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

[Dalius M. (JIRA)]

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

Dalius M. updated ASTERISK-22686:
---------------------------------

    Description: 
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


  was:
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.



    
> 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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