Hi guys,
I see lot of posts related to CUCM to TB sip trunks. I tried those. but still no luck. i think this may be something to do with new Cisco call manager config. :-)
From the X-lite softphone which is registered with TB can make calls to Cisco end without any error. But when Cisco phone dials to the TB its getting fast busy.
my TB ix trixbox CE 2.6.2.2 .
TB ip - 192.168.1.4
CUCM - 192.168.1.201
this is my Trixbox Trunk setting.
[From_CUCM]
host=192.168.1.4
type=peer
context=from-internal
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=yes
qualify=yes
[To_CCUM]
host=192.168.1.201
type=peer
context=from-internal
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=yes
qualify=yes
=============================
from the CUCM to TB end and i have create new security sip trunk profile with UDP and i have apply that to the trunk.
here is the Sip Debug for the reference
===============================
Cisco phone ip - 192.168.1.3
X-lite phone ip - 192.168.1.2
===============================
Reliably Transmitting (no NAT) to 192.168.1.4:5060:
OPTIONS sip:192.168.1.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK644ab17b;rport
From: "Unknown"
To:
Contact:
Call-ID: 073241933d852ebf1f23dfa566e7549c@192.168.1.4
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 24 May 2009 17:28:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
OPTIONS sip:192.168.1.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK644ab17b;rport
From: "Unknown"
To:
Contact:
Call-ID: 073241933d852ebf1f23dfa566e7549c@192.168.1.4
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 24 May 2009 17:28:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
--- (13 headers 0 lines) ---
Looking for s in from-sip-external (domain 192.168.1.4)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK644ab17b;received=192.168.1.4;rport=5060
From: "Unknown"
To:
Call-ID: 073241933d852ebf1f23dfa566e7549c@192.168.1.4
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Accept: application/sdp
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK644ab17b;received=192.168.1.4;rport=5060
From: "Unknown"
To:
Call-ID: 073241933d852ebf1f23dfa566e7549c@192.168.1.4
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Accept: application/sdp
Content-Length: 0
--- (12 headers 0 lines) ---
Really destroying SIP dialog '073241933d852ebf1f23dfa566e7549c@192.168.1.4' Method: OPTIONS
is it something to do with CUCM config. im really confuse about this sip:Unknown@192.168.1.4. why trixbox sending SIP msgs as unknown.
Please help me out. thanks guys
Member Since:
2008-11-06