similar to: rtp.c:487 ast_rtp_read: Unknown RTP codec 72 received

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, 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]:

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

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

Mark, While these samples are pretty good they do not work "out of the box" - there are a couple of issues: 1. the samples are 44100 samples/second and Asterisk needs them to be at 8000 samples/second. This is what happens if you prune out all of the Amercian voicemail prompts and substitute yours: Asterisk 1.2.7, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark

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, after cvs upgrading my * installation yesterday, the prompts in both VoiceMail2 and VoiceMailMain2 have become silent. All I get is the initial "blip" followed by the Voice taking breath and being cut off before she has a chance to say "Comedian Mail". All other prompts (ie the Playback application) seem to work fine. I can still login to VoiceMailMain2, however, each

Hi folks, I'm running Asterisk 11 (at the moment - planning to u/grade to v13.7 LTS), I've just configured the voicemail function, and it's mostly working fine... except when I try to leave a voicemail! This crashes asterisk with no entries in the messages log. The system is running on Centos 6 (or maybe 6.5, I'm not sure how to check this). uname -a returns: Linux

Tom, My current example is a global address; however, it could be any operand in theory. The arch allows for direct mem op support for ex instructions, so it could be any type of address or any type of imm or any type of register. For example, we are using intrinsics for some instructions since LLVM does not support them. Table gen does not allow for matching to direct mem op because the

I was just on 70minute call (IAX2 -> Internet -> IAX2) and during that time I heard several "pops", or "clicks". Each time it happened, I saw the following message: Aug 19 15:36:36 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Any ideas what causes these, and why they turn in to a "pop", instead of just silence, or a

Hi folks, I am having a small problem with asterisk-1.6.2.7 + app_fax on OpenBSD 4.7 -release. Everything seems to work fine. I have a macro which answers, receives the fax to a tiff, and then runs a script (mailfax) to convert that to pdf and email it. It all works perfectly except for some errors I am seeing in the console. After it hangs up I get a dozen or so messages in the cli

Hello List. We are having some problems using t.38 together with a Cisco voice router at one of our providers end. We are using the new digium asterisk fax module to generate the fax, and when we use together with our internal Audiocodes Mediant 2000 gateways, we have no issues what so ever, and the faxes go right through. When we send faxes to our other provider, who has cisco hardware

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

Hi list,* *I am not able to make video calls between two sip accounts. below is the information. please help me where I am missing the configuration.* Extensions.conf* exten => 111,1,Answer() same => n,Dial(SIP/2206,60,r) same => n,Hangup() *SIP.conf* [2218] type=friend secret=******* callerid="Virendra" <9172341457> host=dynamic ;

Hi All, I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router with a PRI card in it, handing off to a PBX and vise verse. Calls in and out are working fine except for DTMF from Asterisk to the 2600. DTMF from the 2600 to Asterisk is fine. Here are the Asterisk console warnings I get when I send DTMF from Asterisk to the 2600: == Forcing Marker bit, because SSRC has changed Jun

Hey Ryan, You end with a large constant immediate offset value because the register operand stores the register id in a union together with the offset that's used by the global address operand. Just add 'setOffset(0)' to your change method and that should solve your problem. 2015-06-16 9:15 GMT-07:00 Ryan Taylor <ryta1203 at gmail.com>: > So I have this for

On 6/26/07, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router > with a PRI card in it, handing off to a PBX and vise verse. Calls in > and out are working fine except for DTMF from Asterisk to the 2600. > DTMF from the 2600 to Asterisk is fine. > > Here are the Asterisk console warnings

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

I'm playing with codecs at the moment and have found some notices errors when x-lite/pro connects to asterisk with Speex or ILBC. Initially I was getting garbled sound, but after changing magic number for both codecs to 97 (as per http://www.voip-info.org/wiki-Asterisk%20phone%20xten%20xlite and http://bugs.digium.com/bug_view_page.php?bug_id=0000918) I was able to get normal voice. BUT,