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== Using SIP RTP CoS mark 5 - Asterisk Support - Asterisk ...
https://community.asterisk.org/t/using-sip-rtp-cos-mark-5/35333
I am using Grandstream GXP285. Please help me out. I am not getting the solution in any of the forums. Here my sip.conf and asterisk.conf. ===== Sip.conf [incoming] exten => 1000,1,Dial(SIP/1000,20) exten => 1000,n,VoiceMail(1000@wc-voicemail,u) exten => 1000,n,Hangup() exten => 1001,1,Dial(SIP/1001,20) exten => 1001,n,VoiceMail(1001@wc ...
No audio and no rtp traffic - Asterisk SIP - Asterisk ...
https://community.asterisk.org/t/no-audio-and-no-rtp-traffic/69420
== Using SIP RTP CoS mark 5 Found RTP audio format 107 Found RTP audio format 119 Found RTP audio format 100 Found RTP audio format 106 Found RTP audio format 0 Found RTP audio format 105 Found RTP audio format 98 Found RTP audio format 8 Found RTP audio format 101 Found unknown media description format BV32 for ID 107
sip - No sound and RTP failure when playing back file …
https://stackoverflow.com/questions/55711015/no-sound-and-rtp-failure-when-playing-back-file-from-asterisk-server
In asterisk CLI set your verbosity up to 3 or more and start SIP debugging ( you can do this in the config.conf or at the CLI ) I returned the following on the call. == Using SIP RTP CoS mark 5 -- Executing [200@LocalSets:1] Answer ("SIP/WB001-00000000", "") in new stack -- Executing [200@LocalSets:2] Playback ("SIP/WB001-00000000", "hello ...
There was no sound on the call - Asterisk SIP - Asterisk ...
https://community.asterisk.org/t/there-was-no-sound-on-the-call/81940
== Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 [Dec 12 19:37:57] NOTICE[100670][C-00000008]: chan_sip.c:10405 process_sdp: Received SAVPF profle in audio offer but AVPF is not enabled, enabling: audio 61497 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
No RTP from PBX for Outbound call, No Audio - Endpoints ...
https://community.freepbx.org/t/no-rtp-from-pbx-for-outbound-call-no-audio/78858
No RTP from PBX for Outbound call, No Audio. I’m quite new to this FreePBX/Asterisk, trying to set up a SIP call using a softphone on my Laptop. Flow - Softphone -> Internet -> PBX on Azure -> Provider. While making an outbound call, Invite messages (with SDP) are acknowledged by the PBX with 200OK (SDP), the the provider end is sending RTP ...
asterisk - use a different "SIP CoS mark" - Server Fault
https://serverfault.com/questions/566652/asterisk-use-a-different-sip-cos-mark
There are four parameters to control 802.1p CoS: cos_sip, cos_audio, cos_video and cos_text. The behavior of these parameters is the same as for the SIP TOS settings described above. By default, the CoS is already set to 5 for audio traffic. Signalling is set to 3. In all honesty, those values are pretty standard.
No Audio - Issabel
https://forum.issabel.org/d/810-no-audio/7
No Audio. Nareach. I am using Elastix 2.4 and i use trunk without registration. Trunk Peer: disallow=all ... == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Begin MixMonitor Recording SIP/201-00000007-- Called SIP/Smart/0963004214
SIP RTP TOS bits 184 in TCLASS field - Asterisk Support ...
https://community.asterisk.org/t/sip-rtp-tos-bits-184-in-tclass-field/41342
Hello all, I have a production server which I just upgrade from v.1.8 to 11, where the ToS I have it configured as: [quote]tos_sip=cs3 tos_audio=ef cos_sip=3 cos_audio=5[/quote] But in the console I see that the using of the SIP ToS is failing, as: [quote] == Using SIP RTP TOS bits 184 == Using SIP RTP TOS bits 184 in TCLASS field. == Using SIP RTP CoS mark 3[/quote] I’m trying to …
pbx1*CLI> == Using SIP RTP TOS bits 184 == Using SIP RTP ...
https://pastebin.com/BA2FH6nj
text 23.79 KB. raw download clone embed print report. pbx1*CLI>. == Using SIP RTP TOS bits 184. == Using SIP RTP CoS mark 5. -- Executing [12016612006@a2billing:1] Set ("SIP/2952049954-00000006", "CALLERID (num)=2952049954") in new stack.
SRTP Failing - Asterisk Support - Asterisk Community
https://community.asterisk.org/t/srtp-failing/42558
Running Asterisk (Ver. 10.12.1) with Freepbx 2.10.1.9. Receiving the following when trying extension to extension call with SRTP. No TLS involved. [2013-04-23 11:13:07] VERBOSE[10256] netsock2.c: == Using SIP RTP TOS bits 184 [2013-04-23 11:13:07] VERBOSE[10256] netsock2.c: == Using SIP RTP CoS mark 5 [2013-04-23 11:13:07] …
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