| Summary: | ASTERISK-16353: One-way audio after exit from queue or after transfer | ||
| Reporter: | Valery (ks) | Labels: | |
| Date Opened: | 2010-07-12 07:42:44 | Date Closed: | 2011-06-07 14:05:04 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Applications/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ( 0) debug.log | |
| Description: | Incomming call from guest SIP user come to queue and after connect with queue member i have not audio from queue member With issue also on 1.6.2.9 | ||
| Comments: | By: Valery (ks) 2010-07-12 07:55:20 audio from queue member is out for 3-5 seconds By: Leif Madsen (lmadsen) 2010-07-12 11:24:54 This looks a lot like a NAT or firewall issue... By: Valery (ks) 2010-07-12 12:36:30 i use nat=yes, canreinvite=no on each SIP peer when i use nat=no, canreinvite=yes on each peer transfer work correctly, but if i use T option in Dial, issue is appear. firewall is disabled Also i try to install another sip server (Communigate Pro with full media proxy option) on this server, this issue is not appear there By: Valery (ks) 2010-07-12 12:44:30 This issue is random, with same settings with issue may both appear or disappear I try SIP -> Asterisk -> IAX2 -> Transfer to IAX2, this issue is appear By: Leif Madsen (lmadsen) 2010-07-20 11:07:38 Please provide the configuration along with the scenario and enough information that someone would be able to reproduce this issue. By: Paul Belanger (pabelanger) 2010-08-04 11:57:53 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 | ||