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Tim Howell
2005-Mar-15 10:15 UTC
[Asterisk-Users] Incoming calls from Cisco 1760 given wrong context...

I've installed Asterisk from the Asterisk @home distribution.
Ultimately I will be using Asterisk for voicemail for about 150 users.
Calls are (mostly) handled by a legacy PBX although we do have a couple
of Cisco 1760 routers that connect a remote office.

I've setup a SIP trunk that routes calls from Asterisk to the 1760, and
that works fine.  I've also configured one of the 1760s to route certain
calls to Asterisk.  However, the calls are placed in the
"from-sip-external" context that Asterisk @home uses for unidentified
SIP calls and are subsequently dropped.  I can make the calls connect by
modifying the from-sip-external context, but I would like to be able to
specify that calls from the router (on a static IP) are placed in a
different context.  Here is part of my sip_additional.conf:

[Cisco1760_mc]
type=friend
host=192.168.0.254
context=from-pstn
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=yes
qualify=yes

However, when I use sip debug to monitor an attempted call (711515 from
one of the phones connected to the Cisco), these lines appears in part
of the debug:

Found no matching peer or user for '192.168.0.254:53464'
Looking for 711515 in from-sip-external

Shouldn't it match Cisco1760_mc?

I've included the full debug below.

Thanks in advance for your help.  I'm happy to provide any additional
information.

--TWH

Sip read:
INVITE sip:711515@192.168.0.47:5060 SIP/2.0
Via: SIP/2.0/UDP  192.168.0.254:5060;branch=z9hG4bK1E6D
From: "Tim Howell" <sip:515@192.168.0.254>;tag=49582394-1813
To: <sip:711515@192.168.0.47>
Date: Tue, 15 Mar 2005 17:12:08 GMT
Call-ID: 32AAD959-94AC11D9-8E759110-AF18A82C@192.168.0.254
Supported: 100rel,timer
Min-SE:  1800
Cisco-Guid: 838919162-2494304729-2389872912-2937628716
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY
, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID: "Tim Howell"
<sip:515@192.168.0.254>;party=calling;screen=no;pr
ivacy=off
Timestamp: 1110906728
Contact: <sip:515@192.168.0.254:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 267

v=0
o=CiscoSystemsSIP-GW-UserAgent 6054 4992 IN IP4 192.168.0.254
s=SIP Call
c=IN IP4 192.168.0.254
t=0 0
m=audio 16946 RTP/AVP 0 100 19
c=IN IP4 192.168.0.254
a=rtpmap:0 PCMU/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:19 CN/8000
a=ptime:20

 20 headers, 12 lines
 Using latest request as basis request
 Sending to 192.168.0.254 : 5060 (non-NAT)
 Found RTP audio format 0
 Found RTP audio format 100
 Found RTP audio format 19
 Peer audio RTP is at port 192.168.0.254:16946
 Found description format PCMU
 Found description format X-NSE
 Found description format CN
 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing)
, combined - 0x4 (ulaw)
 Non-codec capabilities: us - 0x1 (g723), peer - 0x2 (gsm), combined -
0x0 (noth
ing)
 Found no matching peer or user for '192.168.0.254:53464'
 Looking for 711515 in from-sip-external
 list_route: hop: <sip:515@192.168.0.254:5060>
 Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK1E6D
From: "Tim Howell" <sip:515@192.168.0.254>;tag=49582394-1813
To: <sip:711515@192.168.0.47>;tag=as2a29cbba
Call-ID: 32AAD959-94AC11D9-8E759110-AF18A82C@192.168.0.254
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:711515@192.168.0.47>
Content-Length: 0


 to 192.168.0.254:5060
     -- Executing AbsoluteTimeout("SIP/192.168.0.254-094a8648",
"15") in
new sta
ck
     -- Set Absolute Timeout to 15
     -- Executing Congestion("SIP/192.168.0.254-094a8648",
"") in new
stack
Transmitting (no NAT):
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK1E6D
From: "Tim Howell" <sip:515@192.168.0.254>;tag=49582394-1813
To: <sip:711515@192.168.0.47>;tag=as2a29cbba
Call-ID: 32AAD959-94AC11D9-8E759110-AF18A82C@192.168.0.254
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:711515@192.168.0.47>
Content-Length: 0


 to 192.168.0.254:5060
   == Spawn extension (from-sip-external, 711515, 2) exited non-zero on
'SIP/192
.168.0.254-094a8648'
     -- Executing AbsoluteTimeout("SIP/192.168.0.254-094a8648",
"15") in
new sta
ck
     -- Set Absolute Timeout to 15
     -- Executing Congestion("SIP/192.168.0.254-094a8648",
"") in new
stack
   == Spawn extension (from-sip-external, h, 2) exited non-zero on
'SIP/192.168.
0.254-094a8648'
asterisk1*CLI>

Sip read:
ACK sip:711515@192.168.0.47:5060 SIP/2.0
Via: SIP/2.0/UDP  192.168.0.254:5060;branch=z9hG4bK1E6D
From: "Tim Howell" <sip:515@192.168.0.254>;tag=49582394-1813
To: <sip:711515@192.168.0.47>;tag=as2a29cbba
Date: Tue, 15 Mar 2005 17:12:08 GMT
Call-ID: 32AAD959-94AC11D9-8E759110-AF18A82C@192.168.0.254
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0


9 headers, 0 lines
Destroying call '32AAD959-94AC11D9-8E759110-AF18A82C@192.168.0.254'
asterisk1*CLI> sip no debug
SIP Debugging Disabled
asterisk1*CLI>
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