Re: [asterisk-users] phpagi packages

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

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

> > 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. >

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

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 > >

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

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

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

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

[asterisk-users] PJSIP equivalent for SIPDtmfMode?

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? -- _ -- Bandwidth and Colocation Provided by

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

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

[asterisk-users] asterisk ari dialer

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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

[asterisk-users] phpagi packages

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

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

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