Ok remove "from CallCentric" from the subject. Does that change things? This happens on a few of my test DIDs. The days upon days of exhausted searching trying to find some answer to this issue has not been helpful but I do see comments in some places regarding the very specific 32 second timeout but those results again have not helped solve my issue. Ive also found some comments from people here and there regarding CallCentric so thought someone here may have come across the issue and resolved it somehow. Of course I have contacted them, they haven't a clue. Is that surprising? On Mar 13, 2013, at 9:31 PM, Trent Creekmore <tcreek at gmail.com> wrote: > Shouldn?t You be calling them first? > > From: pjsip [mailto:pjsip-bounces@xxxxxxxxxxxxxxx] On Behalf Of Ted Gerold > Sent: Wednesday, March 13, 2013 9:33 PM > To: PJSip > Subject: Calls from CallCentric disconnect after 32 seconds > > Hi, > When using CallCentric all incoming calls to my PJSUA app disconnect after 32 seconds. This does not happen with Vitelity. The only thing I can see is not getting ACK from CallCentric and some retransmission timer destroying the call but I'm kind of new so. Any thoughts would be helpful. > > Here is some log from a CallCentric Call > > 14:56:08.568 strm0x1c333ac !Start talksprut.. > 14:56:08.898 strm0x1c333ac !RTP status: badpt=0, badssrc=0, dup=0, outorder=-1, probation=-1, restart=0 > 14:56:08.936 strm0x1c333ac RTP status: badpt=0, badssrc=0, dup=0, outorder=0, probation=-1, restart=0 > 14:56:08.936 strm0x1c333ac PUT prefetch_cnt=1/0 > 14:56:09.050 tsx0x1c26994 !Retransmit timer event > 14:56:09.050 tsx0x1c26994 .Retransmiting Response msg 200/INVITE/cseq=200 (tdta0x1c2f100), count=0, restart?=1 > 14:56:09.050 pjsua_core.c .TX 1038 bytes Response msg 200/INVITE/cseq=200 (tdta0x1c2f100) to UDP 46.19.209.14:5060: > SIP/2.0 200 OK^M > Via: SIP/2.0/UDP 46.19.209.14;received=46.19.209.14;branch=z9hG4bKd3e3.2f4340539c4cdb92f1df143d9e754d6d.0^M > Via: SIP/2.0/UDP 46.19.209.14:5061;rport=5061;branch=z9hG4bKf96b9b8b6f50d93ea68ae8fb32643403^M > Record-Route: <sip:46.19.209.14;lr;ftag=7c7bdaf9434d14826ba1ccd2cf4a7dd8>^M > Call-ID: 4c14602d268525c70571caa42518da08 at 46.19.209.43-b2b_1^M > From: <sip:0000000000@46.19.209.14>;tag=7c7bdaf9434d14826ba1ccd2cf4a7dd8^M > To: <sip:0000000000 at hidden.com>;tag=jDrOy0asL7vtUMvKTDpTe54q-w1AviGs^M > CSeq: 200 INVITE^M > Contact: <sip:user at 192.168.1.20:5060;ob>^M > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS^M > Supported: replaces, 100rel, timer, norefersub^M > Content-Type: application/sdp^M > Content-Length: 276^M > > > > > Here is some log from a Vitelity call that works fine: > > 14:54:21.458 strm0x11965b4 !Start talksprut.. > 14:54:21.508 sip_endpoint.c !Processing incoming message: Request msg ACK/cseq=102 (rdata0x1181c6c) > 14:54:21.508 pjsua_core.c .RX 425 bytes Request msg ACK/cseq=102 (rdata0x1181c6c) from UDP 66.241.99.28:5060: > ACK sip:user at 192.168.1.20:5060;ob SIP/2.0^M > Via: SIP/2.0/UDP 66.241.99.28:5060;branch=z9hG4bK58e5a5f9;rport^M > From: "+0000000000" <sip:0000000000@66.241.99.28>;tag=as49b47c99^M > To: <sip:0000000000 at 68.2.149.85:5060>;tag=g8V1gfSXWH7RjRHhnsFEyfdDOFvy5J01^M > Contact: <sip:0000000000 at 66.241.99.28>^M > Call-ID: 6d8a0f8a011b8934690dcdf640e96d80 at 66.241.99.28^M > CSeq: 102 ACK^M > User-Agent: packetrino^M > Max-Forwards: 70^M > Content-Length: 0^M > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20130313/eb7afcf8/attachment-0001.html>
