[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:comment-tabpanel&focusedCommentId=218293#comment-218293 ] 

Siva edited comment on ASTERISK-23757 at 5/19/14 5:17 PM:
----------------------------------------------------------

Michael,

Thank you for your quick response, i tried as per your suggestion now they are able to hear my voice very low but still MOH is playing at the background.The voice is low due to MOH is playing at the background.

  -- Started music on hold, class 'default', on SIP/digiumtrunk-0000074b
    -- Called SIP/siptrunkccm/4100
    -- SIP/siptrunkccm-00000754 is ringing
    -- SIP/siptrunkccm-00000754 is ringing
    -- SIP/siptrunkccm-00000754 is ringing
    -- Call on SIP/siptrunkccm-00000754 placed on hold
    -- Started music on hold, class 'default', on SIP/siptrunkccm-00000754
    -- SIP/2656-00000753 requested media update control 20, passing it to SIP/siptrunkccm-00000754
    -- Stopped music on hold on SIP/digiumtrunk-0000074b
    -- SIP/siptrunkccm-00000754 answered SIP/digiumtrunk-0000074b
    -- Locally bridging SIP/digiumtrunk-0000074b and SIP/siptrunkccm-00000754
[May 19 15:31:49] WARNING[14350]: channel.c:4934 ast_write: Codec mismatch on channel SIP/siptrunkccm-00000754 setting write format to slin from ulaw native formats 0x4 (ulaw)


was (Author: siva.percipia):
Michael,

Thank you for your quick response, i tried as per your suggestion now they are able to hear my voice very low but still MOH is playing at the background.The voice is low due to MOH is playing at the background.

  -- Started music on hold, class 'default', on SIP/digiumtrunk-0000074b
    -- Called SIP/owbrccm/4100
    -- SIP/owbrccm-00000754 is ringing
    -- SIP/owbrccm-00000754 is ringing
    -- SIP/owbrccm-00000754 is ringing
    -- Call on SIP/owbrccm-00000754 placed on hold
    -- Started music on hold, class 'default', on SIP/owbrccm-00000754
    -- SIP/2656-00000753 requested media update control 20, passing it to SIP/owbrccm-00000754
    -- Stopped music on hold on SIP/digiumtrunk-0000074b
    -- SIP/owbrccm-00000754 answered SIP/digiumtrunk-0000074b
    -- Locally bridging SIP/digiumtrunk-0000074b and SIP/owbrccm-00000754
[May 19 15:31:49] WARNING[14350]: channel.c:4934 ast_write: Codec mismatch on channel SIP/owbrccm-00000754 setting write format to slin from ulaw native formats 0x4 (ulaw)

> 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.
> 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
> 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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