[asterisk-users] How to use 'Transfer' to send calls to another asterisk?

[Deepesh D]

This doesn't work reliably well with all all clients. I tested it
using a zoiper soft phone and it worked. But from an ATA device it
failed. On the S2 server it failed to authenticate

The console of S2 showed
[Oct 12 18:21:06] WARNING[30483]: chan_sip.c:13952 check_auth:
username mismatch, have <XXXXX>, digest has <>
[Oct 12 18:21:06] NOTICE[30483]: chan_sip.c:22046
handle_request_invite: Failed to authenticate device
<sip:XXXXX at 192.168.1.1:14500>;tag=3047

On Fri, Oct 12, 2012 at 5:00 PM, Joshua Colp <jcolp at digium.com> wrote:
> Deepesh D wrote:
>>
>> I made these changes in dialplan and it worked. Thanks a lot.
>>
>> In most of the cases S1, S2 and C1 are in my control. But in some
>> cases the dialplan of C1 is not in my control. Also in some cases C1
>> can be any SIP client like a softphone or SIP device, so it wont work
>> in those case. Is there some way I can get those also working.
>
>
> You can certainly execute Transfer() and most clients will then send the
> call to where you have specified. If you have the same users on all possible
> servers you would Transfer to with the same username/password a challenge
> should occur and authentication happen.
>
> I haven't tested that though.
>
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at:  www.digium.com  & www.asterisk.org
>
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