[Asterisk-Users] Asterisk connecting to nortel CS 1000 as sip trunk Need help with final piece (incoming call) outgoing works.

Jerry Geis geisj at pagestation.com
Thu Jun 2 10:27:35 MST 2005 

All,

I am connecting to a CS 1000 nortel PBX. I can call out,
I have limited success with call in. I get debug traffic that a call
is coming in but I get the message "Unable to create/find channel".

I was expecting that incoming calls over the trunk would
be handled from my sip definition and goto the nortel context. It is not.

Below is the actual incoming call debug information.
I am using Asterisk 1.0.7

file:sip.conf
; There is no REGISTER for a nortel box - use qualify=yes
[QuadNortel]
type=friend
dtmfmode=rfc2833
username=SXNTM1SS1
disallow=all
allow=ulaw
allow=alaw
context=nortel
host=192.168.45.194
canreinvite=yes
qualify=yes
-------------------------------------------------------------------

Sip read: 
INVITE sip:2828;phone-context=cdp.udp at qg.com:5060;maddr=161.49.198.102;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0
From: <sip:3173241052;phone-context=+1 at qg.com;user=phone>;tag=c22da8c0-13c4-429efa71-657b8da-2ed
To: <sip:2828;phone-context=cdp.udp at qg.com;user=phone>
Call-ID: 103548e8-c22da8c0-13c4-429efa71-657b8da-517 at qg.com
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.45.194:5060;branch=z9hG4bK-429efa71-657b8da-2522
Max-Forwards: 70
Supported: 100rel,sipvc,replaces
User-Agent: Nortel CS1000 SIP GW: release=4.0 version=sse-4.00.31
P-Asserted-Identity: <sip:3173241052;phone-context=+1 at qg.com;user=phone>
Privacy: none
History-Info: <sip:2828;phone-context=cdp.udp at qg.com;transport=udp;user=phone>;index=1
x-nt-corr-id: 000000460c18100206 at 0001af0d4517-c0a82de1
x-nt-calling-id: <sip:3173241052;phone-context=+1 at qg.com>
Contact: <sip:3173241052;phone-context=+1 at qg.com:5060;maddr=192.168.45.194;transport=udp;user=phone>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Type: multipart/mixed ;boundary=unique-boundary-1
Content-Length: 645

--unique-boundary-1
Content-Type: application/SDP

v=0
o=- 95 1 IN IP4 192.168.45.194
s=-
t=0 0
m=audio 5234 RTP/AVP 0 8
c=IN IP4 192.168.45.197
a=ptime:20
a=maxptime:20
a=sendrecv

--unique-boundary-1
Content-Type: application/x-nt-mcdn-frag-hex ;version=sse-4.00.31 ;base=x2611
Content-Disposition: signal ;handling=optional

05006702
0107130081900000a2
09090f00e9a083000100e7
1315070011fa0f00a10d02010102020100cc046605123c
--unique-boundary-1
Content-Type: application/x-nt-epid-frag-hex ;version=sse-4.00.31 ;base=x2611
Content-Disposition: signal ;handling=optional

011201
00:02:b3:f6:58:cc
--unique-boundary-1--

18 headers, 28 lines

Using latest request as basis request

Sending to 192.168.45.194 : 5060 (non-NAT)

Found peer 'QuadNortel'
Jun  2 12:23:53 NOTICE[12001]: chan_sip.c:2638 process_sdp: Content is 'multipart/mixed ;boundary=unique-boundary-1', not 'application/sdp'


Sip read: 
INVITE sip:2828;phone-context=cdp.udp at qg.com:5060;maddr=161.49.198.102;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0
From: <sip:3173241052;phone-context=+1 at qg.com;user=phone>;tag=c22da8c0-13c4-429efa71-657b8da-2ed
To: <sip:2828;phone-context=cdp.udp at qg.com;user=phone>
Call-ID: 103548e8-c22da8c0-13c4-429efa71-657b8da-517 at qg.com
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.45.194:5060;branch=z9hG4bK-429efa71-657b8da-2522
Max-Forwards: 70
Supported: 100rel,sipvc,replaces
User-Agent: Nortel CS1000 SIP GW: release=4.0 version=sse-4.00.31
P-Asserted-Identity: <sip:3173241052;phone-context=+1 at qg.com;user=phone>
Privacy: none
History-Info: <sip:2828;phone-context=cdp.udp at qg.com;transport=udp;user=phone>;index=1
x-nt-corr-id: 000000460c18100206 at 0001af0d4517-c0a82de1
x-nt-calling-id: <sip:3173241052;phone-context=+1 at qg.com>
Contact: <sip:3173241052;phone-context=+1 at qg.com:5060;maddr=192.168.45.194;transport=udp;user=phone>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Type: multipart/mixed ;boundary=unique-boundary-1
Content-Length: 645

--unique-boundary-1
Content-Type: application/SDP

v=0
o=- 95 1 IN IP4 192.168.45.194
s=-
t=0 0
m=audio 5234 RTP/AVP 0 8
c=IN IP4 192.168.45.197
a=ptime:20
a=maxptime:20
a=sendrecv

--unique-boundary-1
Content-Type: application/x-nt-mcdn-frag-hex ;version=sse-4.00.31 ;base=x2611
Content-Disposition: signal ;handling=optional

05006702
0107130081900000a2
09090f00e9a083000100e7
1315070011fa0f00a10d02010102020100cc046605123c
--unique-boundary-1
Content-Type: application/x-nt-epid-frag-hex ;version=sse-4.00.31 ;base=x2611
Content-Disposition: signal ;handling=optional

011201
00:02:b3:f6:58:cc
--unique-boundary-1--

18 headers, 28 lines

Ignoring this request
Jun  2 12:23:53 NOTICE[12001]: chan_sip.c:7427 handle_request: Unable to create/find channel

Transmitting (no NAT):
SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP 192.168.45.194:5060;branch=z9hG4bK-429efa71-657b8da-2522
From: <sip:3173241052;phone-context=+1 at qg.com;user=phone>;tag=c22da8c0-13c4-429efa71-657b8da-2ed
To: <sip:2828;phone-context=cdp.udp at qg.com;user=phone>;tag=as1ca95a06
Call-ID: 103548e8-c22da8c0-13c4-429efa71-657b8da-517 at qg.com
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2828 at 161.49.198.102>
Content-Length: 0


 to 192.168.45.194:5060

Destroying call '103548e8-c22da8c0-13c4-429efa71-657b8da-517 at qg.com'


Sip read: 
ACK sip:2828;phone-context=cdp.udp at qg.com:5060;maddr=161.49.198.102;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0
From: <sip:3173241052;phone-context=+1 at qg.com;user=phone>;tag=c22da8c0-13c4-429efa71-657b8da-2ed
To: <sip:2828;phone-context=cdp.udp at qg.com;user=phone>;tag=as1ca95a06
Call-ID: 103548e8-c22da8c0-13c4-429efa71-657b8da-517 at qg.com
CSeq: 1 ACK
Via: SIP/2.0/UDP 192.168.45.194:5060;branch=z9hG4bK-429efa71-657b8da-2522
Max-Forwards: 70
User-Agent: Nortel CS1000 SIP GW: release=4.0 version=sse-4.00.31
x-nt-corr-id: 000000460c18100206 at 0001af0d4517-c0a82de1
Contact: <sip:3173241052;phone-context=+1 at qg.com:5060;maddr=192.168.45.194;transport=udp;user=phone>
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Length: 0



12 headers, 0 lines

Destroying call '103548e8-c22da8c0-13c4-429efa71-657b8da-517 at qg.com'

*CLI> 
*CLI> stop now

Beginning asterisk shutdown....

Executing last minute cleanups

  == Destroying any remaining musiconhold processes

Asterisk cleanly ending (0).


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