search for: savp

Hello, I have a problem with a grandstream IP Phone. The SIP autentication is OK, but when try to call someone I get the message --> WARNING[14281] chan_sip.c: Error in codec string '=audio 5004 RTP/SAVP 3' I tried to change the CODECs (ulaw, alaw, GSM, etc), the result is always the same. Tried to change the RTP port but the result is the same. The grandstream IPhone is behind a NAT, However there are others IP Phones working fine behind NAT (from other locations, outside my network) Change...

...ple - would be very helpful in such cases: Description of "488 Not acceptable here" - Could be caused by codec problems, when codec negotiation failed. You can check if the negotiation failed by .... - Can be paused by a phone offering encryption, but only offering RTP/AVP instead of RTP/SAVP profile. Check if the sip log contains a crypto line and only RTP/AVP, if yes, change the phone settings from RTP/AVP to RTP/SAVP or disable RTP encryption in the phone's settings. [Even better: Besides throwing the error message also add the reason for it, at least in the Asterisk log files. I...

...and rejects the call with '488 Not acceptable here' Doesn't Asterisk support UNENCRYPTED_SRTCP as crypto parameters in sdp? FYI SDP looks like this. v=0 o=- 1429194215 1 IN IP4 XX.XX.XX.XX s=- c=IN IP4 XX.XX.XX.XX b=TIAS:64000 t=0 0 a=avf:avc=n prio=n a=csup:avf-v0 m=audio 50096 RTP/SAVP 0 18 120 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:120 telephone-event/8000 a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:zUVSWsFB/WjVtLxXojBT7zbNvuQ4BkOwcCkD/AjM|2^20 UNENCRYPTED_SRTCP And on CLI I see, DEBUG[1568][C-00000000] sip/sdp_crypto.c: local_key64 7v...

...t with "Rejecting secure video stream without encryption details". - sipML5, but it won't register, perhaps something to do with not using the Asterisk Websocket server (which I don't see an option to choose) - Janus, but the INVITE SDP contains "RTP/AVP" not "RTP/SAVP, and Asterisk rejects it with "We are requesting SRTP for audio, but they responded without it!" Thanks for any suggestions. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -------------- next part ------...

Hello, a person trying to call me by my phone number is getting the error 488 Not acceptable here. I googled that error, seems like this error is normally caused by a failed codec negotation, though I have no clue how I could have read this out of the logs. Anyway, my setup is as follows: Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider The user calling me is also using Sipgate and is calling my

...nts are successfully connected. But when I made a call from one of the users, the other user doen'st have call notification and for a while the calling process ended. I check the /var/log/asterisk/messages got the following log: [Dec 17 14:54:11] WARNING[11471][C-00000000] chan_sip.c: Received SAVPF profle in audio offer but AVPF is not enabled: audio 52760 RTP/SAVPF 103 104 0 8 107 106 105 13 126 [Dec 17 14:54:11] WARNING[11471][C-00000000] chan_sip.c: Received SAVPF profle in video offer but AVPF is not enabled: video 52760 RTP/SAVPF 100 101 102 [Dec 17 14:54:11] WARNING[11471][C-00000000]...

...er Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 522 v=0 o=testacc77005 8004 8000 IN IP4 192.168.1.104 s=SIP Call c=IN IP4 192.168.1.104 t=0 0 m=audio 5020 RTP/SAVP 0 8 18 9 2 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:8m7ZfG+0t3KBFGK40IfDO11SZ6D54glKKIwdgo00...

Hi, I am running asterisk 1.8.5.0 and have compiled in the srtp module All but Snom phones are working. I have set the srtp tag on the snoms to 80 and RTP/SAVP to mandatory and they worked for a few hours. This morning all snoms are reporting this when trying to make a call (this is snom calling snom). ---------snip------------------ == Using SIP RTP CoS mark 5 -- Executing [10000 at default-outbound08:1] Dial("SIP/10002-00000012", "S...

...CK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-Vox-Evo MP-118/v.5.60A.024.003 Content-Type: application/sdp Content-Disposition: session Content-Length: 528 v=0 o=AudiocodesGW 781713142 781713021 IN IP4 192.168.105.199 s=Phone-Call c=IN IP4 192.168.105.199 t=0 0 m=audio 6000 RTP/SAVP 0 8 18 4 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:EFG/GFJBnNMdfJ2/hBCyJmgdPS6MNkuOscQEJR3E|2^31 a=cry...

This series implements ballooning for Xen on ARM and builds and Mukesh''s PVH privcmd stuff to implement foreign page mapping on ARM, replacing the old "HACK: initial (very hacky) XENMAPSPACE_gmfn_foreign" patch. The baseline is a bit complex, it is basically Stefano''s xenarm-forlinus branch (commit bbd6eb29214e) merged with Konrad''s linux-next-pvh branch

...EGISTER, REFER, MESSAGE Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: PBXe 1.4.0 Content-Type: application/sdp Content-Length: 342 v=0 o=- 772596305 772596305 IN IP4 192.168.1.4 s=Asterisk c=IN IP4 192.168.1.4 t=0 0 m=audio 14476 RTP/SAVP 0 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Ojz7o69EOsPsdsRTgNO/wtRWPsrWc2NSnOidNcqh a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv both phones SPA502, force_rport disabled for tls phone, here is my transports...

...e > video stream without encryption details". > > - sipML5, but it won't register, perhaps something to do with not using > the Asterisk Websocket server (which I don't see an option to choose) > > - Janus, but the INVITE SDP contains "RTP/AVP" not "RTP/SAVP, and Asterisk > rejects it with "We are requesting SRTP for audio, but they responded > without it!" > > Thanks for any suggestions. > > -- > David Cunningham, Voisonics > http://voisonics.com/ > USA: +1 213 221 1092 > UK: +44 (0) 20 3298 1642 > Australia:...

...ork with spandsp-0.0.6 - Closes issue #13688. Reported by and patched by irroot. * chan_h323 with H323Plus for TRUNK (SVN rev. 89183) - Closes issue #11261. Reported by vhatz. Patched by jthurman. * Wrong usage of sscanf with use of uninitialized variable caused accidental parsing of RTP/SAVP - Closes issue #14000. Reported and patched by folke. * Realtime peers are never qualified after 'sip reload' - Closes issue #14196. Reported, tested, and patched by pdf. Thank you for your continued support of Asterisk!

...ork with spandsp-0.0.6 - Closes issue #13688. Reported by and patched by irroot. * chan_h323 with H323Plus for TRUNK (SVN rev. 89183) - Closes issue #11261. Reported by vhatz. Patched by jthurman. * Wrong usage of sscanf with use of uninitialized variable caused accidental parsing of RTP/SAVP - Closes issue #14000. Reported and patched by folke. * Realtime peers are never qualified after 'sip reload' - Closes issue #14196. Reported, tested, and patched by pdf. Thank you for your continued support of Asterisk!

I'm in the process of testing a TLS/SRTP install. My experience is improving with each new challenge, but this one is a great test of my 2 month experience with Asterisk. When I dial 6003 from 6001, it takes 35 seconds until I get the error message that 6003 is circuit-busy. Any help would greatly be appreciated. Below is the error message and the extensions and sip.conf files. *CLI>