Re: [asterisk-users] phpagi packages

[Dovid Bender]

Tzfarir,

We use it still. What changes were made since
https://github.com/welltime/phpagi/ (which seems to have been last updated
4 years ago)?



On Thu, Jun 29, 2017 at 6:49 AM, Tzafrir Cohen 
wrote:

> Hi all,
>
> We packaged phpagi for Centos 7 and Debian 8 (though nothing
> version-specific in those packages, I suppose).
>
> Packaging:
> http://git.xorcom.com/cgit/rpm/phpagi.git/
>
> Packages:
> * RPM: http://updates.xorcom.com/servers/ombutel/
> * Deb: should soon be in http://updates.xorcom.com/servers/spark/
>
> That said, packaging there has indeed been rather trivial.
>
> The files are taken from https://github.com/welltime/phpagi/ that has
> some 20 so commit commits on top of the version from
> https://phpagi.sourceforge.net/ (sadly that git repository started from
> an import of the files and does not preserve the history of the
> Subversion repository[1]).
>
> There is a separate fork from Sangoma / FreePBX
> (https://github.com/sangoma/phpagi/ ), but it has changed more things
> and thus AGIs written to use the original will fail.
>
> I just wonder how many people do use this, given the staggering rate of
> changes in it.
>
> [1] That includes some 20 or so commits of its own, and begins with an
> import of the files from CVS, ignoring the history in the CVS
> repository.
>
> --
>Tzafrir Cohen
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com
>
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Re: [asterisk-users] PJSIP equivalent for SIPDtmfMode?

[Daniel Tryba]

> > Can't find a way to control the dtmf mode on a per session basis with
> > pjsip, used to use SIPDtmfMode from the dialplan with chan_sip. Any
> > hints on how to do this?
> 
> There is no current way, but a community member has recently posted a
> change[1] for review which implements this.
> 
> [1] https://gerrit.asterisk.org/#/c/5909/

Just what I wished for. I'd love to see this added.

But I see later versions (>13.14.1) have an dtmfmode auto_info option,
might also fit my needs.

Thanks for your feedback to my question(s).


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Re: [asterisk-users] DMTF payload bug in 13.14.1 with pjsip and direct_media?

[Daniel Tryba]

On Thu, Jun 29, 2017 at 11:55:51AM -0500, Richard Mudgett wrote:
> > To me this looks like a bug in asterisk. Either asterisk should use the
> > same rtp payloads for telephone-events on both call legs during inital
> > callsetup or asterisk should come to the conclusion there is an
> > incompatible "codec" on both legs so it shouldn't switch to direct
> > media.
> >
> > Has anyone else seen this issue?
> >
> 
> This is an old issue.  One of the latest issues is:
> 
> https://issues.asterisk.org/jira/browse/ASTERISK-25166

I was looking DTMF related problems and found none. Looks like it is a
more general issue related to all capabilities. Thank you for pointing
this out. 

Seeing the history of the bugs the problem and the full fix is larger
than I initially thought. Maybe a quick stopgap is to just not try to
setup direct media if there are numeric differences between call legs
(this would help me since most call would be direct media, I'll try to
look into this is I have the time to look into this and find out if I
have enough knowlegde to try something).


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Re: [asterisk-users] DMTF payload bug in 13.14.1 with pjsip and direct_media?

[Richard Mudgett]

On Thu, Jun 29, 2017 at 8:32 AM, Daniel Tryba  wrote:

> While trying to use direct_media I'm seeing RTP payload mismatches after
> succesful reinvites.
>
> Initial INVITE from endpoint A to asterisk has rfc4733 DMTF
> m=audio 35648 RTP/AVP 9 8 111 96
> a=rtpmap:96 telephone-event/8000
> a=fmtp:96 0-16
>
> From asterisk to upstream U:
> m=audio 14338 RTP/AVP 9 8 111 18 0 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
>
> So the payload types in the RTP streams from A and to U differ. This
> works fine when asterisk is relaying media.
>
> With direct_media=yes there are reinvites sent from asterisk to both A
> and U. The invite to A contains:
> c=IN IP4 ipaddrofU
> m=audio 33142 RTP/AVP 8 96
> a=rtpmap:96 telephone-event/8000
> a=fmtp:96 0-16
>
> And the invite to U contains:
> c=IN IP4 ipaddrofA
> m=audio 35648 RTP/AVP 9 8 111 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
>
> Both sides respond with a 200 OK and asterisk is not
> relaying/transcoding the media anymore. At this moment DTMF send from A
> isn't getting recognized by U, which IMHO is totally understandable
> since U doesn't know about payload 96.
>
> To me this looks like a bug in asterisk. Either asterisk should use the
> same rtp payloads for telephone-events on both call legs during inital
> callsetup or asterisk should come to the conclusion there is an
> incompatible "codec" on both legs so it shouldn't switch to direct
> media.
>
> Has anyone else seen this issue?
>

This is an old issue.  One of the latest issues is:

https://issues.asterisk.org/jira/browse/ASTERISK-25166

Richard
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Re: [asterisk-users] PJSIP equivalent for SIPDtmfMode?

[Joshua Colp]

On Thu, Jun 29, 2017, at 01:09 PM, Daniel Tryba wrote:
> Can't find a way to control the dtmf mode on a per session basis with
> pjsip, used to use SIPDtmfMode from the dialplan with chan_sip. Any
> hints on how to do this?

There is no current way, but a community member has recently posted a
change[1] for review which implements this.

[1] https://gerrit.asterisk.org/#/c/5909/

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] PJSIP equivalent for SIPDtmfMode?

[Daniel Tryba]

Can't find a way to control the dtmf mode on a per session basis with
pjsip, used to use SIPDtmfMode from the dialplan with chan_sip. Any
hints on how to do this?


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[asterisk-users] DMTF payload bug in 13.14.1 with pjsip and direct_media?

[Daniel Tryba]

While trying to use direct_media I'm seeing RTP payload mismatches after
succesful reinvites.

Initial INVITE from endpoint A to asterisk has rfc4733 DMTF
m=audio 35648 RTP/AVP 9 8 111 96
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16

>From asterisk to upstream U:
m=audio 14338 RTP/AVP 9 8 111 18 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

So the payload types in the RTP streams from A and to U differ. This
works fine when asterisk is relaying media.

With direct_media=yes there are reinvites sent from asterisk to both A
and U. The invite to A contains:
c=IN IP4 ipaddrofU
m=audio 33142 RTP/AVP 8 96
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16

And the invite to U contains:
c=IN IP4 ipaddrofA
m=audio 35648 RTP/AVP 9 8 111 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

Both sides respond with a 200 OK and asterisk is not
relaying/transcoding the media anymore. At this moment DTMF send from A
isn't getting recognized by U, which IMHO is totally understandable
since U doesn't know about payload 96. 

To me this looks like a bug in asterisk. Either asterisk should use the
same rtp payloads for telephone-events on both call legs during inital
callsetup or asterisk should come to the conclusion there is an
incompatible "codec" on both legs so it shouldn't switch to direct
media.

Has anyone else seen this issue?

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[asterisk-users] asterisk ari dialer

[marek cervenka]

hi,

do you have someone example of

http://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/

in node.js asterisk-ari ?

thanks

Marek


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[asterisk-users] phpagi packages

[Tzafrir Cohen]

Hi all,

We packaged phpagi for Centos 7 and Debian 8 (though nothing
version-specific in those packages, I suppose).

Packaging:
http://git.xorcom.com/cgit/rpm/phpagi.git/

Packages:
* RPM: http://updates.xorcom.com/servers/ombutel/
* Deb: should soon be in http://updates.xorcom.com/servers/spark/

That said, packaging there has indeed been rather trivial.

The files are taken from https://github.com/welltime/phpagi/ that has
some 20 so commit commits on top of the version from
https://phpagi.sourceforge.net/ (sadly that git repository started from
an import of the files and does not preserve the history of the
Subversion repository[1]).

There is a separate fork from Sangoma / FreePBX
(https://github.com/sangoma/phpagi/ ), but it has changed more things
and thus AGIs written to use the original will fail.

I just wonder how many people do use this, given the staggering rate of
changes in it.

[1] That includes some 20 or so commits of its own, and begins with an
import of the files from CVS, ignoring the history in the CVS
repository.

-- 
   Tzafrir Cohen
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com

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Re: [asterisk-users] PJSIP list of peers online/offline?

[Floimair Florian]

You can try:

pjsip show endpoints

However, there is no summary line in the end (only the total number of objects) 
so you will have to parse the status of each entry yourself to get these 
statistics.

 
 
With best regards

Florian Floimair

COMMEND INTERNATIONAL GMBH
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Antony Stone
Gesendet: Mittwoch, 28. Juni 2017 12:59
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] PJSIP list of peers online/offline?

Hi.

I have some Nagios / Icinga monitoring plugins I've created for Asterisk, and 
one of them checks the percentage of SIP accounts which are currently 
registered on an Asterisk server.

It does this by running "sip show peers" via AMI and analysing the summary line 
at the end:

1066 sip peers [Monitored: 747 online, 310 offline Unmonitored: 3 online, 6 
offline]

I then calculate 747 divided by (747+310) and report the % online (because I 
know I'm not interested in the unmonitored ones).


However, a customer has upgraded one of their servers from Asterisk 11 to 
Asterisk 13, and "sip show peers" no longer works.


I can see a whole list of commands starting with "pjsip" but there's no "pjsip 
show peers", so what's the new command which will tell me how many online and 
how many offline SIP peers there are?


Thanks in advance,


Antony.

-- 
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