[Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly

2005-12-27 Thread James Sizemore
when my Cisco IAD send a call to my Asterisk gateway the gateway treats 
it as if I don't have a peer statement in sip.conf, when I do. Here are 
the first two packets, notice the Found no matching peer or user for 
'192.168.7.250:50437' on the second packet. Any one seen this before, 
or have a clue as to the problem?  Asterisk 1.0.9


sip.conf:
[bna-vonx-iad]
type=friend
context=trusted-out
host=192.168.7.250
canreinvite=no


Sip read:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP  192.168.7.250:5060;branch=z9hG4bK1A60
From: James Sizemore sip:[EMAIL PROTECTED];tag=19D8A640-5E9
To: sip:[EMAIL PROTECTED]
Date: Wed, 06 Mar 2002 00:27:08 GMT
Call-ID: [EMAIL PROTECTED]
Supported: 100rel,timer
Min-SE:  1800
Cisco-Guid: 3128236623-802099670-2154346748-2004044536
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: 6
Remote-Party-ID: James Sizemore 
sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off

Timestamp: 1015374428
Contact: sip:[EMAIL PROTECTED]:5060
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 191

v=0
o=CiscoSystemsSIP-GW-UserAgent 6047 8216 IN IP4 192.168.7.250
s=SIP Call
c=IN IP4 192.168.7.250
t=0 0
m=audio 16434 RTP/AVP 0
c=IN IP4 192.168.7.250
a=rtpmap:0 PCMU/8000
a=ptime:20

20 headers, 9 lines
Using latest request as basis request
Sending to 192.168.7.250 : 5060 (non-NAT)
Found no matching peer or user for '192.168.7.250:50437'
Found RTP audio format 0
Peer audio RTP is at port 192.168.7.250:16434
Found description format PCMU
Capabilities: us - 0x78e (gsm|ulaw|alaw|lpc10|g729|speex|ilbc), peer - 
audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined 
- 0x0 (nothing)

Looking for 615917 in default
list_route: hop: sip:[EMAIL PROTECTED]:5060
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.7.250:5060;branch=z9hG4bK1A60
From: James Sizemore sip:[EMAIL PROTECTED];tag=19D8A640-5E9
To: sip:[EMAIL PROTECTED];tag=as43478a8a
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Memphis ISDN-NET PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0 




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Re: [Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly

2005-12-27 Thread James Sizemore

I think I found what is munging up the peer lookup:

This call from another Asterisk box starts:

-- SIP read from 192.168.69.254:5060:

The peer lookup that fail reads:

-- SIP read from 192.168.7.250:52141:

Asterisk seem to be thrown off by the fact that the return port is not
5060, and fails the peer lookup.  This is a * bug then. I have 
documented it with both 1.0.9 and 1.2.1. Time to dig through the sip code.



James Sizemore wrote:
when my Cisco IAD send a call to my Asterisk gateway the gateway treats 
it as if I don't have a peer statement in sip.conf, when I do. Here are 
the first two packets, notice the Found no matching peer or user for 
'192.168.7.250:50437' on the second packet. Any one seen this before, 
or have a clue as to the problem?  Asterisk 1.0.9


sip.conf:
[bna-vonx-iad]
type=friend
context=trusted-out
host=192.168.7.250
canreinvite=no


Sip read:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP  192.168.7.250:5060;branch=z9hG4bK1A60
From: James Sizemore sip:[EMAIL PROTECTED];tag=19D8A640-5E9
To: sip:[EMAIL PROTECTED]
Date: Wed, 06 Mar 2002 00:27:08 GMT
Call-ID: [EMAIL PROTECTED]
Supported: 100rel,timer
Min-SE:  1800
Cisco-Guid: 3128236623-802099670-2154346748-2004044536
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: 6
Remote-Party-ID: James Sizemore 
sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off

Timestamp: 1015374428
Contact: sip:[EMAIL PROTECTED]:5060
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 191

v=0
o=CiscoSystemsSIP-GW-UserAgent 6047 8216 IN IP4 192.168.7.250
s=SIP Call
c=IN IP4 192.168.7.250
t=0 0
m=audio 16434 RTP/AVP 0
c=IN IP4 192.168.7.250
a=rtpmap:0 PCMU/8000
a=ptime:20

20 headers, 9 lines
Using latest request as basis request
Sending to 192.168.7.250 : 5060 (non-NAT)
Found no matching peer or user for '192.168.7.250:50437'
Found RTP audio format 0
Peer audio RTP is at port 192.168.7.250:16434
Found description format PCMU
Capabilities: us - 0x78e (gsm|ulaw|alaw|lpc10|g729|speex|ilbc), peer - 
audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined 
- 0x0 (nothing)

Looking for 615917 in default
list_route: hop: sip:[EMAIL PROTECTED]:5060
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.7.250:5060;branch=z9hG4bK1A60
From: James Sizemore sip:[EMAIL PROTECTED];tag=19D8A640-5E9
To: sip:[EMAIL PROTECTED];tag=as43478a8a
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Memphis ISDN-NET PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0


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