[asterisk-bugs] [JIRA] (ASTERISK-27138) ooh323, no audio from Cisco CallManager Express ver.11.5 to asterisk

Asterisk Team (JIRA) noreply at issues.asterisk.org
Tue Jul 18 01:43:58 CDT 2017


    [ https://issues.asterisk.org/jira/browse/ASTERISK-27138?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=237722#comment-237722 ] 

Asterisk Team commented on ASTERISK-27138:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> ooh323, no audio from  Cisco CallManager Express ver.11.5 to asterisk
> ---------------------------------------------------------------------
>
>                 Key: ASTERISK-27138
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27138
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Addons/chan_ooh323
>    Affects Versions: 13.13.1
>         Environment: Centos 6
>            Reporter: Dmitry Melekhov
>
> We need to establish h323 trunk from asterisk to cisco call manager express.
> Here is scheme
> PBX--isdn pri--asterisk--h323--ast-neftisa--h323--cisco
> Here is config from asterisk side:
> [sladzar]
> type=friend
> ;type=peer
> context=sladzar
> ip=10.56.6.1
> port=1720
> ;e164=101
> disallow=all
> allow=alaw
> allow=ulaw
> ;allow=g729
> fastStart=no
> ;fastStart=yes
> h245tunneling=yes
> ;canreinvite=no
> directmedia=yes
> directrtpsetup=yes
> dtmfmode=inband
> ;nat=yes
> When I as PBX user place call from asterisk to cisco call manager user eveything is fine.
> But if the same user calls me- call passes, but there is no audio at all.
> Looks like bug, because they say they can call cisco-cisco.
> Thank you!



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