SIP Transfer (REFER) to external SIP address?

Asterisk Asterisk Support
1

I’m using Asterisk 1.8.4.2, and I’m trying to Transfer() a call from one Asterisk server to another.

The REFER request comes in, with the correct Refer-To header- showing the correct SIP address to transfer to, but Asterisk gives me an error saying it can’t find that extension (the extension that exists on the remote server), in a local context- which is right- it doesn’t exist locally.

Why is asterisk looking for this extension in a local context, when the Refer-To header clearly points to a remote server?

My setup is like this:

Server A calls server B; server B has a dial plan

s,1,Answer()
s,2,do some stuff over agi
s,3,Transfer(100@c.c.c.c)

So server B sends server A a REFER request to server C, with Refer-To: sip:100@c.c.c.c

Server A receives the REFER request, but then fails with:

Call 10cfd8e65c0df2c759731d685ef6fefc@b.b.b.b got a SIP call transfer from callee: (REFER)!
Failed SIP Transfer to non-existing extension 100 in context main

Just as a test, I add a local 100 extension on server A under the main context, and it works- but the goal is to transfer to server C, and it seem to just be ignoring the remote IP address of server C.

I’m I missing something to make this work?

Thanks,

Mike

2

Asterisk is a back to back user agent, so it normally only looks at the user part and interprets it locally.

However, I believe there is a sip.conf option that tells it, at least for transfers, that it should take account of the domain part, in which case I got the impression from the code that it would generate real outgoing INVITEs.

Please note that the error handling for failed REFERs is fairly broken in 1.6 and I don’t believe it has been fixed in 1.8, although I haven’t checked this.

3

Thanks David,

Yeah- I haven’t been able to find any SIP commands that indicate that they make that work- plus- I went through the source code and- after a quick scan- it looks like the only way the transfer request handling code finishes successfully, is if it’s an attended transfer, or if it finds the extension locally- I don’t see anything in the current code base to suggest that it can send an INVITE out to another machine.

I guess I can do something really hokey and add a local extension that I transfer to, with a dialplan that just does a Dial() out to the real host.

Mike

4

I believe allowexternaldomains may have an effect.

5

Asterisk 1.8.7.0. Me too facing the same issue even after playing with allowexternaldomains.
However 302 works perfectly with external SIP address when promiscredir is set to yes.
@Panic, did you find solution to your problem?