You might have to disable srtp negotiations inside the phone web ui options.
Mitul On Nov 12, 2015 8:53 PM, "(lists) Denis BUCHER" <dbuche...@hsolutions.ch> wrote: > Dear all, > > I have a very strange problem : > > - external calls work perfectly, > - internal calls between some phones too, > - but internal call between two similar phones don't work !!! (Snom > 710) > > When we have sound, there are no errors in asterisk. When we do not have > sound, there is the following error : > > - [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP > module loaded, can't setup SRTP session. > > This is a working internal call : > > == Using SIP RTP CoS mark 5 > -- Executing [301@local:1] Dial("SIP/dbucher-00000000", "SIP/phone1") > in new stack > == Using SIP RTP CoS mark 5 > -- Called phone1 > -- SIP/phone1-00000001 is ringing > -- SIP/phone1-00000001 is ringing > -- SIP/phone1-00000001 is ringing > -- SIP/phone1-00000001 is ringing > -- SIP/phone1-00000001 is ringing > -- SIP/phone1-00000001 answered SIP/dbucher-00000000 > -- Remotely bridging SIP/dbucher-00000000 and SIP/phone1-00000001 > Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160) > Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160) > Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160) > Got RTP packet from 192.168.128.99:49646 (type 126, seq 031575, ts > 000001, len 000000) > [Nov 10 17:50:50] NOTICE[21513]: res_rtp_asterisk.c:2190 ast_rtp_read: > Unknown RTP codec 126 received from '192.168.128.99:49646' > Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160) > Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160) > Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160) > Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160) > Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160) > Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160) > Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160) > Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160) > == Spawn extension (local, 301, 1) exited non-zero on > 'SIP/dbucher-00000000' > > This is a non-working call : > > == Using SIP RTP CoS mark 5 > [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP > module loaded, can't setup SRTP session. > -- Executing [301@local:1] Dial("SIP/hsolutionspf5-00000002", > "SIP/phone1") in new stack > == Using SIP RTP CoS mark 5 > -- Called phone1 > -- SIP/phone1-00000003 is ringing > -- SIP/phone1-00000003 is ringing > -- SIP/phone1-00000003 is ringing > -- SIP/phone1-00000003 is ringing > -- SIP/phone1-00000003 is ringing > -- SIP/phone1-00000003 answered SIP/hsolutionspf5-00000002 > -- Remotely bridging SIP/hsolutionspf5-00000002 and SIP/phone1-00000003 > Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160) > Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160) > Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 000033) > Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 000033) > Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 000033) > Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 000033) > Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 000033) > Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 000033) > == Spawn extension (local, 301, 1) exited non-zero on > 'SIP/hsolutionspf5-00000002' > > I tried many options to disable SRTP but without success : > > - canreinvite = no > - canreinvite = nonat > - srtpcapable=no > - encryption=no > - directmedia=nonat > - ...or noload => res_srtp.so in modules.conf > > > Any help would be GREATLY appreciated ! > > Denis > > P. S. We have Asterisk 1.8.4.4 under CentOS release 5.11 (Final) > > > -- > _____________________________________________________________________ > -- 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