similar to: "Received packet with bad UDP checksum" - whats the real problem?

Displaying 20 results from an estimated 10000 matches similar to: ""Received packet with bad UDP checksum" - whats the real problem?"

2007 Apr 23
1
app_rxfax produces "RTP: Received packet with bad UDP checksum"
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",
2004 Aug 19
1
Received packet with bad UDP checksum
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
2005 May 20
2
How to get in touch with sixTel?
If anybody here is a sixTel customer, can you share any tips & tricks for getting in touch with anybody there? They are absurdly hard to get a hold of, particularly when you have a technical issue needing to be resolved. If anyone has any phone numbers other than their main 800 line, I'd sure appreciate it. Thank you, Bryan -------------- next part -------------- An HTML attachment was
2005 Jun 01
1
RFC2833 & firewall problems? (16-byte UDP packets)
We are tracking the following situation: SIP client connects to our Asterisk server, and then connects to another SIP user. Re-invite is OFF, so Asterisk is in the middle of the whole conversation. When one SIP client sends DTMF tones, the SIP client uses RFC2833 to send the tones to the server. (This is correct). The server then sends RFC2833 tones out to the other SIP client. The problem is,
2006 Jun 19
1
Asterisk 1.2.9 cli "-x" doesn't flush?
We have a script which executes "asterisk -n -r -x ....." periodically against the running server, to check the status of a few things, and pipe the output to a file. With prior versions of Asterisk this worked fine, but having just upgraded to 1.2.9, we are finding that if the output is lengthy, then Asterisk seems to terminate before fully flushing stdout. Is this a known bug, is
2003 Jul 31
1
RTP codec 13 received - Cisco incompatibility?
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
2003 Jul 31
1
RTP codec 13 received - Cisco incompatibilit y?
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
2012 Mar 16
0
How to know if packet is dropped by kernel for udp checksum mismatch
Hi all, I have a netfilter_queue app which de-obfuscates a already obfuscated udp packets. de-obfuscation process ends successfully but somehow packet is not reaching to the udp daemon. i'm suspecting this is happening because of a udp checksum mismatch. so i'm wondering how can i get logs of packets which are dropped because of udp checksum mismatch? i've heard linux by default
2013 Apr 17
3
Attempting to checksum a non-TCP/UDP packet, dropping a protocol 1 packet
Pasi, in http://readlist.com/lists/lists.xensource.com/xen-users/10/50495.html you mention a netback fix without identifying which one. In going over the changes as well as looking at the code, I can''t spot anything related, or see how non-TCP, non-UDP packets could pass checksum_setup(), and hence I don''t really understand how this problem can be considered fixed (for, in the
2008 Feb 29
1
Received UDP packet from unknown source 1.2.3.4 (port 12345)
Hi list, I have a VPN mesh with ~10 nodes. A recently added node experience the 'Received UDP packet from unknown source' problem. I read in the faq this is probably caused by a NAT rule on wither side, but I dont have such rules. The thing is that IP in the 'Received UDP packet from unknown source ' message is exactly what I have configured. The problem solves itself with
2003 Apr 17
1
Unknown RTP codec 101 received
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
2006 Feb 25
2
Unknown RTP codec 100 received
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
2010 May 28
2
Asterisk 1.6.2.7 + app_fax + OpenBSD 4.7 minor issue
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
2011 Nov 21
1
video calls not working
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 ;
2007 Jun 26
1
Asterisk to Cisco 2600 GW DTMF Not Working
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
2005 Jul 11
4
Video phone settings???
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
2011 Apr 16
5
Google Voice receiving call problem
Hello, I have a Google Voice phone number and want to connect it to my asterisk box to have calls handled to my SIP account. When I call the number I receive the correct INCOMING request on Jabber portion of asterisk, but the call is not connected to the gtalk part. JABBER: asterisk INCOMING: <iq from="+ 17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" to="
2004 Nov 23
1
Fax over SIP Problems (sorry for this topic ...)
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
2003 Dec 16
2
Unable to Receive Fax -- RxFAX Application
Hi, Below if the error message which I got from asterisk. I was trying to fax to asterisk from my fax machine. I really dunno what is the problem. I use alaw & ulaw codec only through my ATA 186. Can anyone help me what could be the problem. -- Executing Goto("SIP/-080ef9a0", "13732|s|1") in new stack -- Goto (13732,s,1) -- Executing
2009 Jul 16
3
T38 negotiation, the last step !
Hi, I've managed to get HYLAFAX---->T38MODEM----->ASTERISK---->CISCOAS5400 working, but when they are negotiating asterisk drops a message telling "Unknown RTP codec 96 received from gateway" Do somebody know how to fix it ? Thank you ! << [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/GWCISCO5400O-600bfcc8] << [ TYPE: Control (4) SUBCLASS: Answer (4) ]