[asterisk-bugs] [Asterisk 0010436]: indications.c: Can't generate that much data!
noreply at bugs.digium.com
noreply at bugs.digium.com
Mon Aug 13 10:26:18 CDT 2007
The following issue has been RESOLVED.
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http://bugs.digium.com/view.php?id=10436
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Reported By: bradley
Assigned To: qwell
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Project: Asterisk
Issue ID: 10436
Category: Resources/res_indications
Reproducibility: always
Severity: minor
Priority: normal
Status: resolved
Asterisk Version: 1.2.23
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
Resolution: fixed
Fixed in Version:
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Date Submitted: 08-13-2007 05:32 CDT
Last Modified: 08-13-2007 10:26 CDT
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Summary: indications.c: Can't generate that much data!
Description:
Asterisk 1.2.23, mISDN 1_1_5 with Digium B410P card.
When incoming calls from mISDN driver deliver a call to an extension or
ringgroup the destinarion SIP device(s) rings but the ISDN caller hears no
ringing tone and the asterisk log shows:
Aug 10 17:43:44 VERBOSE[1810] logger.c: -- Executing Dial("mISDN/1-1",
"SIP/1000|15|tr") in new stack
Aug 10 17:43:44 DEBUG[1810] chan_sip.c: Setting NAT on RTP to 524288
Aug 10 17:43:44 DEBUG[1810] chan_sip.c: Outgoing Call for 1000
Aug 10 17:43:44 VERBOSE[1810] logger.c: -- Called 1000
Aug 10 17:43:44 DEBUG[1810] channel.c: Driver for channel 'mISDN/1-1' does
not support indication 3, emulating it
Aug 10 17:43:44 DEBUG[1810] channel.c: Scheduling timer at 160 sample
intervals
Aug 10 17:43:44 DEBUG[1810] channel.c: Building translator from alaw to
SLINEAR for spies on channel mISDN/1-1
Aug 10 17:43:44 DEBUG[1810] channel.c: Generator got voice, switching to
phase locked mode
Aug 10 17:43:44 DEBUG[1810] channel.c: Scheduling timer at 0 sample
intervals
Aug 10 17:43:44 WARNING[1810] indications.c: Can't generate that much
data!
Aug 10 17:43:44 DEBUG[1810] channel.c: Auto-deactivating generator
Aug 10 17:43:44 DEBUG[1810] channel.c: Scheduling timer at 0 sample
intervals
Aug 10 17:43:44 VERBOSE[1810] logger.c: -- SIP/1000-082184d0 is
ringing
Calls between SIP handsets do hear ringing tone as expected.
If "r" is removed from the Dial options then an ISDN caller does hear a
ringing tone.
/etc/asterisk/indications.conf is the stock Asterisk file.
When "r" Dial is used then asterisk should be able to generate a ringing
tone for all channels.
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qwell - 08-13-07 10:26
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Please test using Asterisk 1.4, and reopen if you can reproduce.
Issue History
Date Modified Username Field Change
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08-13-07 10:26 qwell Assigned To => qwell
08-13-07 10:26 qwell Note Added: 0068773
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