on both asterisk versions

is that something that you set on/off? if that is the case how can it be done?

what is the alternative? what are their differences/characteristics? how to 
choose one over among others?

thank you again


> From: fbo...@hotmail.com
> To: asterisk-users@lists.digium.com
> Subject: Question about Warning message
> Date: Mon, 23 Feb 2015 12:27:05 -0500
>
>
> Starting with Asterisk 13.1 we are seeing this WARNING messages a lot in our 
> logs and console:
>
>
> WARNING[25164][C-0004865e]: chan_sip.c:7364 sip_write: Can't send 10 type 
> frames with SIP write)
>
>
> We found that line in function "sip_write" inside "chan_sip.c".
>
> In our previous version (11.2.1) we did not see those messages being printed 
> (same verbosity level). We compared both versions of the functions and see no 
> difference at all in the 'default' switch case that handles that. We 
> think/assume that that function is being called in
> different places on each version (11.2-1 vs 13-1).
>
> We also think it has to do with the asterisk receiving rtp packets with 
> comfort noise which is not supported by asterisk.
>
> We would like to know what can we do about it to behave more like the version 
> 11?
>
> We are not sure but could it be that version 11 handles it better ?. I am 
> attaching the functions on both versions for your review.
>
> Thank you
>
>
>
>
  
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Re: [asterisk-users] Question regarding custom announcements used by several Asterisk servers

[Paul Belanger]

On Fri, Feb 6, 2015 at 5:03 AM, Ishfaq Malik  wrote:
>
>
> On 6 February 2015 at 07:54, Olli Heiskanen 
> wrote:
>>
>>
>> Hello,
>>
>> Got a question regarding custom announcements in Asterisk.
>>
>> My goal is to allow my users record their own queue announcements and
>> choose which announcements they want to use in each queue. I have several
>> Asterisk servers and a Kamailio server which dispatches call traffic between
>> the Asterisks. Question is, is it possible to have something like a NSF disk
>> shared between several asterisk servers and store custom announcements
>> there, where all Asterisks would use them? I expect to have to place the
>> files under whatever I configure in asterisk.conf. Additionally, can I place
>> the announcements in subfolders under that directory and in my realtime
>> queue table use values something like '/subfldr/myannouncement'?
>>
>> Keep up the good work!
>>
>> cheers,
>> Olli
>>
>> --
>> _
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>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> Hi
>
> All of that is possible and is exactly what we do, both for customer sounds
> and for call recordings. Just make sure you have resilience in your shared
> storage device.
>
> Alternatively, you could use something like Puppet to deploy the files to
> all the servers.
>
This is basically what we do, we use puppet to help distribute files
to remote servers while still using app_queue.  Shared network drive
also works.

-- 
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Re: [asterisk-users] Question regarding custom announcements used by several Asterisk servers

[Olli Heiskanen]

Oops, quite right, how typoful of me!

Thanks for the excellent points, I'll look into gluster and puppet and see
may way onwards from there.

cheers,
Olli

2015-02-06 12:32 GMT+02:00 Gareth Blades :

> On 06/02/15 07:54, Olli Heiskanen wrote:
>
>> My goal is to allow my users record their own queue announcements and
>> choose which announcements they want to use in each queue. I have several
>> Asterisk servers and a Kamailio server which dispatches call traffic
>> between the Asterisks. Question is, is it possible to have something like a
>> NSF disk shared between several asterisk servers and store custom
>> announcements there, where all Asterisks would use them? I expect to have
>> to place the files under whatever I configure in asterisk.conf.
>> Additionally, can I place the announcements in subfolders under that
>> directory and in my realtime queue table use values something like
>> '/subfldr/myannouncement'?
>>
>
> I assume you mean NFS.
> Yes you can do that although using NFS you will then have a single point
> of failure and in the standard NFS client configuration if you try to
> access a file which is on NFS but it is unavailable then the file access
> will hang.
>
> So you might be better off having the files copied onto each of the
> asterisks servers local file storage or use a redundant file system such as
> gluster.
>
>
>
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Re: [asterisk-users] Question regarding custom announcements used by several Asterisk servers

[Gareth Blades]

On 06/02/15 07:54, Olli Heiskanen wrote:
My goal is to allow my users record their own queue announcements and 
choose which announcements they want to use in each queue. I have 
several Asterisk servers and a Kamailio server which dispatches call 
traffic between the Asterisks. Question is, is it possible to have 
something like a NSF disk shared between several asterisk servers and 
store custom announcements there, where all Asterisks would use them? 
I expect to have to place the files under whatever I configure in 
asterisk.conf. Additionally, can I place the announcements in 
subfolders under that directory and in my realtime queue table use 
values something like '/subfldr/myannouncement'?


I assume you mean NFS.
Yes you can do that although using NFS you will then have a single point 
of failure and in the standard NFS client configuration if you try to 
access a file which is on NFS but it is unavailable then the file access 
will hang.


So you might be better off having the files copied onto each of the 
asterisks servers local file storage or use a redundant file system such 
as gluster.



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Re: [asterisk-users] Question regarding custom announcements used by several Asterisk servers

[Ishfaq Malik]

On 6 February 2015 at 07:54, Olli Heiskanen 
wrote:

>
> Hello,
>
> Got a question regarding custom announcements in Asterisk.
>
> My goal is to allow my users record their own queue announcements and
> choose which announcements they want to use in each queue. I have several
> Asterisk servers and a Kamailio server which dispatches call traffic
> between the Asterisks. Question is, is it possible to have something like a
> NSF disk shared between several asterisk servers and store custom
> announcements there, where all Asterisks would use them? I expect to have
> to place the files under whatever I configure in asterisk.conf.
> Additionally, can I place the announcements in subfolders under that
> directory and in my realtime queue table use values something like
> '/subfldr/myannouncement'?
>
> Keep up the good work!
>
> cheers,
> Olli
>
> --
> _
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>

Hi

All of that is possible and is exactly what we do, both for customer sounds
and for call recordings. Just make sure you have resilience in your shared
storage device.

Alternatively, you could use something like Puppet to deploy the files to
all the servers.

-- 

Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
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Manchester, M1 2JW
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Re: [asterisk-users] Question about SIP warning

[dotnetdub]

Hi,

upto asterisk 1.8 you used to get this error if there were more than 1
m= line in an invite... Asterisk was just telling you it was declining
the second. I belive from 10.0 onwards asterisk now just replies back
with port 0 to the stream it isn't interested in...

You can ignore it - if its bothering you upgrade to asterisk 11 which
is very solid now.

On 6 September 2014 10:28, CDR  wrote:
> I get tons of these messages
> chan_sip.c:10088 process_sdp: Declining non-primary audio stream:
> audio 30660 RTP/AVP 4 101 13
> What does it mean and does it show a problem like one-way audio?
> Thanks for your help.
>
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Re: [asterisk-users] Question about SIP Dial

[Gopalakrishnan N]

It supposed to be like this Dial(SIP/${EXTEN}#ip.add.re.ss)

Regards


On Fri, Aug 15, 2014 at 6:20 AM, CDR  wrote:

> In channel PJSIP I use this format
> Dial(PJSIP/endpoint/sip:${EXTEN}@ip.add.re.ss)
> what would be the equivalent of this format in old SIP?
> I tried
> Dial(SIP/peer/${EXTEN}@ip.add.re.ss)
> but it does not work. I just cannot embed the IP address in the peer's
> definition, but I need to use some other configuration features that
> are unique to each peer.
>
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Re: [asterisk-users] Question about PJSIP

[Matthew Jordan]

On Mon, Jul 21, 2014 at 7:00 PM, CDR  wrote:
> I found that PJSIP allows only one asterisk per box. I tried to start
> several asterisks with the parameter -C and PJSIP only worked on the
> first process. In the other processes, the command "pjsip reload" was
> absent. Each pjsip transport in the second and subsequent processes
> was bound to a different IP in a multihomed box, something I routinely
> do with regular SIP.
> Am I wrong?

We routinely run multiple Asterisk instances on a single machine using
the PJSIP stack.

A log showing messages why the res_pjsip_* modules couldn't be loaded
on a particular instance of Asterisk would be helpful.

-- 
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Re: [asterisk-users] Question about Asterisk 12

[John Kiniston]

>
> I looked on http://www.voip-info.org - maybe I missed it?
> The Digium/Asterisk site - I see all sorts of cool things about the REST
> API, but CLI - maybe I missed it!!??  - again, I could be looking in the
> wrong place?
>
>
https://wiki.asterisk.org/wiki/display/AST/Home

To my knowledge the voip-info domain is mostly outdated information these
days. The official asterisk wiki is where you want to look for current
information.

Sorry I can't help you with anything else, I've not had time to play with
12 Yet.

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build a wall, set a bone, comfort the dying, take orders, give orders,
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Re: [asterisk-users] Question about Asterisk 12

[Jacob.E.Miles]

Maybe it's just me if I'm not mistaken the three things you listed are
usually configured using the config files not on CLI.

 

Jacob 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James
Wystead
Sent: Wednesday, January 22, 2014 3:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question about Asterisk 12

 

Okay - maybe I'm just suffering from a moment of horrible ADD - but, I'm
a little lost. 

I see that Asterisk 12 has a nice REST API - very nice - something I can
use. However, and this is gonna sound dumb - but all the CLI commands
are different now. What did I miss?

 

Can anyone, please, anyone point me to a good, simple to understand
tutorial on the new CLI? I am so, so freaking lost! I'm not looking for
hand-holding, I just want to understand.

 

Something that will show me how to:

 

*   create users
*   configure SIP trunks
*   configure basic dialplan

I'm lost - anyone point me to a resource that is easy to follow? Once I
get the jist, I think I'll be fine.

 

I looked on http://www.voip-info.org   -
maybe I missed it?

The Digium/Asterisk site - I see all sorts of cool things about the REST
API, but CLI - maybe I missed it!!??  - again, I could be looking in the
wrong place?

 

 

 

Overwhelming - sigh.

 

Thank much - any help would be appreciated - next time you are in
Manchester NH - I'll make you my fave Tequila Sour drink!

 

G

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Re: [asterisk-users] Question about Management Interface

[Joshua Colp]

CDR wrote:

I am trying to identify the module (*.so) that contains the Asterisk
Management Interface, so as to set  noload=XXX.so in modules.conf. Any
idea?


There is no module, it's provided as core functionality. Disabling it 
can be done in manager.conf


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Re: [asterisk-users] Question about how Asterisk works with RTP ports

[Joshua Colp]

Jonas Kellens wrote:



So if I understand correct, you don't need to look at the amount of
concurrent calls to calculate the RTP range in rtp.conf, you need to
look at the amount of INVITES that are being send at one moment ?


The number of concurrent channels in existence which are using RTP. 
While a channel may not be answered, it's still in existence.


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Re: [asterisk-users] Question about how Asterisk works with RTP ports

[Jonas Kellens]

On 10/29/2013 05:14 PM, Joshua Colp wrote:

Jonas Kellens wrote:

Hello,

short question : does Asterisk reserve RTP ports for every IP-phone that
is being called ?


It uses 2 ports per channel under normal circumstances, 1 for RTP and 
1 for RTCP.



If for instance an incoming call makes 10 IP-phones ring, does this mean
that Asterisk preserves 10 x 2 RTP ports for audio ?


Yes.


I guess Asterisk sends in the SIP INVITE an SDP body with an RTP port
number for audio ? If this is the case for the 10 IP-phones to which an
INVITE is send to, this means at least 10 RTP ports are reserved for
incoming audio, correct ???


Yes.




So if I understand correct, you don't need to look at the amount of 
concurrent calls to calculate the RTP range in rtp.conf, you need to 
look at the amount of INVITES that are being send at one moment ?




Kind regards,

Jonas.

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Re: [asterisk-users] Question about how Asterisk works with RTP ports

[Joshua Colp]

Jonas Kellens wrote:

Hello,

short question : does Asterisk reserve RTP ports for every IP-phone that
is being called ?


It uses 2 ports per channel under normal circumstances, 1 for RTP and 1 
for RTCP.



If for instance an incoming call makes 10 IP-phones ring, does this mean
that Asterisk preserves 10 x 2 RTP ports for audio ?


Yes.


I guess Asterisk sends in the SIP INVITE an SDP body with an RTP port
number for audio ? If this is the case for the 10 IP-phones to which an
INVITE is send to, this means at least 10 RTP ports are reserved for
incoming audio, correct ???


Yes.

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Re: [asterisk-users] Question on AEL2 string comparisons

[James B. Byrne]

On Thu, July 4, 2013 02:14, Satish Barot wrote:

>>
>
> ${CALLERID(num)} should give you only number and not technology i.e.
> 41712.
>
> Give this a shot,
>
> exten => _417XX,n,Noop(CALLERIDNUM=${CALLERID(num)})
> exten => _417XX,n,GotoIf($[$["${CALLERID(num)}" > "41799"] |
> $["${CALLERID(num)}" < "41700"]]?notfromlocal:)
>
> --Satish Barot
> Ahmedabad, India
>

That works.  Thank you.
-- 
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Re: [asterisk-users] Question on AEL2 string comparisons

[Satish Barot]

On Thu, Jul 4, 2013 at 12:24 AM, James B. Byrne wrote:

> I have this code in a dial plan:
>
> exten => _417XX,n,GotoIf($["${CALLERID(num)}" >
> "SIP/41799"]?notfromlocal)
> exten => _417XX,n,GotoIf($["${CALLERID(num)}" <
> "SIP/41700"]?notfromlocal)
>
> The value of "${CALLERID(num)}" appears to be "SIP/41712-0181"
>
> -- Executing [41720@from-internal:5] GotoIf("SIP/41712-0181",
> "0?notfromlocal") in new stack
> -- Executing [41720@from-internal:6] GotoIf("SIP/41712-0181",
> "1?notfromlocal") in new stack
> -- Goto (from-internal,41720,8
>
> This value is evidently comparing to be less than "SIP/41799" as
> expected but also is considered less than "SIP/41700" as well, which
> is not expected (by me).  What am I doing wrong here?
>
> What I am attempting to accomplish is to detect calls originally made
> from internal extension numbers in the range 41700..41799 inclusive.
> What is the correct method to accomplish this?
>
> James B. Byrne
>

${CALLERID(num)} should give you only number and not technology i.e. 41712.

Give this a shot,

exten => _417XX,n,Noop(CALLERIDNUM=${CALLERID(num)})
exten => _417XX,n,GotoIf($[$["${CALLERID(num)}" > "41799"] |
$["${CALLERID(num)}" < "41700"]]?notfromlocal:)

--Satish Barot
Ahmedabad, India
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Re: [asterisk-users] Question

[Joshua Colp]

CDR wrote:

Is it me or Google just blocked Asterisk's chan_motif? I get "violation
of terms of service" audio message whenever I send a call.


Works fine here. Their automated security system probably determined 
your usage behavior was not consistent with normal usage and terminated 
your access.


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Re: [asterisk-users] question about CDR

[Salaheddine Elharit]

thanks asghar for your help and support  and thanks ishfaq


2013/5/9 Asghar Mohammad 

> hi,
> asterisk insert cdr when call is hangup and last dial statment,
> i dont understatnd why you are using 2 dial statment on same extenstion?
> if you you want dial to both extensions you can use
> 506,1,Dial(SIP/223&SIP/276) if you want dial both same time or if you want
> to do failover the check Dial status and gotoif dialstatus = NO ANSWER or
> what ever you need.
>
>
>
> On Thu, May 9, 2013 at 10:46 AM, Salaheddine Elharit <
> salah.elharit...@gmail.com> wrote:
>
>> hello list,
>>
>> i need your help about cdr ,i have installed the module cdr in my
>> asterisk 1.4 .
>>
>> for the inbound calls when i call my sip exten like below :
>>
>> exten => 506,1,Dial(SIP/223, 10)
>> exten => 506,n,Dial(SIP/276, 10)
>>
>> in CDR i have just one line with SIP /276 the last line but there is no 
>> historic
>> for the first SIP 223
>>
>> recid Record ID | calldate   |clid   |src   |
>> dst |dcontext |channel | dstchannel   |lastapp |lastdata |duration
>> |billsec |disposition |amaflags |accountcode |uniqueid
>> |3 |
>>
>> 626747 |2013-05-09 09:22:55|0661551203  |0661551203|506
>>  |default  |Zap/14-1 |SIP/276-092ac7b0 |Dial|SIP/276| 10|21  |0
>>  |NO ANSWER
>>
>>
>> any help please to have the historic for 223 and 276
>>
>> thanks and regards
>>
>> --
>> _
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>>
>
>
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Re: [asterisk-users] question about CDR

[Asghar Mohammad]

hi,
asterisk insert cdr when call is hangup and last dial statment,
i dont understatnd why you are using 2 dial statment on same extenstion?
if you you want dial to both extensions you can use
506,1,Dial(SIP/223&SIP/276) if you want dial both same time or if you want
to do failover the check Dial status and gotoif dialstatus = NO ANSWER or
what ever you need.



On Thu, May 9, 2013 at 10:46 AM, Salaheddine Elharit <
salah.elharit...@gmail.com> wrote:

> hello list,
>
> i need your help about cdr ,i have installed the module cdr in my asterisk
> 1.4 .
>
> for the inbound calls when i call my sip exten like below :
>
> exten => 506,1,Dial(SIP/223, 10)
> exten => 506,n,Dial(SIP/276, 10)
>
> in CDR i have just one line with SIP /276 the last line but there is no 
> historic
> for the first SIP 223
>
> recid Record ID | calldate   |clid   |src   | dst
> |dcontext |channel | dstchannel   |lastapp |lastdata |duration |billsec
> |disposition |amaflags |accountcode |uniqueid
> |3 |
>
> 626747 |2013-05-09 09:22:55|0661551203  |0661551203|506
>  |default  |Zap/14-1 |SIP/276-092ac7b0 |Dial|SIP/276| 10|21  |0
>  |NO ANSWER
>
>
> any help please to have the historic for 223 and 276
>
> thanks and regards
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] question about CDR

[Salaheddine Elharit]

 thanks i verify but i don't understanding if can someone give me an example

best regards




2013/5/9 Ishfaq Malik 

> On Thu, 2013-05-09 at 08:46 +, Salaheddine Elharit wrote:
> > hello list,
> >
> >
> > i need your help about cdr ,i have installed the module cdr in my
> > asterisk 1.4 .
> >
> >
> > for the inbound calls when i call my sip exten like below :
> >
> >
> > exten => 506,1,Dial(SIP/223, 10)
> > exten => 506,n,Dial(SIP/276, 10)
> >
> >
> > in CDR i have just one line with SIP /276 the last line but there is
> > no historic for the first SIP 223
> >
> >
> > recid Record ID | calldate   |clid   |src   |
> > dst |dcontext |channel | dstchannel   |lastapp |lastdata |duration
> > |billsec |disposition |amaflags |accountcode |uniqueid
> > |3 |
> >
> >
> > 626747 |2013-05-09 09:22:55|0661551203  |0661551203|
> > 506  |default  |Zap/14-1 |SIP/276-092ac7b0 |Dial|SIP/276| 10|21
> >|0  |NO ANSWER
> >
> >
> >
> >
> > any help please to have the historic for 223 and 276
> >
> >
> Hi
>
> You need to look into Channel Event Logging
>
> https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=5242932
>
> Regards
>
> Ish
>
> --
> Ishfaq Malik 
> Department: VOIP Support
> Company: Packnet Limited
> t: +44 (0)845 004 4994
> f: +44 (0)161 660 9825
> e: i...@pack-net.co.uk
> w: http://www.pack-net.co.uk
>
> Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
> NORTH, MANCHESTER
> SCIENCE PARK, MANCHESTER, M156SE
> COMPANY REG NO. 04920552
>
>
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Re: [asterisk-users] question about CDR

[Ishfaq Malik]

On Thu, 2013-05-09 at 08:46 +, Salaheddine Elharit wrote:
> hello list,
> 
> 
> i need your help about cdr ,i have installed the module cdr in my
> asterisk 1.4 .
> 
> 
> for the inbound calls when i call my sip exten like below :
> 
> 
> exten => 506,1,Dial(SIP/223, 10)
> exten => 506,n,Dial(SIP/276, 10)
> 
> 
> in CDR i have just one line with SIP /276 the last line but there is
> no historic for the first SIP 223 
> 
> 
> recid Record ID | calldate   |clid   |src   |
> dst |dcontext |channel | dstchannel   |lastapp |lastdata |duration
> |billsec |disposition |amaflags |accountcode |uniqueid 
> |3 |
> 
> 
> 626747 |2013-05-09 09:22:55|0661551203  |0661551203|
> 506  |default  |Zap/14-1 |SIP/276-092ac7b0 |Dial|SIP/276| 10|21
>|0  |NO ANSWER
> 
> 
> 
> 
> any help please to have the historic for 223 and 276 
> 
> 
Hi

You need to look into Channel Event Logging

https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=5242932

Regards

Ish

-- 
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Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

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NORTH, MANCHESTER
SCIENCE PARK, MANCHESTER, M156SE
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Re: [asterisk-users] question about zapata.conf

[Salaheddine Elharit]

ok thanks for your help and support i really appreciated

2013/3/26 Tzafrir Cohen 

> On Mon, Mar 25, 2013 at 10:44:47AM +, Salaheddine Elharit wrote:
> > hello list,
> >
> > i have a question related to zapata.conf,if i do any change in
> zapata.conf
> > i must restart asterisk or just i restart zapata ,and how to do .
> >
> > “service zaptel restart” or there is any other command
>
> /etc/asterisk/zapata.conf is a configuration ifle of Asterisk's
> chan_zap.so alone. So changes to it would generally require no more than
> restart of Asterisk. The simpler of them would be applied with a simple
> reload (or 'reload chan_zap.so' as you mention).
>
> --
>Tzafrir Cohen
> icq#16849755  jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
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>
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Re: [asterisk-users] question about zapata.conf

[Tzafrir Cohen]

On Mon, Mar 25, 2013 at 10:44:47AM +, Salaheddine Elharit wrote:
> hello list,
> 
> i have a question related to zapata.conf,if i do any change in zapata.conf
> i must restart asterisk or just i restart zapata ,and how to do .
> 
> “service zaptel restart” or there is any other command

/etc/asterisk/zapata.conf is a configuration ifle of Asterisk's
chan_zap.so alone. So changes to it would generally require no more than
restart of Asterisk. The simpler of them would be applied with a simple
reload (or 'reload chan_zap.so' as you mention).

-- 
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Re: [asterisk-users] question about zapata.conf

[Tzafrir Cohen]

On Mon, Mar 25, 2013 at 03:15:24PM +, Salaheddine Elharit wrote:
> thank you so much
> 
> fo the upgrade from zptel to dahdi, if there is any possibility to upgrade
> to dahdi without impacting my installation of asterisk and other
> application already installed in my server.
> 
> if you can tell how to upgrade using dahdi drivers

Asterisk 1.4 is at build time set to use either DAHDI or Zaptel (but not
both). (try: 'strings /usr/sbin/asterisk | grep /dev'). So you'll have
to at least rebuild Asterisk vs. DAHDI.

Asterisk of older versions does not support DAHDI at all.

You should also note that even the branch 1.4.x is no longer actively
supported, and this would be a good time to upgrade.

-- 
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Re: [asterisk-users] question about zapata.conf

[Salaheddine Elharit]

ok thank you so much for your help and support

2013/3/25 Yves A. 

>  hi,
> migrating from zaptel to dahdi HAS an impact... new config files, new
> options and a new channeldriver that has to be
> used in your dialplan ... you would have to select the DAHDI channel
> instead of your ZAP channel when dialing...
> if you´re to afraid to do it... then leave it as it is and follow the
> ntars-maxime (never touch a running system)...
> regards,
> yves
>
> Am 25.03.2013 16:15, schrieb Salaheddine Elharit:
>
>  thank you so much
>
>  fo the upgrade from zptel to dahdi, if there is any possibility to
> upgrade to dahdi without impacting my installation of asterisk and other
> application already installed in my server.
>
>  if you can tell how to upgrade using dahdi drivers
>
>  thanks and best regards
>
>
> 2013/3/25 Eric Wieling 
>
>> Service asterisk stop
>> Service zaptel restart
>> Service asterisk start
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit
>> Sent: Monday, March 25, 2013 11:04 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] question about zapata.conf
>>
>> i use asterisk 1.4, how i can do to reload dirver
>>
>> 1.service asterisk stop
>> 2 CLI> reload chan_zap.so
>> 3 service asterisk start
>>  that is right or i miss something ?
>>
>>
>>
>>
>>
>> 2013/3/25 Yves A. 
>>
>>
>> it depends a little bit on the driver and asterisk version...
>> the safest way to become changes applied is to stop asterisk,
>> reload the driver and than start asterisk again.
>>
>> regards,
>> yves
>>
>> btw..:
>> zaptel ist outdated... you should definitely upgrade using dahdi
>> drivers...
>>
>>
>> Am 25.03.2013 11:44, schrieb Salaheddine Elharit:
>>
>>
>> hello list,
>>
>> i have a question related to zapata.conf,if i do any
>> change in zapata.conf i must restart asterisk or just i restart zapata ,and
>> how to do .
>>
>> "service zaptel restart" or there is any other command
>>
>> Thanks and regards
>>
>>
>>
>>
>>
>> --
>>
>> _
>> -- Bandwidth and Colocation Provided by
>> http://www.api-digital.com --
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>> every Thurs:
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>>
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>>
>>
>>
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Re: [asterisk-users] question about zapata.conf

[Yves A.]

hi,
migrating from zaptel to dahdi HAS an impact... new config files, new 
options and a new channeldriver that has to be
used in your dialplan ... you would have to select the DAHDI channel 
instead of your ZAP channel when dialing...
if you´re to afraid to do it... then leave it as it is and follow the 
ntars-maxime (never touch a running system)...

regards,
yves

Am 25.03.2013 16:15, schrieb Salaheddine Elharit:

thank you so much

fo the upgrade from zptel to dahdi, if there is any possibility to 
upgrade to dahdi without impacting my installation of asterisk and 
other application already installed in my server.


if you can tell how to upgrade using dahdi drivers

thanks and best regards


2013/3/25 Eric Wieling mailto:ewiel...@nyigc.com>>

Service asterisk stop
Service zaptel restart
Service asterisk start

-Original Message-
From: asterisk-users-boun...@lists.digium.com
<mailto:asterisk-users-boun...@lists.digium.com>
[mailto:asterisk-users-boun...@lists.digium.com
<mailto:asterisk-users-boun...@lists.digium.com>] On Behalf Of
Salaheddine Elharit
Sent: Monday, March 25, 2013 11:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] question about zapata.conf

i use asterisk 1.4, how i can do to reload dirver

1.service asterisk stop
2 CLI> reload chan_zap.so
3 service asterisk start
 that is right or i miss something ?





2013/3/25 Yves A. mailto:yves...@gmx.de>>


it depends a little bit on the driver and asterisk version...
the safest way to become changes applied is to stop
asterisk, reload the driver and than start asterisk again.

regards,
yves

btw..:
zaptel ist outdated... you should definitely upgrade using
dahdi drivers...


Am 25.03.2013 11:44, schrieb Salaheddine Elharit:


hello list,

i have a question related to zapata.conf,if i do
any change in zapata.conf i must restart asterisk or just i
restart zapata ,and how to do .

"service zaptel restart" or there is any other command

Thanks and regards





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Re: [asterisk-users] question about zapata.conf

[Salaheddine Elharit]

thank you so much

fo the upgrade from zptel to dahdi, if there is any possibility to upgrade
to dahdi without impacting my installation of asterisk and other
application already installed in my server.

if you can tell how to upgrade using dahdi drivers

thanks and best regards


2013/3/25 Eric Wieling 

> Service asterisk stop
> Service zaptel restart
> Service asterisk start
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit
> Sent: Monday, March 25, 2013 11:04 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] question about zapata.conf
>
> i use asterisk 1.4, how i can do to reload dirver
>
> 1.service asterisk stop
> 2 CLI> reload chan_zap.so
> 3 service asterisk start
>  that is right or i miss something ?
>
>
>
>
>
> 2013/3/25 Yves A. 
>
>
> it depends a little bit on the driver and asterisk version...
> the safest way to become changes applied is to stop asterisk,
> reload the driver and than start asterisk again.
>
> regards,
> yves
>
> btw..:
> zaptel ist outdated... you should definitely upgrade using dahdi
> drivers...
>
>
> Am 25.03.2013 11:44, schrieb Salaheddine Elharit:
>
>
> hello list,
>
> i have a question related to zapata.conf,if i do any
> change in zapata.conf i must restart asterisk or just i restart zapata ,and
> how to do .
>
> "service zaptel restart" or there is any other command
>
> Thanks and regards
>
>
>
>
>
> --
>
> _
> -- Bandwidth and Colocation Provided by
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Re: [asterisk-users] question about zapata.conf

[Eric Wieling]

Service asterisk stop
Service zaptel restart
Service asterisk start

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine 
Elharit
Sent: Monday, March 25, 2013 11:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] question about zapata.conf

i use asterisk 1.4, how i can do to reload dirver 

1.service asterisk stop
2 CLI> reload chan_zap.so 
3 service asterisk start
 that is right or i miss something ?





2013/3/25 Yves A. 


it depends a little bit on the driver and asterisk version...
the safest way to become changes applied is to stop asterisk, reload 
the driver and than start asterisk again.

regards,
yves

btw..:
zaptel ist outdated... you should definitely upgrade using dahdi 
drivers...


Am 25.03.2013 11:44, schrieb Salaheddine Elharit:


hello list,

i have a question related to zapata.conf,if i do any change in 
zapata.conf i must restart asterisk or just i restart zapata ,and how to do .

"service zaptel restart" or there is any other command 

Thanks and regards 


 


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Re: [asterisk-users] question about zapata.conf

[Salaheddine Elharit]

i use asterisk 1.4, how i can do to reload dirver

1.service asterisk stop
2 CLI> reload chan_zap.so
3 service asterisk start
 that is right or i miss something ?




2013/3/25 Yves A. 

>  it depends a little bit on the driver and asterisk version...
> the safest way to become changes applied is to stop asterisk, reload the
> driver and than start asterisk again.
>
> regards,
> yves
>
> btw..:
> zaptel ist outdated... you should definitely upgrade using dahdi drivers...
>
>
> Am 25.03.2013 11:44, schrieb Salaheddine Elharit:
>
>  hello list,
>
>  i have a question related to zapata.conf,if i do any change in
> zapata.conf i must restart asterisk or just i restart zapata ,and how to do
> .
>
>  “service zaptel restart” or there is any other command
>
>  Thanks and regards
>
>
>
> --
> _
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>
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Re: [asterisk-users] question about zapata.conf

[Yves A.]

it depends a little bit on the driver and asterisk version...
the safest way to become changes applied is to stop asterisk, reload the 
driver and than start asterisk again.


regards,
yves

btw..:
zaptel ist outdated... you should definitely upgrade using dahdi drivers...


Am 25.03.2013 11:44, schrieb Salaheddine Elharit:

hello list,

i have a question related to zapata.conf,if i do any change in 
zapata.conf i must restart asterisk or just i restart zapata ,and how 
to do .


"service zaptel restart" or there is any other command

Thanks and regards



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Re: [asterisk-users] question on SIP trunk and AMI to place call

[Jerry Geis]

I just put a break at dial_exec_full (app/app_dial.c for Asterisk 11.0.2)
did my AMI call

Action: Originate
Async: yes
Channel: SIP/testsystem/XXX

(calls from my machine over SIP trunk to another 11.0.2 box that has
a PRI card to make a call out to my cell)

and did not get a break.

Why is a SIP call not logging the Dial event as a DAHDI call does???

jerry



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Re: [asterisk-users] question on SIP trunk and AMI to place call

[Jerry Geis]

The Dial events are created by app_dial. So long as you are using
app_dial to create your outbound channel, you should have that event.
Channel technology shouldn't matter.



I am using the same AMI method to start both calls.
Action: Originate
Channel: DAHDI/18/XX
or
Action: Originate
Channel: SIP/machine/XX

Jerry
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Re: [asterisk-users] question on SIP trunk and AMI to place call

[Matthew Jordan]

On 01/24/2013 01:13 PM, Jerry Geis wrote:
>>
>>
>> You probably want the Dial event. It is raised both at the beginning of
>> the Dial, as well as when the Dial completes.
>>
>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial
>>
>> Note that the Channel: field will contain the name initiating the Dial,
>> the Destination: field will contain the channel name being dialled, and
>> the Dialstring: field will contain the non-technology specific portion
>> of the thing being dialled.
> I get that even on the system with the PRI card and using DAHDI
> however I am not getting that event on the system with the SIP trunk .
> 
> Is there something to enable to get that???
> Both systems are running Asterisk 11.0.2.
> 

The Dial events are created by app_dial. So long as you are using
app_dial to create your outbound channel, you should have that event.
Channel technology shouldn't matter.

-- 
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Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org



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Re: [asterisk-users] question on SIP trunk and AMI to place call

[Danny Nicholas]

This might have changed but IIRC /etc/asterisk/manager.conf controls what
events you have access to.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, January 24, 2013 1:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] question on SIP trunk and AMI to place call

 

 

 
 
You probably want the Dial event. It is raised both at the beginning of
the Dial, as well as when the Dial completes.
 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial
 
Note that the Channel: field will contain the name initiating the Dial,
the Destination: field will contain the channel name being dialled, and
the Dialstring: field will contain the non-technology specific portion
of the thing being dialled.

I get that even on the system with the PRI card and using DAHDI
however I am not getting that event on the system with the SIP trunk .

Is there something to enable to get that???
Both systems are running Asterisk 11.0.2.

Thanks,

Jerry

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Re: [asterisk-users] question on SIP trunk and AMI to place call

[Jerry Geis]

You probably want the Dial event. It is raised both at the beginning of
the Dial, as well as when the Dial completes.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial

Note that the Channel: field will contain the name initiating the Dial,
the Destination: field will contain the channel name being dialled, and
the Dialstring: field will contain the non-technology specific portion
of the thing being dialled.

I get that even on the system with the PRI card and using DAHDI
however I am not getting that event on the system with the SIP trunk .

Is there something to enable to get that???
Both systems are running Asterisk 11.0.2.

Thanks,

Jerry
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Re: [asterisk-users] question on SIP trunk and AMI to place call

[Matthew Jordan]

On 01/24/2013 10:46 AM, Jerry Geis wrote:
> When I am monitoring the AMI I see the following event
> for a call I just made over a SIP trunk.
> 
> Event: Newchannel
> Privilege: call,all
> Channel: SIP/testmachine-000d
> ChannelState: 0
> ChannelStateDesc: Down
> CallerIDNum:
> CallerIDName:
> AccountCode:
> Exten:
> Context: testmachine
> Uniqueid: 1359035395.20
> 
> In this event or any event following I do not see
> the phone number that I dialled. How do I "correlate"
> the "SIP/testmachine-000d" to the number I just dialed
> (purpose is to hangup the call later if I need to interrupt it)
> 
> Now if I am using a machine with actual hardware cards, the phone
> number is included as part of the Channel so I can look that up.
> but for a SIP trunk the phone number dialled does not come over the AMI.
> 
> How do I match up the call I just started (using AMI over SIP trunk) to
> the number I called?
> 

You probably want the Dial event. It is raised both at the beginning of
the Dial, as well as when the Dial completes.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial

Note that the Channel: field will contain the name initiating the Dial,
the Destination: field will contain the channel name being dialled, and
the Dialstring: field will contain the non-technology specific portion
of the thing being dialled.

For example:

exten => 500,1,Dial(SIP/digium02)

Results in:

Event: Dial
Privilege: call,all
SubEvent: Begin
Channel: SIP/10.x.x.x-0002
Destination: SIP/digium02-0003
CallerIDNum: 657-5309
CallerIDName: digium01
ConnectedLineNum: 
ConnectedLineName: 
UniqueID: Asterisk-01-1359052866.2
DestUniqueID: Asterisk-01-1359052866.3
Dialstring: digium02

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



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Re: [asterisk-users] question on SIP trunk and AMI to place call

[Jerry Geis]

Not the greatest solution, but since you are most likely using a script for the 
AMI process, you could do an

Asterisk --rx "core show channels verbose"|grep SIP/testmachine-000d

And get the dialed number from that.

Actually you could issue the AMI command core show channels verbose.
there is no core show channels verbose on Asterisk 11. There is on 
asterisk 1.4,


core show channels on asterisk 11 has been changed.

jerry
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Re: [asterisk-users] question on SIP trunk and AMI to place call

[Jerry Geis]

Have you tried and looked up all events generated when you place the call?

some of them are bound to have the variable callerid set
yes I have looked at all of them, CallerID is not set to the number I am 
calling.


Jerry
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Re: [asterisk-users] question on SIP trunk and AMI to place call

[Danny Nicholas]

Not the greatest solution, but since you are most likely using a script for the 
AMI process, you could do an 

Asterisk –rx “core show channels verbose”|grep SIP/testmachine-000d 

And get the dialed number from that.

Actually you could issue the AMI command core show channels verbose.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada
Sent: Thursday, January 24, 2013 11:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] question on SIP trunk and AMI to place call

 

Have you tried and looked up all events generated when you place the call?

 

some of them are bound to have the variable callerid set

 

On 24 January 2013 16:46, Jerry Geis  wrote:

When I am monitoring the AMI I see the following event
for a call I just made over a SIP trunk.

Event: Newchannel
Privilege: call,all
Channel: SIP/testmachine-000d
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
AccountCode:
Exten:
Context: testmachine
Uniqueid: 1359035395.20

In this event or any event following I do not see
the phone number that I dialled. How do I "correlate"
the "SIP/testmachine-000d" to the number I just dialed
(purpose is to hangup the call later if I need to interrupt it)

Now if I am using a machine with actual hardware cards, the phone
number is included as part of the Channel so I can look that up.
but for a SIP trunk the phone number dialled does not come over the AMI.

How do I match up the call I just started (using AMI over SIP trunk) to the 
number I called?

Thanks,

jerry



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Re: [asterisk-users] question on SIP trunk and AMI to place call

[Tiago Geada]

Have you tried and looked up all events generated when you place the call?

some of them are bound to have the variable callerid set


On 24 January 2013 16:46, Jerry Geis  wrote:

> When I am monitoring the AMI I see the following event
> for a call I just made over a SIP trunk.
>
> Event: Newchannel
> Privilege: call,all
> Channel: SIP/testmachine-000d
> ChannelState: 0
> ChannelStateDesc: Down
> CallerIDNum:
> CallerIDName:
> AccountCode:
> Exten:
> Context: testmachine
> Uniqueid: 1359035395.20
>
> In this event or any event following I do not see
> the phone number that I dialled. How do I "correlate"
> the "SIP/testmachine-000d" to the number I just dialed
> (purpose is to hangup the call later if I need to interrupt it)
>
> Now if I am using a machine with actual hardware cards, the phone
> number is included as part of the Channel so I can look that up.
> but for a SIP trunk the phone number dialled does not come over the AMI.
>
> How do I match up the call I just started (using AMI over SIP trunk) to
> the number I called?
>
> Thanks,
>
> jerry
>
>
>
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Re: [asterisk-users] Question on variables and asterisk 11

[A J Stiles]

On Saturday 08 December 2012, Jerry Geis wrote:
> I am sending something like this to the AMI (along with other commands):
> 
> Variable: agi_pa_list=box1,box2,
> 
> however, in the dialplan I do:
> exten => app_confbridge_call_out,n,noop(${agi_pa_list})
> 
> and it ONLY reports box1 and it should be "box1,box2"
> 
> Is something different in Asterisk 11 vs 1.4.43 in this regard?

You probably need to put speech marks around values with a comma in.  Commas 
are now used as delimiters in modern Asterisk versions, where ancient versions 
used to use pipe symbols.

-- 
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Answers come *after* questions.

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Re: [asterisk-users] Question on AMI and ChanIsAvail

[Aldo Bergamini]

On 18 Oct 2012, at 17:58, Jerry Geis wrote:

> I was wanting to call ChanIsAvail from AMI.
> 
> I logged in and issues command,
> 
> Action: Command
> Command: ChanIsAvail DAHDI/1
> 
> my response was this:
> event_list=0 ret=158 Response: Follows[CR ][LF ]Privilege: Command[CR ][LF 
> ]No such command 'ChanIsAvail DAHDI/1'
> 
> Is there any way to tell if a channel is available through AMI?
> Did I format my request wrong?
> I am on 1.4.43.
> 
> Thanks,
> 
> Jerry


You can get (on the CLI) the list of AMI commands supported by your Asterisk 
installation issuing "manager show commands".

The AMI command that tells you what a terminal is doing at any time is 
"ExtensionState" (I guess it's available on 1.4..). But if you add hints to 
your dialplan, then Asterisk will send an AMI event whenever a terminal in the 
list of hints is changing state.

You would need to listen to all such events and keep track of each 
line/extension; issuing a list of ExtensionState events only when you launch 
your tracking process.

Cheers,
Aldo


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Re: [asterisk-users] Question on AMI and ChanIsAvail

[Danny Nicholas]

Using AMI you might want to use hints instead of ChanIsAvail since that
command puts its' output back into the dialplan as variables. 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, October 18, 2012 10:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question on AMI and ChanIsAvail

I was wanting to call ChanIsAvail from AMI.

I logged in and issues command,

Action: Command
Command: ChanIsAvail DAHDI/1

my response was this:
event_list=0 ret=158 Response: Follows[CR ][LF ]Privilege: Command[CR ][LF
]No such command 'ChanIsAvail DAHDI/1'

Is there any way to tell if a channel is available through AMI?
Did I format my request wrong?
I am on 1.4.43.

Thanks,

Jerry

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Re: [asterisk-users] question on softhangup

[Mitch Claborn]

Dave Platt provided the following answer to a similar question of mine 
last week.  I was trying to use SoftHangup() to prempt a DAHDI line for 
an emergency call.  Here is his reply.


That may be due to a common characteristic of PSTN lines (at least,
it's common here in the U.S.)

By design, most U.S. PSTN lines have a very asymmetrical response
to a physical hangup:

-  If the calling party hangs up, the call is terminated
   immediately.

-  If the called party hangs up, and the calling party does not,
   the line remains "live" for some time (typically around 30
   seconds, I believe).  If the called party goes off-hook again
   during this period, they can resume the call.

If I recall correctly, things were designed this way so that
the called party could say "Oh, hang on, I answered this call
in the bedroom and the stuff I need is in the living room",
hang up the extension phone, go to another room, pick up the
other phone and carry on with the call.

If that's what you're running into here - if the line you
are trying to SoftHangup() was handing an inbound call - then
there may be no good solution.  As far as I know, there is no
way to force an incoming PSTN call to release the line, other
than "go on-hook, and wait for 30 seconds to pass".

Several possible workarounds, roughly in order of increasing
complexity and decreasing reliability:

(1) Keep one of your PSTN lines reserved for emergency calls
only;  remove it from your inbound hunt group and place
it in a Dahdi line group of its own (or don't group it at
all).

(2) Keep one of your PSTN lines reserved for *outbound* calls
only;  you should be able to SoftHangup() an outbound call
within a second or two.

(3) Figure out a way to check the PSTN lines that are in use
at the time of an emergency - if they're all in use,
somehow find one which was in use for an outbound call,
and select it as the one to SoftHangup() and dial upon.

(4) If you must keep all of your PSTN lines in bidirectional
use, you may have to *tell* the parties that the line is
needed for an emergency call, and ask them to release the
line.  Do a barge-in on the channel, play an alert sound,
play a message saying "Emergency call in progress, please hang
up this line immediately, play the alert sound again for
a few seconds, SoftHangup(), Wait(2), and then try dialing.


Mitch

On 10/16/2012 08:59 PM, Jerry Geis wrote:

How do I use softhangup through the AMI interface?

I am using 1.4.43. Will softhangup hangup a DAHDI channel?

I have found that "Action: Hangup" does not hangup a DAHDI channel only
sip.

Thanks,

jerry

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Re: [asterisk-users] Question about cli

[Shitian Long]

As far as I know there is no function add SIP extension directly from CLI, and 
actually, it is not convenience that add an extension from CLI as well. 


On Aug 31, 2012, at 4:18 PM, Giuseppe Longo  wrote:

> Hello guys,
> i would like to ask a question about cli.
> 
> Today, while i was using the cli, i thinked that there could be more features.
> IMHO, might be interesting, for example, to add a sip extensions from
> cli, or other similar functions, without having to modify the
> configuration files.
> 
> Or not? What do you think?
> 
> Regards
> 
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Re: [asterisk-users] Question about cli

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo
Sent: Friday, August 31, 2012 9:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question about cli

Hello guys,
i would like to ask a question about cli.

Today, while i was using the cli, i thinked that there could be more
features.
IMHO, might be interesting, for example, to add a sip extensions from cli,
or other similar functions, without having to modify the configuration
files.

Or not? What do you think?

Regards

Many things like that Can be done from CLI - the wisdom or danger of that
practice depends on who's at the keyboard.


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Re: [asterisk-users] Question on app_confbridge

[Jerry Geis]

On 08/10/2012 11:23 AM, Jerry Geis wrote:

I have a profile in confbridge
[MessageNetConfBridge]
and more...

Asterisk is reading it at startup.
[1;30m  == ^[[0mParsing '/etc/asterisk/confbridge.conf': ^[[1;30m  == 
^[[0mFound
^[[1;30m ^[[0mapp_confbridge.so => (^[[0;33mConference Bridge 
Application^[[0m)


When I try to use it I get a warning about
 WARNING[20678] app_confbridge.c: Conference bridge profile  
MessageNetConfBridge does not exist



My dialplan looks like:
exten => new_app_confbridge,1,ConfBridge(${agi_pa_meetme}, 
MessageNetConfBridge)


Not sure how its not picking it up?

What did I miss?

Jerry

Clearly I have conf bridge:
confbridge show profile bridge MessageNetConfBridge

Name: MessageNetConfBridge
Internal Sample Rate: 8000
Mixing Interval:  Default 20ms
Record Conference:no
Record File:  Auto Generated
Max Members:  No Limit
Video Mode:   no video
sound_join:   confbridge-join
sound_leave:  confbridge-leave
sound_only_person:conf-onlyperson
sound_has_joined: conf-hasjoin
sound_has_left:   conf-hasleft
sound_kicked: conf-kicked
sound_muted:  conf-muted
sound_unmuted:conf-unmuted
sound_there_are:  conf-thereare
sound_other_in_party: conf-otherinparty
sound_place_into_conference: conf-placeintoconf
sound_wait_for_leader:   conf-waitforleader
sound_leader_has_left:   conf-leaderhasleft
sound_get_pin:conf-getpin
sound_invalid_pin:conf-invalidpin
sound_locked: conf-locked
sound_unlocked_now:   conf-unlockednow
sound_lockednow:  conf-lockednow
sound_error_menu: conf-errormenu

So why is it telling me I do not?

jerry

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Re: [asterisk-users] Question for a Jira bug marshal

[Matthew Jordan]

- Original Message -
> From: "Administrator TOOTAI" 
> To: asterisk-users@lists.digium.com
> Sent: Tuesday, April 24, 2012 5:19:36 AM
> Subject: Re: [asterisk-users] Question for a Jira bug marshal
> 
> Le 24/04/2012 09:37, Stefan Schmidt a écrit :
> > Am 24.04.12 09:27, schrieb Administrator TOOTAI:
> >> Hello,
> >>
> >> I opened bug #19763 on jira last friday (20/04) and didn't get any
> >> feedback till now. Is this a normal delay?
> >>
> >> Regards
> >>

You may have noticed that we did some security releases yesterday; a
few of us were tied up with that.  Issue triage is something that we
attempt to do several days a week, but it is not always possible to
do it every day.

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

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Re: [asterisk-users] Question for a Jira bug marshal

[Administrator TOOTAI]

Le 24/04/2012 09:37, Stefan Schmidt a écrit :

Am 24.04.12 09:27, schrieb Administrator TOOTAI:

Hello,

I opened bug #19763 on jira last friday (20/04) and didn't get any
feedback till now. Is this a normal delay?

Regards


Hi,

i didnt want to say that it is a "normal delay" but most bug marshals
and devs work on asterisk bugs in their free time. So it could take some
time for an response.


Well, it's the first time since years that I post on issues that it took 
so long to get at least one feedback, that's why I was wondering ;-)




btw you allready got some response on it ;)


I saw, many thanks :)

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Re: [asterisk-users] Question for a Jira bug marshal

[Stefan Schmidt]

Am 24.04.12 09:27, schrieb Administrator TOOTAI:
> Hello,
> 
> I opened bug #19763 on jira last friday (20/04) and didn't get any
> feedback till now. Is this a normal delay?
> 
> Regards
> 

Hi,

i didnt want to say that it is a "normal delay" but most bug marshals
and devs work on asterisk bugs in their free time. So it could take some
time for an response.

btw you allready got some response on it ;)

best regards

stefan

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Re: [asterisk-users] Question about asterisk to Cisco

[Danny Nicholas]

According to the documentation yes, but YMMV.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Friday, April 13, 2012 1:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question about asterisk to Cisco

If I have a connection from Asterisk to Cisco Call Manager using SIP can I
send a text message using  "SendText" from asterisk across the SIP trunk to
CCM and it convert that to text message to the skinny cisco phones?

Does that work?

Thanks,

Jerry


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Re: [asterisk-users] Question for the group

[Steve Edwards]

Great subject, regardless of the name you post from or the list you post 
to.


I'm sure this is the only question that will ever be asked of either 
group.


Better subject = better replies = better value for future archive 
searchers.


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-
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Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Question for the group

[C. Savinovich]

>>>Thanks for this - but I am looking really for a software type solution.I would venture say that he means he wants it for free.Christian SavinovichVoIP & Telephony Consultant646-982-3572 


 Original Message ----
Subject: Re: [asterisk-users] Question for the group
From: James Wystead <szilvertho...@gmail.com>
Date: Fri, February 10, 2012 11:57 am
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>

Yes, I like the look of that.Researching it too - the commercial one looks nice too, but I don't know if there is a budget.GOn Fri, Feb 10, 2012 at 11:52, Terry Brummell <te...@brummell.net> wrote: I assume that solution was A2Billing?  -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Friday, February 10, 2012 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question for the group  - Original Message - > Hello Folks; > > I know this is a non-commercial discussion group, but I am looking for > some open-source software suggestions > > > We are going to be setting up a prepaid PBX service with the following > features: > > > • Email to Fax and Fax to Email > • Inward DID local and 800 services > • Calling card SIP based and ANI authenticated > > > I see there are many types of software that can be addons/installs/etc > to Asterisk. > > So, the question that I ask is which one would be best suited for > these needs? Of course, it needs to be scalable and work well (most > opensource software does) > > So, any thoughts? >  You just posted this to the asterisk-biz list under a different name/email address. The one response you received was immediately brushed off because you apparently cannot read: "Thanks for this - but I am looking really for a software type solution".  The product offered *IS A SOFTWARE SOLUTION* that would run on your hardware. The posted option is more than suitable to your needs, and offered by folks with a highly deserved great reputation.  Good luck to you.  --tim  -- _ -- 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 -- _ -- 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] Question for the group

[asterisk jobs]

What have you looked for yet? There are no commercial ones that do all that
in one.

On Fri, Feb 10, 2012 at 11:57 AM, James Wystead wrote:

> Yes, I like the look of that.
>
> Researching it too - the commercial one looks nice too, but I don't know
> if there is a budget.
>
> G
>
>
> On Fri, Feb 10, 2012 at 11:52, Terry Brummell  wrote:
>
>> I assume that solution was A2Billing?
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
>> Sent: Friday, February 10, 2012 11:41 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Question for the group
>>
>> - Original Message -
>> > Hello Folks;
>> >
>> > I know this is a non-commercial discussion group, but I am looking for
>> > some open-source software suggestions
>> >
>> >
>> > We are going to be setting up a prepaid PBX service with the following
>> > features:
>> >
>> >
>> > • Email to Fax and Fax to Email
>> > • Inward DID local and 800 services
>> > • Calling card SIP based and ANI authenticated
>> >
>> >
>> > I see there are many types of software that can be addons/installs/etc
>> > to Asterisk.
>> >
>> > So, the question that I ask is which one would be best suited for
>> > these needs? Of course, it needs to be scalable and work well (most
>> > opensource software does)
>> >
>> > So, any thoughts?
>> >
>>
>> You just posted this to the asterisk-biz list under a different
>> name/email address. The one response you received was immediately brushed
>> off because you apparently cannot read: "Thanks for this - but I am looking
>> really for a software type solution".  The product offered *IS A SOFTWARE
>> SOLUTION* that would run on your hardware. The posted option is more than
>> suitable to your needs, and offered by folks with a highly deserved great
>> reputation.
>>
>> Good luck to you.
>>
>> --tim
>>
>> --
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>>
>
>
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Re: [asterisk-users] Question for the group

[James Wystead]

Yes, I like the look of that.

Researching it too - the commercial one looks nice too, but I don't know if
there is a budget.

G

On Fri, Feb 10, 2012 at 11:52, Terry Brummell  wrote:

> I assume that solution was A2Billing?
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
> Sent: Friday, February 10, 2012 11:41 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Question for the group
>
> - Original Message -
> > Hello Folks;
> >
> > I know this is a non-commercial discussion group, but I am looking for
> > some open-source software suggestions
> >
> >
> > We are going to be setting up a prepaid PBX service with the following
> > features:
> >
> >
> > • Email to Fax and Fax to Email
> > • Inward DID local and 800 services
> > • Calling card SIP based and ANI authenticated
> >
> >
> > I see there are many types of software that can be addons/installs/etc
> > to Asterisk.
> >
> > So, the question that I ask is which one would be best suited for
> > these needs? Of course, it needs to be scalable and work well (most
> > opensource software does)
> >
> > So, any thoughts?
> >
>
> You just posted this to the asterisk-biz list under a different name/email
> address. The one response you received was immediately brushed off because
> you apparently cannot read: "Thanks for this - but I am looking really for
> a software type solution".  The product offered *IS A SOFTWARE SOLUTION*
> that would run on your hardware. The posted option is more than suitable to
> your needs, and offered by folks with a highly deserved great reputation.
>
> Good luck to you.
>
> --tim
>
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Re: [asterisk-users] Question for the group

[asterisk jobs]

There is none. We are looking to develop our own currently and in the
process of hunting down best developers. We have a great deal of experience
with billing systems but doing a fron-end for this purpose just requires
multiple developers. You can e-mail me in private if interested in a shared
project.

Best,

On Fri, Feb 10, 2012 at 11:34 AM, James Wystead wrote:

> Hello Folks;
>
> I know this is a non-commercial discussion group, but I am looking for
> some open-source software suggestions
>
>
> We are going to be setting up a prepaid PBX service with the following
> features:
>
>
> *
>>
>>- Email to Fax and  Fax to Email
>>- Inward DID local and 800 services
>>- Calling card SIP based and ANI authenticated
>>
>> *
>>
>
> I see there are many types of software that can be addons/installs/etc to
> Asterisk.
>
> So, the question that I ask is which one would be best suited for these
> needs? Of course, it needs to be scalable and work well (most opensource
> software does)
>
> So, any thoughts?
>
> Thanks
>
> G
>
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Re: [asterisk-users] Question for the group

[Terry Brummell]

I assume that solution was A2Billing?

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Friday, February 10, 2012 11:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question for the group

- Original Message -
> Hello Folks;
> 
> I know this is a non-commercial discussion group, but I am looking for 
> some open-source software suggestions
> 
> 
> We are going to be setting up a prepaid PBX service with the following
> features:
> 
> 
> • Email to Fax and Fax to Email
> • Inward DID local and 800 services
> • Calling card SIP based and ANI authenticated
> 
> 
> I see there are many types of software that can be addons/installs/etc 
> to Asterisk.
> 
> So, the question that I ask is which one would be best suited for 
> these needs? Of course, it needs to be scalable and work well (most 
> opensource software does)
> 
> So, any thoughts?
> 

You just posted this to the asterisk-biz list under a different name/email 
address. The one response you received was immediately brushed off because you 
apparently cannot read: "Thanks for this - but I am looking really for a 
software type solution".  The product offered *IS A SOFTWARE SOLUTION* that 
would run on your hardware. The posted option is more than suitable to your 
needs, and offered by folks with a highly deserved great reputation.

Good luck to you.

--tim

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Re: [asterisk-users] Question for the group

[Tim Nelson]

- Original Message -
> Hello Folks;
> 
> I know this is a non-commercial discussion group, but I am looking for
> some open-source software suggestions
> 
> 
> We are going to be setting up a prepaid PBX service with the following
> features:
> 
> 
> • Email to Fax and Fax to Email
> • Inward DID local and 800 services
> • Calling card SIP based and ANI authenticated
> 
> 
> I see there are many types of software that can be addons/installs/etc
> to Asterisk.
> 
> So, the question that I ask is which one would be best suited for
> these needs? Of course, it needs to be scalable and work well (most
> opensource software does)
> 
> So, any thoughts?
> 

You just posted this to the asterisk-biz list under a different name/email 
address. The one response you received was immediately brushed off because you 
apparently cannot read: "Thanks for this - but I am looking really for a 
software type solution".  The product offered *IS A SOFTWARE SOLUTION* that 
would run on your hardware. The posted option is more than suitable to your 
needs, and offered by folks with a highly deserved great reputation.

Good luck to you.

--tim

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Re: [asterisk-users] question on CDR

[Danny Nicholas]

The information should be available in the ${EXTEN} variable, but you will
have to probably record it somewhere other than the default 1.4 CDR.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, January 05, 2012 12:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] question on CDR

I used my cell to call in and create a CDR record here from asterisk 1.4.43:

"","317XXX","s","default","""GEIS JERRY "" 
<317XXX>","DAHDI/23-1","","BackGround","SM_ATTENDANT","2012-01-05
18:12:09","2012-01-05 18:12:10","2012-01-05
18:12:18",9,8,"ANSWERED","DOCUMENTATION","1325787129.624",""

I am surprised that the number DIALED or DNIS is not present in the CDR.
How do I get that as part of the CDR?

I am using a PRI so that information should be delivered and available.

Thanks,

jerry


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Re: [asterisk-users] question sangoma vs digium

[James zhu]

hello:
i think it can be done, please refer this link:
http://wiki.sangoma.com/Asterisk-FAQ#Digium
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri<->SIP).
website: www.voipviews.com 


Date: Wed, 4 Jan 2012 18:47:28 -0200
From: agustina.berre...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] question sangoma vs digium

Hi!
Hello! I wanted to know if you have experienced problems installing both a 
Sangoma and a Digium card in the same Server.

Thnks a lot!


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Re: [asterisk-users] Question on system command 1.4.43

[Eric Wieling]

You should confirm with "ps -aux | grep asterisk"

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Tuesday, January 03, 2012 1:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question on system command 1.4.43

On 01/03/2012 08:48 AM, Jerry Geis wrote:
> I have a USB to serial converter attached to my box.  pl2303: Prolific
> PL2303 USB to serial adaptor driver
> if I login to the box and send/receive serial commands over this unit 
> it works without error EVERY time.
>
> however, if I run the same command set from with-in the 
> extensions.conf with System() I get errors in dmesg like "pl2303 
> ttyUSB0: pl2303_open - failed submitting read urb, error -22"
> and then obviusly my command does not work.
>
> however, again - If I go back to the command line and my command it 
> runs just fine.
>
> Any ideas on what might be happening here?
>
> Jerry
>
> extensions.conf
> exten => s,1,ChanIsAvail(Console/Dsp)
> exten => s,n,GotoIf($["${AVAILCHAN}" = ""]?smvoice-busy,s,1) exten => 
> s,n,System(/home/silentm/bin/usbserial -start) exten => 
> s,n,Playback(beep) exten => s,n,Dial(Console/dsp) exten => s,n,Hangup 
> exten => h,1,System(/home/silentm/bin/usbserial -stop)
>
>
Jim and Doug

Yes the process is running as root. I should have mentioned that.
Also - yes the usbserial is a paging notification program.

So if its not permissions (since its running as root) what else might it be?

Thanks,

Jerry

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Re: [asterisk-users] Question on system command 1.4.43

[Steve Edwards]

On 01/03/2012 08:48 AM, Jerry Geis wrote:



I have a USB to serial converter attached to my box.


if I login to the box and send/receive serial commands over this unit 
it works without error EVERY time.


however, if I run the same command set from with-in the extensions.conf 
with System() I get errors in dmesg like "pl2303 ttyUSB0: pl2303_open - 
failed submitting read urb, error -22"


On Tue, 3 Jan 2012, Jerry Geis wrote:

So if its not permissions (since its running as root) what else might it 
be?


(Assuming /home/silentm/bin/ is not an NFS mount that root can't access as 
root.)


PATH or other environment variables?

Does

sudo env --ignore /home/silentm/bin/usbserial -start

yield any clues?

--
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] Question on system command 1.4.43

[Jerry Geis]

On 01/03/2012 08:48 AM, Jerry Geis wrote:
I have a USB to serial converter attached to my box.  pl2303: Prolific 
PL2303 USB to serial adaptor driver
if I login to the box and send/receive serial commands over this unit 
it works without error EVERY time.


however, if I run the same command set from with-in the 
extensions.conf with System()
I get errors in dmesg like "pl2303 ttyUSB0: pl2303_open - failed 
submitting read urb, error -22"

and then obviusly my command does not work.

however, again - If I go back to the command line and my command it 
runs just fine.


Any ideas on what might be happening here?

Jerry

extensions.conf
exten => s,1,ChanIsAvail(Console/Dsp)
exten => s,n,GotoIf($["${AVAILCHAN}" = ""]?smvoice-busy,s,1)
exten => s,n,System(/home/silentm/bin/usbserial -start)
exten => s,n,Playback(beep)
exten => s,n,Dial(Console/dsp)
exten => s,n,Hangup
exten => h,1,System(/home/silentm/bin/usbserial -stop)



Jim and Doug

Yes the process is running as root. I should have mentioned that.
Also - yes the usbserial is a paging notification program.

So if its not permissions (since its running as root) what else might it be?

Thanks,

Jerry

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Re: [asterisk-users] Question on system command 1.4.43

[Doug Lytle]

Jerry Geis wrote:
Any ideas on what might be happening here? 


If Asterisk isn't running as root, my guess would be a permissions issue.

By the way, out of curiosity, what does "usbserial -start" do?   I'm 
guessing, some type of paging busy notification?


Doug


--

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Re: [asterisk-users] Question on system command 1.4.43

[Jim DeVito]

Does the user Asterisk is running as have access to this device?

Thanks!!



- Original message -
> I have a USB to serial converter attached to my box.   pl2303: Prolific 
> PL2303 USB to serial adaptor driver
> if I login to the box and send/receive serial commands over this unit it 
> works without error EVERY time.
> 
> however, if I run the same command set from with-in the extensions.conf 
> with System()
> I get errors in dmesg like "pl2303 ttyUSB0: pl2303_open - failed 
> submitting read urb, error -22"
> and then obviusly my command does not work.
> 
> however, again - If I go back to the command line and my command it runs 
> just fine.
> 
> Any ideas on what might be happening here?
> 
> Jerry
> 
> extensions.conf
> exten => s,1,ChanIsAvail(Console/Dsp)
> exten => s,n,GotoIf($["${AVAILCHAN}" = ""]?smvoice-busy,s,1)
> exten => s,n,System(/home/silentm/bin/usbserial -start)
> exten => s,n,Playback(beep)
> exten => s,n,Dial(Console/dsp)
> exten => s,n,Hangup
> exten => h,1,System(/home/silentm/bin/usbserial -stop)
> 
> 
> 
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Re: [asterisk-users] Question on hung channel

[Jerry Geis]

On 12/28/2011 03:57 PM, Jerry Geis wrote:

I ran into a "rare" situation today.

A really short message is being played over the ALSA or console 
channel from one asterisk box to another. Both running 1.4.30.
the incoming context on the ALSA or Console port box first runs an AGI 
before connecting the audio path.

The AGI got hung up for an non-asterisk issue. then finally continued.
The audio from the server was DONE, the server already hung up the call.
So the console port is just setting there with nothing to do.
the next call in says  - opps the port is busy so just give busy signal.

The original call never "frees" up and goes away.

How do I get the orphaned call to just close and go away?

Thanks,

Jerry

I have updated to 1.4.43 and I have the same situation.

Any thoughts?

Jerry

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Re: [asterisk-users] Question on PAP2 linksys showing off-hook

[giovanni.v]

Il 30/11/2011 14.38, Jerry Geis ha scritto:

I configured the PAP2 device with asterisk. I have the registration,
thought I was good to go.
Plugged in my Valcom 2924 public address analog connection, called the
extension
and I got busy...  very strange I thought


Not so strange ;-)

According to the Valcom 2924 manual it is designed to connect to an 
analog phone or a pbx analog c.o. port.
So you're connecting the PAP2 FXS port to another pbx port... the one on 
Valcom, not only can't work but may damage one or both devices.


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Re: [asterisk-users] Question about Read() application

[Kingsley Tart]

Danny,

Did you try my local channel idea?

Cheers,
Kingsley.

On Mon, 2011-11-21 at 08:25 -0600, Danny Nicholas wrote:
> I tried to patch app_read on my development dahdi box as follows:
> static int unload_module(void)
> {
> int res;
> 
> res = ast_unregister_application(app);
> 
> /*  ast_module_user_hangup_all(); */
> 
> return res;
> }
> But the offending behavior persists - it's not a "show-stopper" but it
> eventually could be.
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kingsley Tart
> Sent: Saturday, November 19, 2011 4:43 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Question about Read() application
> 
> Hi,
> 
> Did you get a workaround for this? I sent you a message offlist but you
> didn't reply so I don't know whether you saw it.
> 
> Cheers,
> Kingsley.
> 
> On Fri, 2011-11-18 at 13:15 -0600, Danny Nicholas wrote:
> > My IVR wouldn't sound right if I allowed 2 or 3 times before it was
> > considered a failure.   The big(ger) problem is that it just hangs up when
> > it fails, no warning or work around to do.
> > 
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug 
> > Lytle
> > Sent: Friday, November 18, 2011 1:10 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] Question about Read() application
> > 
> > 
> > Danny Nicholas wrote:
> > > The user reported to me that "I punched 1 and it hung up" - in my 
> > > testing,
> > I found that slow DTMF entry (1 digit every 2 seconds or so) or fast 
> > entry (more than 10 digits per second) was most likely to cause the
> problem.
> > 
> > I've never had mine just hangup on a mis-key, but then again I have it 
> > try 3 times before considering it a failure.
> > 
> > exten => s,1,Read(get-admin-password|enter-password|||3|)
> > 
> > Doug
> > 
> > 
> 
> 
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Re: [asterisk-users] Question about Read() application

[Danny Nicholas]

I tried to patch app_read on my development dahdi box as follows:
static int unload_module(void)
{
int res;

res = ast_unregister_application(app);

/*  ast_module_user_hangup_all(); */

return res;
}
But the offending behavior persists - it's not a "show-stopper" but it
eventually could be.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kingsley Tart
Sent: Saturday, November 19, 2011 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about Read() application

Hi,

Did you get a workaround for this? I sent you a message offlist but you
didn't reply so I don't know whether you saw it.

Cheers,
Kingsley.

On Fri, 2011-11-18 at 13:15 -0600, Danny Nicholas wrote:
> My IVR wouldn't sound right if I allowed 2 or 3 times before it was
> considered a failure.   The big(ger) problem is that it just hangs up when
> it fails, no warning or work around to do.
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug 
> Lytle
> Sent: Friday, November 18, 2011 1:10 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Question about Read() application
> 
> 
> Danny Nicholas wrote:
> > The user reported to me that "I punched 1 and it hung up" - in my 
> > testing,
> I found that slow DTMF entry (1 digit every 2 seconds or so) or fast 
> entry (more than 10 digits per second) was most likely to cause the
problem.
> 
> I've never had mine just hangup on a mis-key, but then again I have it 
> try 3 times before considering it a failure.
> 
> exten => s,1,Read(get-admin-password|enter-password|||3|)
> 
> Doug
> 
> 


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Re: [asterisk-users] Question about Read() application

[Kingsley Tart]

Hi,

Did you get a workaround for this? I sent you a message offlist but you
didn't reply so I don't know whether you saw it.

Cheers,
Kingsley.

On Fri, 2011-11-18 at 13:15 -0600, Danny Nicholas wrote:
> My IVR wouldn't sound right if I allowed 2 or 3 times before it was
> considered a failure.   The big(ger) problem is that it just hangs up when
> it fails, no warning or work around to do.
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
> Sent: Friday, November 18, 2011 1:10 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Question about Read() application
> 
> 
> Danny Nicholas wrote:
> > The user reported to me that "I punched 1 and it hung up" - in my testing,
> I found that slow DTMF entry (1 digit every 2 seconds or so) or fast entry
> (more than 10 digits per second) was most likely to cause the problem.
> 
> I've never had mine just hangup on a mis-key, but then again I have it try 3
> times before considering it a failure.
> 
> exten => s,1,Read(get-admin-password|enter-password|||3|)
> 
> Doug
> 
> 


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Re: [asterisk-users] Question about Read() application

[Danny Nicholas]

My IVR wouldn't sound right if I allowed 2 or 3 times before it was
considered a failure.   The big(ger) problem is that it just hangs up when
it fails, no warning or work around to do.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Friday, November 18, 2011 1:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about Read() application


Danny Nicholas wrote:
> The user reported to me that "I punched 1 and it hung up" - in my testing,
I found that slow DTMF entry (1 digit every 2 seconds or so) or fast entry
(more than 10 digits per second) was most likely to cause the problem.

I've never had mine just hangup on a mis-key, but then again I have it try 3
times before considering it a failure.

exten => s,1,Read(get-admin-password|enter-password|||3|)

Doug


-- 

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] Question about Read() application

[Doug Lytle]

Danny Nicholas wrote:

The user reported to me that "I punched 1 and it hung up" - in my testing, I 
found that slow DTMF entry (1 digit every 2 seconds or so) or fast entry (more than 10 
digits per second) was most likely to cause the problem.


I've never had mine just hangup on a mis-key, but then again I have it 
try 3 times before considering it a failure.


exten => s,1,Read(get-admin-password|enter-password|||3|)

Doug


--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


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