[Asterisk-Dev] sip refer

[Olle E. Johansson]

Richard wrote:

>>So my humble advice is: Fix your dialplan! If you are using SIP,
>>the dialplan should be able to handle any sip URI.
> 
> Not really. I don't define any domain or user in *. Everything is defined in
> ser and * just uses autopeer. For example, if I have a 3 digit dialing for
> company A and B. Both uses extension 200 but on different SIP domains. When
> * gets a sip REFER, it will be 200 at companyA.com and 200 at companyB.com. If it
> goes through the dial plan, there is no way to tell which company 200
> belongs to. In chan_sip.c, I see that it strips the domain part and only
> looks at the extension.

Check the SIPDOMAIN variable. If the domain doesn't show up there on a
REFER, it's a bug.

/O