Summary:ASTERISK-11584: No sound during bringe ZAP<->SIP
Reporter:Marcin Kowalczyk (kowalma)Labels:
Date Opened:2008-03-05 14:42:58.000-0600Date Closed:2011-06-07 14:08:21
Versions:Frequency of
Description:I've got problem with no sound between SIP <-> ZAP. Problem causes randomly. rasterisk says:

-- B-channel 0/1 successfully restarted on span 4
   -- Accepting call from '240' to '511' on channel 0/31, span 4
   -- Executing [511@na-miasto:1] NoCDR("Zap/124-1", "") in new stack
   -- Executing [511@na-miasto:2] Dial("Zap/124-1", "SIP/kowal&SIP/kowal2|600") in new stack
   -- Called kowal
   -- Called kowal2
   -- SIP/kowal2-0847a790 is ringing
   -- SIP/kowal-08475448 is ringing
   -- SIP/kowal2-0847a790 answered Zap/124-1
[Mar  5 21:16:25] WARNING[3937]: rtp.c:2047 ast_rtp_stop: Unable to cancel schedule ID 0.  This is probably a bug (rtp.c: ast_rtp_stop, line 2047).
== Spawn extension (na-miasto, 511, 2) exited non-zero on 'Zap/124-1'
   -- Executing [h@na-miasto:1] Hangup("Zap/124-1", "16") in new stack
 == Spawn extension (na-miasto, h, 1) exited non-zero on 'Zap/124-1'
   -- Hungup 'Zap/124-1'
   -- B-channel 0/2 successfully restarted on span 4
   -- Remote UNIX connection disconnected


According to issue 12142 and 12132 I've tried to install 1.4.19rc1 but svn reports 1.4.18r106038 as newest version.
In this r106038 I've got problem with sound on SIP.
Comments:By: Joshua C. Colp (jcolp) 2008-03-05 17:06:46.000-0600

You need to provide much more information as attachments, such as sip debug and rtp debug as well as a general network topology. Are the devices behind NAT, are you, etc.

By: Marcin Kowalczyk (kowalma) 2008-03-18 05:11:03

This issue can be closed. I've installed 1.4.rc2 and problem dissipated

By: Joshua C. Colp (jcolp) 2008-03-18 08:29:01

Closed per reporter.