Re: [asterisk-users] PJSIP signaling question

On Tue, Mar 1, 2016 at 5:37 PM, Kevin Long wrote: > > > Interesting, thanks George. I pulled Asterisk 13 from git and the new > pjproject from the SVN and will test accordingly . > ​Yeah, actually you do need Asterisk 13 from git because pjproject deprecated an api

Re: [asterisk-users] PJSIP signaling question

Interesting, thanks George. I pulled Asterisk 13 from git and the new pjproject from the SVN and will test accordingly . I have a few more questions about PJSIP in Asterisk 13: 1. Is there any way to list current ongoing calls and see what codecs are being used in the RTP streams? With

[asterisk-users] DAHDI-Linux and DAHDI-Tools 2.11.1 Now Available

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.11.1 DAHDI-Tools-v2.11.1 dahdi-linux-complete-2.11.1+2.11.1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux

[asterisk-users] DTMF issues between Asterisk and Callmanager with Zoiper

I had an old Asterisk installation die recently and we decided to upgrade to Asterisk 13 to replace the old server. Everything seems to be working with PJSIP but there is one issue. Asterisk talks to a callmanager via a SIP trunk and send calls to PSTN (another country). Most of the

Re: [asterisk-users] Can't send 10 type frames with PJSIP

For right now, you could replace line 712 in channels/chan_pjsip.c with the following and recompile. ast_log(LOG_WARNING, "Can't send %u type frames with PJSIP. Endpoint: %s\n", frame->frametype, ast_sorcery_object_get_id(channel->session->endpoint)); You could also run Wireshark (or capture

[asterisk-users] Can't send 10 type frames with PJSIP

I am getting flooded with these messages: [Mar 1 12:25:29] WARNING[6962][C-005a]: chan_pjsip.c:712 chan_pjsip_write: Can't send 10 type frames with PJSIP [Mar 1 12:25:30] WARNING[6962][C-005a]: chan_pjsip.c:712 chan_pjsip_write: Can't send 10 type frames with PJSIP [Mar 1

[asterisk-users] Abandoned SIP-TCP connection causes Asterisk to crash

Hi All, We are using SIP over TCP transport but often we got an Asterisk crash with following error. [Mar 1 11:23:13] WARNING[1509]: chan_sip.c:3755 __sip_xmit: sip_xmit of 0x7f294000cac0 (len 680) to Soft.Phone.IP.Address:56780 returned -2: Interrupted system call Asterisk uncleanly