Re: [asterisk-users] No media being sent in SIP call
Hi! I've turned off t.38 and all of the codecs except ulaw; I still have the same problems. SOMETIMES it works. Other times, the sniffer clearly shows that the media simply isn't being sent. NOTHING is being sent. Anything else I should check? Look at the firewalls and possible SIP ALGs that are between the devices. Check for UDP port forwarding settings, and check that the RTP ports that have been negotiated for the call are not conflicting with those of other devices/calls/port forwarding settings. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No media being sent in SIP call
There are NO ACL's in place, either at the network level, or application level. We have a public address, so as far as I know, there are no forwarding rules in place. On Wednesday 27 October 2010 4:04:16 pm Philipp von Klitzing wrote: Hi! I've turned off t.38 and all of the codecs except ulaw; I still have the same problems. SOMETIMES it works. Other times, the sniffer clearly shows that the media simply isn't being sent. NOTHING is being sent. Anything else I should check? Look at the firewalls and possible SIP ALGs that are between the devices. Check for UDP port forwarding settings, and check that the RTP ports that have been negotiated for the call are not conflicting with those of other devices/calls/port forwarding settings. Philipp -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No media being sent in SIP call
Do you have canreinvite=yes anywhere? If yes, try setting it to no. Also pasting your sip.conf here would be helpful. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-27 6:16 PM, Mike Diehl mdi...@diehlnet.com wrote: There are NO ACL's in place, either at the network level, or application level. We have a public address, so as far as I know, there are no forwarding rules in place. On Wednesday 27 October 2010 4:04:16 pm Philipp von Klitzing wrote: Hi! I've turned off t Take care and have fun, Mike Diehl. -- ___... -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No media being sent in SIP call
Hi all, I seem to be having a strange problem with a sip trunk. On a fairly frequent basis, I'll make a call, ore receive a call, and there will be NO sound. The strange part is that both endpoints are public IP addresses so NAT isn't in play and a sniffer trace reveals that the packets simply aren't being sent. It only seems to happen on a particular trunk. The same phone calling on a different trunk works just fine. Any ideas? -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No media being sent in SIP call
2010/10/26 Mike Diehl mdi...@diehlnet.com Hi all, I seem to be having a strange problem with a sip trunk. On a fairly frequent basis, I'll make a call, ore receive a call, and there will be NO sound. The strange part is that both endpoints are public IP addresses so NAT isn't in play and a sniffer trace reveals that the packets simply aren't being sent. It only seems to happen on a particular trunk. The same phone calling on a different trunk works just fine. Any ideas? codec incompatibilities ? t.38 ? -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
