[asterisk-bugs] [Asterisk 0012152]: No sound during bringe ZAP<->SIP
noreply at bugs.digium.com
noreply at bugs.digium.com
Tue Mar 18 08:29:02 CDT 2008
The following issue has been RESOLVED.
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http://bugs.digium.com/view.php?id=12152
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Reported By: kowalma
Assigned To: file
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Project: Asterisk
Issue ID: 12152
Category: Channels/chan_sip/General
Reproducibility: random
Severity: minor
Priority: normal
Status: resolved
Asterisk Version: 1.4.18
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!): 106038
Disclaimer on File?: N/A
Request Review:
Resolution: no change required
Fixed in Version:
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Date Submitted: 03-05-2008 14:42 CST
Last Modified: 03-18-2008 08:29 CDT
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Summary: No sound during bringe ZAP<->SIP
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 at na-miasto:1] NoCDR("Zap/124-1", "") in new stack
-- Executing [511 at 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 at 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
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file - 03-18-08 08:29
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Closed per reporter.
Issue History
Date Modified Username Field Change
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03-18-08 08:29 file Status feedback => resolved
03-18-08 08:29 file Resolution open => no change
required
03-18-08 08:29 file Assigned To => file
03-18-08 08:29 file Note Added: 0084130
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