[asterisk-bugs] [Asterisk 0010415]: [have fix] Transfers stopped working when migrating from 1.4.9 to 1.4.10

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10415 
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Reported By:                atis
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   10415
Category:                   Resources/res_features
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.10  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             08-09-2007 03:47 CDT
Last Modified:              08-10-2007 10:05 CDT
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Summary:                    [have fix] Transfers stopped working when migrating
from 1.4.9 to 1.4.10
Description: 
I use SIP and Zap channels, and when switching from version 1.4.9 to 1.4.10
transfers stopped working. 

Solution was to unmerge changes in res_features.c
(http://bugs.digium.com/view.php?id=10327).

2007-07-30 17:11 +0000 [r77768-77778]  Joshua Colp <jcolp at digium.com>

        * res/res_features.c: (closes issue
http://bugs.digium.com/view.php?id=10327) Reported by: kkiely
          Instead of directly mucking with the extension/context/priority
          of the channel we are transferring when it has a PBX simply
call
          ast_async_goto on it. This will ensure that the channel gets
          handled properly and sent to the right place.


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---------------------------------------------------------------------- 
 atis - 08-10-07 10:05  
---------------------------------------------------------------------- 
For example:

1) SIP/90090 dials SIP/90220

Dial(SIP/90220|25|gtM(agent_call_answer^21167))

2) SIP/90220 transfers to SIP/90221 (by using blindxfer from
features.conf).

Both channels hungs up without any messages in log/console.

Providing dialplan would be quite complex for me - i believe you can read
affected dialplan from console output, but if you need it - ask me, ill try
to create test case. 

Console output goes in attached file.
Here goes SIP config (from mysql realtime) for 90090/90220/90221
(basically the same)

*************************** 1. row ***************************
            id: 601
        ext_id: 34648
          name: 90090
   accountcode: NULL
      amaflags: NULL
     callgroup: NULL
      callerid: device <90090>
   canreinvite: no
       context: default-sip
     defaultip: NULL
      dtmfmode: rfc2833
      fromuser: NULL
    fromdomain: NULL
   fullcontact: NULL
          host: dynamic
      insecure: NULL
      language: NULL
       mailbox: 90090 at device
     md5secret: NULL
           nat: yes
          deny: NULL
        permit: NULL
          mask: NULL
   pickupgroup: NULL
          port: 5060
       qualify: no
   restrictcid: NULL
    rtptimeout: NULL
rtpholdtimeout: NULL
        secret: xxxxx
          type: friend
      username: 90090
      disallow:
         allow: all
   musiconhold: NULL
    regseconds: 1186797160
        ipaddr: 81.198.164.2
      regexten:
cancallforward: yes
        setvar: 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
08-10-07 10:05  atis           Note Added: 0068725                          
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