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

[Joshua Colp]

Deepesh D wrote:
> 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

Yes, it all depends on how the implementation in question handles it and 
how smart it is. You can't expect them all to behave as I mentioned. In 
the case of Asterisk it treats it as a SIP URI when promiscredir is 
enabled and doesn't use a peer entry, so it has no idea how to authenticate.

Cheers,

-- 
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