similar to: Does Asterisk just pass thru RTP if the codec is the same between two extension?

Hi all, I'd like to know how Asterisk process a RTP data flow. Is there any clue to find out about this? The rtp.c? Thanks. Regards, Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050928/1572210f/attachment.htm

Hi all, I'd like to know how Asterisk process a RTP data flow. Is there any clue to find out about this? The rtp.c? Thanks. Regards, Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050928/beb436a4/attachment.htm

Hello, Trying to use g729 in pass-thru mode. Call flow: SIP IP Phone (G.729)-->Asterisk(1.6.2.9)--->SIP Trunk to ITSP(G.729) When using G.729, call is not getting connected. Below is the extract from CLI. == Using SIP RTP CoS mark 5 -- Executing [12127773456 at default:1] AGI("SIP/100-00000000", "call.php") in new stack -- Launched AGI Script

I want to make sure that RTP is not going thru my asterisk. I read you should avoid in the dial commands: m music while ringing t,T transfer calls from caller and called party What else do I need to take care? remote phone ===> registered to local asterisk ===> calling remote gateway should have the RTP remote phone ===(RTP)==> calling remote gateway bye Ronald

hi We take calls inbound via SIP from our Cisco PSTN gateways, and pass it to customers using IAX (they run their own asterisk servers). We've noticed that asterisk is transcoding the call into a different codec, if the customer prefers a codec different to that which our cisco gw prefers. As such, the quality of the call can degrade. We'd rather asterisk just passed through the RTP

Hello All, Currently my setup uses Xlite and Asterisk and i found that all the RTP voice packets are transfered via the asterisk server from one xlite to another. Is there any possibility that we can make all the RTP Packets to be transfered directly between the two clients once the connection is established?. Any one please help me. Thanks and Regards, Senthil Murugan.V

Hello, In our SIP network, Asterisk is the central PBX, and it routes calls to the PSTN thru a Cisco Router - IOS 12.2(11)T9. If a client softphone calls directly via Cisco to the PSTN, the call works successfully. If the client softphone calls via Asterisk to other SIP internal extension, it work fine too. The problem is when a client calls an Asterisk extension, and Asterisk transfers

I have a similar setup to you and get the same message regularly. I don't think it's the cause of your problem. I did some research on it a while ago: IIRC the cisco uses codec 13 for "silence suppression" whereas asterisk (correctly) uses codec 19. The router can be configured to use 19 also, but I didn't bother. I'm sure somebody will correct me if I'm wrong about

Is it possible that a RTP session between two end users (so i want to use asterisk as a signaling proxy and bypass RTP sessions)? I used "canreinvite=yes" but it didn't work. Description from asterisk conf. File; (canreinvite=yes ; allow RTP voice traffic to bypass Asterisk) Thanks Erdem HAKI - erdemh@tesas.com -------------- next part

After just having update to the latest CVS I am getting the following message when I call VoicemailMain(): -- Executing VoiceMailMain("SIP/2302-1a12", "") in new stack -- Playing 'vm-login' (language 'en') -- Playing 'vm-password' (language 'en') -- Playing 'vm-youhave' (language 'en') -- Playing

Any one see these? Are they benign, or is some system tuning required to remove them? Can't seem to find a resolution in the archives. If you have a link, it would be appreciated. Jun 2 10:58:58 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 19 received Jun 2 10:58:59 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 72 received Jun 2 10:59:00 NOTICE[163044272]:

FYI, the Opus RTP payload format is now RFC7587: https://tools.ietf.org/html/rfc7587 Cheers, Jean-Marc -------- Forwarded Message -------- Subject: [payload] RFC 7587 on RTP Payload Format for the Opus Speech and Audio Codec Date: Tue, 30 Jun 2015 16:33:17 -0700 (PDT) From: rfc-editor at rfc-editor.org To: ietf-announce at ietf.org, rfc-dist at rfc-editor.org CC: drafts-update-ref at

Hi all, I get "Unknown RTP codec 72 received" message in console when call in progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN over voicepulse connect (IAX) and to FWD echo test (SIP). But this message only with one SIP client, others (X-Lite too) not giving this message. All X-Lite settings are identical. Asterisk is last cvs version This what I see in console

Hi all! I am frustrated. I am new to asterisk. My system is ASTLINUX if receive a Fax on my sipura spa2000 i get this: Feb 25 07:41:00 NOTICE[1708]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 received -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060225/ca251876/attachment.htm

could anyone help explaining what does this error mean? i get this error when make a video/ audio call from X-lite to Bria prof. phone rtp.c:1739 ast_rtp_read: Unknown RTP codec 126 received from '10.0.0.26' Gres -------------- next part -------------- An HTML attachment was scrubbed... URL:

I updated to the latest CVS tonight and now DTMF detection does not appear to work on my Cisco 7960 sip phones (can't check voice mail etc). The asterisk console is displaying these messages over and over again any time a DTMF tone is sent: NOTICE[15376]: File rtp.c, Line 292 (ast_rtp_read): Unknown RTP codec 101 received Downgraded to a known working CVS of about three weeks ago, and

I'm testing NVBackgroundDetect with Sipura-300 and I get this error: rtp.c:505 ast_rtp_read: Unknown RTP codec 100 received21 Does anybody know what is it? -- #Joseph

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

hi .. when placing a SIP call to a sip host in the states every few seconds I get an RTP codec 19 error. I know this is related to comfort noise, and the call goes through OK ... how can I suppress the error message ? Also, many times I get "Invalid CSeq Number" back from 216.52.153.207 (which is the host i'm calling) and the call drops.. is there a solution for this ? cheers Dave

hi .. when placing a SIP call to a sip host in the states every few seconds I get an RTP codec 19 error. I know this is related to comfort noise, and the call goes through OK ... how can I suppress the error message ? Also, many times I get "Invalid CSeq Number" back from 216.52.153.207 (which is the host i'm calling) and the call drops.. is there a solution for this ? cheers Dave