[asterisk-bugs] [JIRA] (ASTERISK-22686) Asterisk does not sends RTP when transfer is done in telco side

[Matt Jordan (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-22686?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Matt Jordan reassigned ASTERISK-22686:
--------------------------------------

    Assignee:     (was: Matt Jordan)
    
> Asterisk does not sends RTP when transfer is done in telco side
> ---------------------------------------------------------------
>
>                 Key: ASTERISK-22686
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22686
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 1.8.23.1
>            Reporter: Dalius M.
>         Attachments: 2013-10-08_2.pcap
>
>
> Hello,
> We have found strange Asterisk issue.
> Phone attached to Asterisk (A) calls to SIP provider (B).
> (B) Puts call on hold and does attended transfer to (C).
> Asterisk plays MOH to (A) phone. When transfer is done, SIP provider sends INVITE, but Asterisk does not strart to send RTP.
> SIP provider told us that it is Asterisk bug.
> I have attached dump file from PBX's side.
> There is 2 VoIP calls, one from my softphone (IP 78.59.84.76) to PBX (IP 82.135.219.194) and other from PBX to SIP provider (IP 85.206.138.84).
> I have also tried Asterisk 11.3 version and Cisco SPA 504G (firmware 7.5.5) phone with same results, transfered calls are dropped

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