[asterisk-dev] Extesion check after a SIP REFER?

Fernando Urzedo Fernando.Urzedo at locaweb.com.br
Thu Aug 14 17:47:36 CDT 2008


Hi all,
 
As I could understand, looks like after each REFER, Asterisk checks
whether the transferor can reach the target extension via its default
context. See bellow an example:
 
REFER sip:235 at 200.XXX.XXX.35 SIP/2.0
Via: SIP/2.0/UDP 200.YYY.YYY.87;branch=z9hG4bK76bec12d2FD069B6
From: <sip:10.ext101 at 200.YYY.YYY.87>;tag=F810E8F1-6A9966A
To: "EXT 1" <sip:235 at 200.XXX.XXX.35>;tag=as638c860f
CSeq: 2 REFER
Call-ID: 376487393bb717380578a58b3ad9707c at 200.XXX.XXX.35
Contact: <sip:10.ext101 at 200.YYY.YYY.87>
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.0.2.0972
Refer-To:
<sip:2 at servername.domain.com.br;user=phone?Replaces=fac455c9-735967e2-bf
510297%40200.YYY.YYY.87%3Bto-tag%3Das39a366f4%3Bfrom-tag%3D840EBF78-DF8C
E1D5>
Referred-By: <sip:10.ext101 at servername.domain.com.br>
Max-Forwards: 70
Content-Length: 0
 
<------------->
--- (12 headers 0 lines) ---
Call 376487393bb717380578a58b3ad9707c at 200.XXX.XXX.35
376487393bb717380578a58b3ad9707c at 200.XXX.XXX.35  got a SIP call transfer
from caller: (REFER)!
SIP transfer to extension 2 at 10.profile_controller
2 at 10.profile_controller by 10.ext101 at servername.domain.com.br
 

In this case, where the transfer was made by a Polycom IP Phone, my
assuption seems to be correct, since 10.profile_controller is the
default context for extension 10.ext101. However, when I do somehthing
similar using EyeBeam, I get the following:

REFER sip:235 at 200.XXX.XXX.35 SIP/2.0
To: <sip:235 at servername.domain.com.br>;tag=as177f9062
From: 10.ext2<sip:10.ext2 at servername.domain.com.br>;tag=1474c877
Via: SIP/2.0/UDP
200.YYY.YYY.206:10123;branch=z9hG4bK-d87543-928016851-1--d87543-;rport
Call-ID: 3b6519606a2c7276
CSeq: 4 REFER
Contact: <sip:10.ext2 at 200.YYY.YYY.206:10123>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Proxy-Authorization: Digest
username="10.ext2",realm="domain",nonce="4dc6929d",uri="sip:235 at 200.XXX.
XXX.35",response="4a827112db90724f2d6428f804051613",algorithm=MD5
User-Agent: eyeBeam release 3007n stamp 17816
Refer-To: "EXT
101"<sip:101 at 200.XXX.XXX.35?Replaces=6f8c3ea77bffb4665d2b04377e76b1c2%40
200.XXX.XXX.35%3Bto-tag%3Das069787dd%3Bfrom-tag%3D707d2653>
Referred-By: 10.ext2<sip:10.ext2 at servername.domain.com.br>
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Call 3b6519606a2c7276 got a SIP call transfer from caller: (REFER)!
Failed SIP Transfer to non-existing extension 101 in context
10.outgoing_names



In this case, the transfer fails because context 10.outgoing_names is
not prepared to handle calls to extension 101. Also, this is not the
dafault context for any of the involved peers.

My question is what defines the context Asterisk uses to check the
extensions after a REFER. Also, why Eyebeam's behaviour is different
from Polycom's?

Thanks in advance!



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