[asterisk-bugs] [JIRA] (ASTERISK-23757) One way audio and MOH at the Background for destination while transferring the calls

[Siva (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-23757?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Siva updated ASTERISK-23757:
----------------------------

    Status: Waiting for Feedback  (was: Waiting for Feedback)

Hello Rusty
I followed as per Michael suggestion, some times it's working and some times it's not working.I didn't upgraded the latest version in production environment.

I tried in my lab it's working.Sorry for the delay.

Michael thanks for your help on this.

Siva



> One way audio and MOH at the Background for destination while transferring the calls
> ------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-23757
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23757
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 1.8.12.2
>         Environment: OS-CentOS release 5.8, Asterisk 1.8 with Freepbx 2.9
>            Reporter: Siva
>            Assignee: Siva
>
> We are experiencing the problem with couple of sites.Below is our setup
> Setup 1 Freepbx(Asterisk) - Sip Trunk - Call Manager
> Setup 2 Freepbx(Asterisk) - Call Transfer over T1 trunk - Call Manager
> Note:It's working fine if we call the extension directly.Issue occurs only during the call transfer
> We are receiving calls to asterisk from outside and we are transferring the calls to another pbx cisco call manager over sip trunk.While transferring the calls they unable hear and they hear only MOH at the background.
> {noformat}
> FreePBX-CUCM
> host=CUCM IP
> type=friend
> disallow=all
> allow=ulaw
> qualify=yes
> dtmfmode=rfc2833
> nat=no
> FreePBX-Digium T1 GW(G200)
> host=G200IP
> username=xxxxxx
> secret=xxxxxx
> type=peer
> allow=all
> qualify=yes
> nat=no
> insecure=invite,port
> context=from-trunk
> {noformat}
> I already opened a case with digium case # is 00397744.Since i created ticket for gateway they are unable to provide more assistance on this.Please check the below email from digium support.
> I apologize for the delay, getting to the bottom of this issue has taken a fair amount of research. The good news is that this issue does not involve your G200 gateway. The bad news is that this is a bug with Asterisk itself. The offending parties are actually the two PBXs themselves.
> The problem stems from the way Cisco Call Manager formats its SDP reinvite, and how Asterisk interprets the request to re-activate an inactive media session when an offer does not contain a=sendrecv.
> Developers are aware of this issue, and are actively working to address it. It is a somewhat rare occurrence to see this issue, but luckily the information from your case can be added to the others to enhance development of the fix. I do not have a time frame on when a resolution will be released.
> I am sorry that I could not currently provide more assistance. As this ticket deals with your gateway device and not Asterisk, I will be closing the ticket.
> I recommend keeping an eye on the Asterisk mailing list if you'd like to keep abreast of development.



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