similar to: ast_rtp_read: Unknown RTP codec 100 received21 when receiving fax

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

Hi, i'd like to know more about this issue, i'm always getting this message while in call with anyone from sip to zap or zap to sip. ast_rtp_read: Unknown RTP codec 72 received here is my current setup: client side, x-lite, with the transmit silence to yes, using ulaw,alaw on asterisk server side: sip.conf contain allow=ulaw and allow=alaw dtmfmode=inband So i always get this anoying

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]:

the above-message keep popping up every second during a conversation between a zap(fxs) channel and sip channel. * eventually hung after a long while we can talk to each other and we can ring one another without any problem. i've had x-lite and x-pro register with * without this problem. furthermore, i have ask my friend to turn off all codec expect g.711MLAW; that did not help i then

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

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

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

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:

While sending calls to a SIP provider, the following warning generates: -- Executing Dial("SIP/1000-c317", "SIP/13057671523@209.120.202.94:5060|55|o") in new stack -- Called 13057671523@209.120.202.94:5060 -- SIP/209.120.202.94:5060-0533 is making progress passing it to SIP/1000-c317 -- SIP/209.120.202.94:5060-0533 answered SIP/1000-c317 -- Attempting

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

Can anybody share information how to install NVBackgroundDetect? I have the app_nv_backgrounddetect.c but I'm missing: app_nv_backgrounddetect.o and app_nv_backgrounddetect.so and have no idea how to generate them. Wiki point to contact Newman Telecom but all I received was the app_nv_backgrounddetect.c and no instruction how how to install it. The installation instruction from wiki

To me it's a error if I can't complete calls using the ATA configured to use the g726 codec. I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I received NOTICES and WARNINGS, but I can't complete a call. On a zap channel: -- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack -- Called 1/2217008 -- Zap/1-1 answered

Hello All, Has anyone ever seen this before. This only happens when i'm on phone call -- Zap/2-1 is ringing -- SIP/2203-c48d is ringing -- SIP/2202-f2ad is ringing -- SIP/2204-11cd is ringing -- SIP/2205-ce62 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- SIP/2205-ce62 answered Zap/1-1 -- Hungup 'Zap/2-1' Jan

When I make a call using sip, I get the line NOTICE[327696]: File rtp.c, Line 292 (ast_rtp_read): Unknown RTP codec 19 received Repeated many times on the console ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to ;bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls allow=gsm allow=ulaw

Hello everyone! I tried to send a fax over SIP with an Asterisk Server in the middle (no Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway is external). Whenever I start sending a Fax to a PSTN destination, the Call gets answered and asterisk tries to build a native bridging: -- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 Then the following

I have tried to set up app_rxfax to receive faxes over IP. I realise there are mixed stories about how reliable this is at the best of times, but at this point all I'm after is some guidance in interpreting the log below. What does "RTP: Received packet with bad UDP checksum" suggest? Here is the full log: -- Executing SetVar("SIP/0892130888-b27c",

Hi, im using a Asterisk Server which is not behind NAT. First i had problems with the fax detection. But this is now solved after adding a wait(2) at the correct place. But i'm still unable to receive a fax due to res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short after the Fax session has started. My sip.conf includes [general] allowguest=no alwaysauthreject=yes sendrpid=rpid

Has anyone sucessfully implimented a fax or modem detection dial plan? I'm originating calls from asterisk using a list of numbers and dropping the destination into an IVR menu but need to do something different if a modem or fax answers. I tried to use the NVBackgroundDetect() application but i think that is for receiving faxes only. Any help would be appreciated. Thanks

Hello, I've been trying to setup a Fax2Email mecanism on my Asterisk box. I have been using the following: 1) An incoming IAX line on Unlimitel (Im not even sure if it's worth mentionning the provider, but I do just in case) 2) NVBackGroundDetect from Newman Telecom 3) The following extension to test: exten => fax,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif) exten