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

[Alexander Anikin (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-27138?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Alexander Anikin updated ASTERISK-27138:
----------------------------------------

    Assignee: Dmitry Melekhov  (was: Alexander Anikin)
      Status: Waiting for Feedback  (was: Triage)

Hi Dmitry,

I did some analysis of the issue. First thing is strange working of H.245 tunneling from CCME. You have enabled it on CCME peer (sladzar) but he reply with tunneling off in call from asterisk to CCME, you can see it on first call in the attached dump.
Second call initiated by CCME and contain h245tunneling is on, but he don't reply to facility packet which tunnel H.245 TCS signal. It's reason of silence in that call. And it's could be CCME don't understand PROGRESS signal and block call completeley due to it.
I recommend switch off h245tunneling on asterisk and retest this case.

Second thing is about codec and can be adressed to ASTERISK-27137. First call inititated with G.729 codec which is disabled on sladzar peer and it's subject of 27137. But some time later call switch G.711U and work ok. I think CCME switch call codec to codec configured on final subscriber of CCME. Normal way for that is to send empty tcs packet and renegotiate full TCS/MSD procedure but CCME just replace codec on existing rtp session. Asterisk allow that so there are no sound problem.





> 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
>            Assignee: Dmitry Melekhov
>         Attachments: asterisk.cap, h323_log, sladzar.cap
>
>
> 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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