| Summary: | ASTERISK-17618: Local Bridging not working | ||
| Reporter: | jojo (jojo) | Labels: | |
| Date Opened: | 2011-03-29 17:30:11 | Date Closed: | 2012-01-26 09:03:36.000-0600 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Applications/app_dial |
| Versions: | 1.8.3 | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | I seem to be having a very strange problem on 1.8.3.2. Bridging doesn't seem to be working for a call which comes inbound on our DID numbers and is then diverted to an extension, which is further instructed to dial an outbound number. There is no ringtone, and no audio available. However, dialing the extension directly works, as does inbound DID to a SIP phone (or IVR). This setup worked on our previous (1.4) installation. extensions.conf [from-reception] include => ext-internal [ext-internal] exten => 107,1,Dial(SIP/sipcall/00353000000000,60) sip.conf [general] port = 5060 bindaddr = 0.0.0.0 disallow=all allow=g729 allow=ulaw allow=alaw srvlookup=yes callerid = Unknown alwaysauthreject=yes dtmfmode=rfc2833 register => 41610000007:xxxxxx@myvoipprovider.com/107 [sipcall] type=peer defaultuser=41610000000 secret=xxxxxxx context=from-reception host= myvoipprovider.com fromuser=41615111100 qualify=yes fromdomain=myvoipprovider.com insecure=port,invite caninvite=no canreinvite=no nat=no that's literally it. I even stripped this right down, so these are the only config files running. I get the following result: == Using SIP RTP CoS mark 5 -- Executing [107@from-reception:1] Dial("SIP/sipcall-00000023", "SIP/sipcall/00353000000000,60") in new stack == Using SIP RTP CoS mark 5 -- Called sipcall/00353000000000 -- SIP/sipcall-00000024 is making progress passing it to SIP/sipcall-00000023 -- SIP/sipcall-00000024 answered SIP/sipcall-00000023 -- Locally bridging SIP/sipcall-00000023 and SIP/sipcall-00000024 but no audio. Have also tried using another provider on the outbound, no luck. this is pretty serious. We divert our incoming landlines to our mobiles using this method, and nothing is working. | ||
| Comments: | By: jojo (jojo) 2011-03-29 17:33:30 note that all of our other features are working with no issues, i.e. SIP to PSTN, DID to SIP, Skype4Asterisk to PSTN, etc, etc, etc. By: Leif Madsen (lmadsen) 2011-04-05 15:21:22 Have you tried adding Progress() as the first line prior to the Dial()? By: jojo (jojo) 2011-04-06 17:21:27 Yes I have tried it. It doesn't work. By: jojo (jojo) 2011-04-06 18:37:45 well, it seems that local bridging is considered "minor" around here and noone answered on the forums either. fine. have rolled back to 1.6 *without a single change to a config file* and everything works just fine again. given the absolute basic setup we are talking about here, i do think that something is a bit odd about all of this; however, we don't have time to wait. By: Paul Belanger (pabelanger) 2011-04-25 21:21:23 We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information By: Leif Madsen (lmadsen) 2011-05-10 15:40:39 A SIP trace would have certainly helped here. By: Matt Jordan (mjordan) 2011-12-13 13:12:54.881-0600 This may have been resolved in the current 1.8 branch. Would you mind retesting your scenario with the lastest from the 1.8 branch? If you find it still exhibits the same problem in performing the local bridge, please attach a debug trace along with a packet capture. That should allow us to debug the issue. By: Matt Jordan (mjordan) 2012-01-26 09:03:29.118-0600 Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at http://www.asterisk.org/developers/bug-guidelines | ||