[asterisk-dev] qeustion about app_dial.c do_forward function getting the forwarding peer

Stefan Schmidt sst at sil.at
Mon Sep 28 03:18:35 CDT 2009


hi klaus,

thanks for your patch, ive testet it and everything works fine.

does this patch also will come to the next 1.6.1 version or does i have
to patch it again?

best regards

steve

Klaus Darilion schrieb:
> Hi Stefan!
> 
> Stefan Schmidt schrieb:
>> Hello klaus,
>>
>> thanks for the link to this issue, this was the patch i was thinking of,
>> my problem is that when i use your patch:
>> pbx_builtin_setvar_helper(o->chan, "FORWARDER", winner->name);
>> after inherit the orig and the winner channel, i just got the Local
>> channel 1 in this Var. What i need was the name of the redirecting channel.
>>
>> in 1.6 the app_dial looks diferent to 1.4 so there is the incoming orig
>> chan (in) the winner which is first c = o->chan which is the channel
>> which starts the forward (with the peer i need) but is set to the new
>> outgoing channel after
>> c = o->chan = ast_request(tech, in->nativeformats, stuff, &cause); (row:
>> 533)
> 
> Not sure what you mean. I have tested with 1.6.2 and it works as 
> expected. See new uploaded patch at:
>   https://issues.asterisk.org/view.php?id=14592
> 
> In my patch I only use the SIP peer name, no channel names.
> 
> regards
> klaus
> 
>> what i have done to make it useable for me is a
>> pbx_builtin_setvar_helper(in,"FORWARDING_PEER",c->name);
>> before this row, so i really have the channel name of the peer which
>> starts the forward, but set it to the orig incoming channel (in).
>>
>> so i have to use
>> pbx_builtin_getvar_helper(chan->chan_list.next->chan_list.next,"FORWARDING_PEER")
>> to get the channel name which starts the forward.
>>
>> i still dont know why there are 2 local channels when doing a forward,
>> but i think that the first local points to the incoming and the second
>> local channel starts the outgoing channel and when this is successfull
>> the orig and the outgoing is bridged and the local channels are stopped.
>>
>> maybe you shout think about this when doing this patch.
>>
>> best regards
>>
>> steve
>>
>> Klaus Darilion schrieb:
>>> Hi Stefan!
>>>
>>> Maybe this helps too:
>>> https://issues.asterisk.org/view.php?id=14592
>>>
> 
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
> 
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev

-- 
Für weitere Fragen stehen wir gerne unter voip at sil.at oder
059944 - 2440 zur Verfügung.

Mit freundlichen Grüssen
-- 
Stefan Schmidt
Sysadmin/VOIP // voip at sil.at // Tel 059944-2440//
-------------------------------------------------
SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 //
A-1160 Wien // Fax 059944-9000 // www.sil.at  //
-------------------------------------------------



More information about the asterisk-dev mailing list