similar to: Received packet with bad UDP checksum

We have a customer trying to dial through our server, and our server is throwing tons of these log messages: Jul 27 14:21:02 NOTICE[29210]: rtp.c:431 ast_rtp_read: RTP: Received packet with bad UDP checksum Is it pretty certain, that these are caused by a bad or misconfigured router along the path, or something else network-related? As opposed to the SIP hardware itself? The SIP ATA is the same

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 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

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

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

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

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

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,

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]:

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 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

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

I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic