[Home]

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
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
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

****** ADDITIONAL INFORMATION ******

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.