[asterisk-bugs] [JIRA] (ASTERISK-17883) SIP CANCEL is broken when phone is not registered to asterisk (sip friend and/or sip user)
Walter Doekes (JIRA)
noreply at issues.asterisk.org
Tue Oct 16 02:41:34 CDT 2012
[ https://issues.asterisk.org/jira/browse/ASTERISK-17883?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=198471#comment-198471 ]
Walter Doekes edited comment on ASTERISK-17883 at 10/16/12 2:39 AM:
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Artem: are you able to test a newer version and/or include 'core set debug 10' logs?
was (Author: wdoekes):
Atrem: are you able to test a newer version and/or include 'core set debug 10' logs?
> SIP CANCEL is broken when phone is not registered to asterisk (sip friend and/or sip user)
> ------------------------------------------------------------------------------------------
>
> Key: ASTERISK-17883
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-17883
> Project: Asterisk
> Issue Type: Bug
> Components: Channels/chan_sip/General
> Affects Versions: 1.8.4
> Reporter: Artem Makhutov
> Severity: Minor
> Attachments: sip.log
>
>
> I have discovered a bug while playing with asterisk and with the embedded sip client of an android mobile phone.
> Canceling a call is broken when the mobile phone is not registered to asterisk.
> When calling from the android phone and the other party picks up the phone everything works just fine. But when the other party does not pick up the phone and I hangup the call on the mobile then the other phone will never stop ringing.
> This problem does not occur when the mobile phone is registered to asterisk.
> Here is the relevant configuration of sip.conf:
> [2005]
> type=friend
> defaultuser=2005
> transport=udp
> fromuser=2005
> context=internal
> host=dynamic
> fromdomain=xxxxxxx
> secret=xxxxxxx
> qualify=yes
> nat=yes
> directmedia=no
> disallow=all
> allow=speex
> allow=alaw
> allow=ulaw
> allow=gsm
> allow=g729
> allow=h264
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