trixter http://www.0xdecafbad.com
2005-Apr-13 22:56 UTC
[Asterisk-Users] RTP not being sent by asterisk
I am having an odd problem that started somepoint in the last couple days with no known config change. Asterisk will receive RTP data but will not send it. If someone calls my asterisk box, it will hang on any Playback() or Background() call. No data is ever sent on the RTP stream, verified with a packet sniffer. I disabled all bandwidth shaping and firewall settings while testing which had no effect on resolving this. SIP traffic goes back and forth, and a sip debug shows everything being set up. I have deinstalled and reinstalled what was previously working. A friend who has the same version installed from the same place has no problems with his setup. I started with asterisk from debian testing however built from CVS a few minutes ago and have exactly the same problem. I am now stuck on where to look next to find the problem and need to get my asterisk system working again quickly. Any ideas would be greatly appreciated. Sample I called from extension.conf exten => 123,1,answer exten => 123,2,wait,2 exten => 123,3,playback(beep) ; it hangs on this beep exten => 123,4,playback(beep) exten => 123,5,playback(beep) exten => 123,6,hangup sip.conf was not changed at all, and that works for in/out. The only problem I have is people dialing into my asterisk box, the applciations run, DTMF is read, callers just get absolutly no prompts. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 881 8487 FreeWorldDialup: 635378 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050414/3a9de5bc/attachment.pgp
Are the calls coming from SIP or PSTN? ----- Original Message ----- From: "trixter http://www.0xdecafbad.com" <trixter@0xdecafbad.com> To: <asterisk-users@lists.digium.com> Sent: Thursday, April 14, 2005 3:56 PM Subject: [Asterisk-Users] RTP not being sent by asterisk> _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-user
Does maximum debugging show anything? ----- Original Message ----- From: "trixter http://www.0xdecafbad.com" <trixter@0xdecafbad.com> To: <asterisk-users@lists.digium.com> Sent: Thursday, April 14, 2005 3:56 PM Subject: [Asterisk-Users] RTP not being sent by asterisk> _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-user
trixter http://www.0xdecafbad.com
2005-Apr-13 23:19 UTC
[Asterisk-Users] RTP not being sent by asterisk
On Thu, 2005-04-14 at 16:15 +1000, Rod Bacon wrote:> Are the calls coming from SIP or PSTN?from sip, and I can see packets going from sip -> asterisk just nothing outside of sip going from asterisk -> sip phone. Its like there is a blocking issue, although I dont know why this would have happened. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 881 8487 FreeWorldDialup: 635378 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050414/d5076727/attachment.pgp
trixter http://www.0xdecafbad.com
2005-Apr-13 23:20 UTC
[Asterisk-Users] RTP not being sent by asterisk
On Thu, 2005-04-14 at 16:15 +1000, Rod Bacon wrote:> Are the calls coming from SIP or PSTN? >further investigation shows the first RTP packet is sent and nothing after that. btw everything is on localnet and the IPs are all correct. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 881 8487 FreeWorldDialup: 635378 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050414/ebb706d2/attachment.pgp
Can you capture Ethernet traffic with ethereal or similar tools and show what is happening? Alex -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of trixter http://www.0xdecafbad.com Sent: Thursday, April 14, 2005 1:56 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RTP not being sent by asterisk I am having an odd problem that started somepoint in the last couple days with no known config change. Asterisk will receive RTP data but will not send it. If someone calls my asterisk box, it will hang on any Playback() or Background() call. No data is ever sent on the RTP stream, verified with a packet sniffer. I disabled all bandwidth shaping and firewall settings while testing which had no effect on resolving this. SIP traffic goes back and forth, and a sip debug shows everything being set up. I have deinstalled and reinstalled what was previously working. A friend who has the same version installed from the same place has no problems with his setup. I started with asterisk from debian testing however built from CVS a few minutes ago and have exactly the same problem. I am now stuck on where to look next to find the problem and need to get my asterisk system working again quickly. Any ideas would be greatly appreciated. Sample I called from extension.conf exten => 123,1,answer exten => 123,2,wait,2 exten => 123,3,playback(beep) ; it hangs on this beep exten => 123,4,playback(beep) exten => 123,5,playback(beep) exten => 123,6,hangup sip.conf was not changed at all, and that works for in/out. The only problem I have is people dialing into my asterisk box, the applciations run, DTMF is read, callers just get absolutly no prompts. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 881 8487 FreeWorldDialup: 635378