Accepting REFER command from an Asterix IPBX

[Joris06]

Hi,

I try to establish communication between an Asterisk IPBX and an OXE, and I have a problem :
I have 3 users :
A and B are OXE users, and C is an Asterisk user.
If A call C, then C want to transfer the call to B, my Asterisk send a REFER command to the OXE, but, the OXE refuse this command, so the transfer fail.

Is there a way to force my OXE to accept the REFER command ?

Both of my IPBX are directly interconnected, on the same LAN, with no nat between.

Joris

PS : excuse my poor english, I'm french, and I'm not good with foreign languages :/

[cavagnaro]

OXE rejects? Why?
Can you post some logs?
Also, can't Asterisk maybe configure their OXE trunk to not to use REFER and test?

[Joris06]

Here are the logs of my Asterisk server, the last line "503 Service Unavailable" seem to indicate that the OXE reject the command.
I don't know how to log SIP errors on the OXE.

Code: Select all

set_destination: Parsing <sip:6041@10.10.11.30;transport=UDP> for address/port to send to
set_destination: set destination to 10.10.11.30:5060
Reliably Transmitting (no NAT) to 10.10.11.30:5060:
REFER sip:6041@10.10.11.30;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.10.11.81:5060;branch=z9hG4bK2b74a42d
Max-Forwards: 70
From: <sip:6199@10.10.11.81;user=phone>;tag=as5546fe5d
To: <sip:6041@10.10.11.30;user=phone>;tag=9da5aa2f809d80b0ef3870f033d27d93
Contact: <sip:6199@10.10.11.81:5060>
Call-ID: ff8706c24b59bcc1d3784d17a17e8587@10.10.11.30
CSeq: 102 REFER
User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
Refer-To: <sip:6042@oxe>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Referred-By: <sip:6199@10.10.11.81:5060>
Content-Length: 0


---

<--- SIP read from UDP:10.10.11.30:5060 --->
SIP/2.0 202 Accepted
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE
Contact: <sip:6041@10.10.11.30;transport=UDP>
Supported: replaces,timer,path,100rel
User-Agent: OmniPCX Enterprise R11.0.1 k1.520.22.b
To: <sip:6041@10.10.11.30;user=phone>;tag=9da5aa2f809d80b0ef3870f033d27d93
From: <sip:6199@10.10.11.81;user=phone>;tag=as5546fe5d
Call-ID: ff8706c24b59bcc1d3784d17a17e8587@10.10.11.30
CSeq: 102 REFER
Via: SIP/2.0/UDP 10.10.11.81:5060;branch=z9hG4bK2b74a42d
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:10.10.11.30:5060 --->
NOTIFY sip:6199@10.10.11.81:5060 SIP/2.0
Content-Type: message/sipfrag
Contact: <sip:6041@10.10.11.30;transport=UDP>
Supported: replaces,timer,path,100rel
User-Agent: OmniPCX Enterprise R11.0.1 k1.520.22.b
Event: refer
Subscription-State: terminated;reason=noresource
To: <sip:6199@10.10.11.81;user=phone>;tag=as5546fe5d
From: <sip:6041@10.10.11.30;user=phone>;tag=9da5aa2f809d80b0ef3870f033d27d93
Call-ID: ff8706c24b59bcc1d3784d17a17e8587@10.10.11.30
CSeq: 672202134 NOTIFY
Via: SIP/2.0/UDP 10.10.11.30;branch=z9hG4bK6f3776962e2a15a1bae0cf4de440d132
Content-Length: 33

SIP/2.0 503 Service Unavailable

I could use an INVITE command instead, but if I do, the SIP channels aren't released, and 2 channels are used until the call is finished.

Joris

[Joris06]

Hi,

I found what was wrong in my configuration

It was a newbie mistake : in my Asterisk configuration, I had named my peer "oxe", so in the field "Refer-To" of the "REFER" command, there was "SIP/6042@oxe".
But my OXE was not named "oxe", so the command was rejected because "SIP/6042@oxe" was unknown.

Now it works just fine

Joris

[cavagnaro]

Nice. Other option is you to use ip address.