Sorry Olle, I bet you wanted this from the SIP Proxy:)

Chris


-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson
Sent: Friday, February 24, 2006 7:53 AM
To: Asterisk Developers Mailing List
Cc: asterisk-users@lists.digium.com
Subject: Re: [asterisk-dev] Possible Bug in SIP Stack.

Chris Modesitt wrote:
> I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is 
> APX 8000 -> Interaction SIP Proxy 3.0.013 -> asterisk server.   When I 
> use Asterisk version 10.0.10 everything works perfectly, however when I 
> use 1.2.4 I lose the ability to receive calls from the PSTN.  All I get 
> is the following error in my SIP Proxies error logs:
> 
>  
> 
> SIPSession::proxyResponseImmediately(): Failed to retrieve next Via, 
> don't know where to send responseSIP/2.0 180 Ringing
> 
> From: "MODESITT,CHRIS " 
>
<sip:[EMAIL PROTECTED]:5060;user=phone>;tag=4fdc9d0e-1e600f94-ed7e6
23f
> 
> To: <sip:[EMAIL PROTECTED]:5060;user=phone>;tag=as4fc8aa8a
> 
> Call-ID: [EMAIL PROTECTED]
> 
> CSeq: 5466974 INVITE
> 
> User-Agent: Asterisk PBX
> 
>  
> 
> I still can make outbound calls with no-problems, any ideas?
> 
>  
Can you get SIP debug logs from a call setup with Asterisk 1.0.10 and 
1.2.4 so we can compare them and see what happened?

Thanks
/Olle
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Sip read: 
INVITE sip:[EMAIL PROTECTED] SIP/2.0
To: <sip:[EMAIL PROTECTED]:5060;user=phone>
From: "MODESITT,CHRIS " <sip:[EMAIL 
PROTECTED]:5060;user=phone>;tag=5a670be4-1e606464-ed7e623f
Remote-Party-Id: "MODESITT,CHRIS " <sip:[EMAIL 
PROTECTED]:5060;user=phone>;screen=yes;id-type=subscriber;party=calling;privacy=off
Call-ID: [EMAIL PROTECTED]
CSeq: 5684092 INVITE
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK125da0ff92aa0fb40559dbdd9, 
SIP/2.0/UDP 63.98.126.237:5060;branch=z9hG4bK03635e104072f0ae
Max-Forwards: 69
Contact: <sip:[EMAIL PROTECTED]:5060;user=phone>
Supported: replaces
Content-Type: application/sdp
Accept: application/sdp
Accept-Encoding: 
Accept-Language: en
User-Agent: Lucent-Universal-Gateway
Content-Length: 241

v=0
o=voicecore1 509633636 509633636 IN IP4 63.98.126.237
s=Session SDP
c=IN IP4 63.98.126.237
t=0 0
m=audio 43486 RTP/AVP 0 101
a=silenceSupp:off
a=ecan:b on g168
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=rtpmap:0 PCMU/8000

16 headers, 11 lines
Using latest request as basis request
Sending to 67.137.28.10 : 5060 (non-NAT)
Found peer 'dialout'
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 63.98.126.237:43486
Found description format telephone-event
Found description format PCMU
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
Looking for 8019324299 in from-pstn
list_route: hop: <sip:[EMAIL PROTECTED]:5060;user=phone>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK125da0ff92aa0fb40559dbdd9, 
SIP/2.0/UDP 63.98.126.237:5060;branch=z9hG4bK03635e104072f0ae
From: "MODESITT,CHRIS " <sip:[EMAIL 
PROTECTED]:5060;user=phone>;tag=5a670be4-1e606464-ed7e623f
To: <sip:[EMAIL PROTECTED]:5060;user=phone>
Call-ID: [EMAIL PROTECTED]
CSeq: 5684092 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0


 to 67.137.28.10:5060
  dialparties.agi: callerid = MODESITT,CHRIS
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK125da0ff92aa0fb40559dbdd9, 
SIP/2.0/UDP 63.98.126.237:5060;branch=z9hG4bK03635e104072f0ae
From: "MODESITT,CHRIS " <sip:[EMAIL 
PROTECTED]:5060;user=phone>;tag=5a670be4-1e606464-ed7e623f
To: <sip:[EMAIL PROTECTED]:5060;user=phone>;tag=as1eb1110f
Call-ID: [EMAIL PROTECTED]
CSeq: 5684092 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0


 to 67.137.28.10:5060
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK125da0ff92aa0fb40559dbdd9, 
SIP/2.0/UDP 63.98.126.237:5060;branch=z9hG4bK03635e104072f0ae
From: "MODESITT,CHRIS " <sip:[EMAIL 
PROTECTED]:5060;user=phone>;tag=5a670be4-1e606464-ed7e623f
To: <sip:[EMAIL PROTECTED]:5060;user=phone>;tag=as1eb1110f
Call-ID: [EMAIL PROTECTED]
CSeq: 5684092 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0


 to 67.137.28.10:5060
We're at 132.1.42.180 port 13582
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK125da0ff92aa0fb40559dbdd9, 
SIP/2.0/UDP 63.98.126.237:5060;branch=z9hG4bK03635e104072f0ae
From: "MODESITT,CHRIS " <sip:[EMAIL 
PROTECTED]:5060;user=phone>;tag=5a670be4-1e606464-ed7e623f
To: <sip:[EMAIL PROTECTED]:5060;user=phone>;tag=as1eb1110f
Call-ID: [EMAIL PROTECTED]
CSeq: 5684092 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 2254 2254 IN IP4 132.1.42.180
s=session
c=IN IP4 132.1.42.180
t=0 0
m=audio 13582 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 to 67.137.28.10:5060
<-- SIP read from 67.137.28.10:5060: 
INVITE sip:[EMAIL PROTECTED] SIP/2.0
To: <sip:[EMAIL PROTECTED]:5060;user=phone>
From: "MODESITT,CHRIS " <sip:[EMAIL 
PROTECTED]:5060;user=phone>;tag=7262fc9a-1e60654a-ed7e623f
Remote-Party-Id: "MODESITT,CHRIS " <sip:[EMAIL 
PROTECTED]:5060;user=phone>;screen=yes;id-type=subscriber;party=calling;privacy=off
Call-ID: [EMAIL PROTECTED]
CSeq: 5686393 INVITE
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK3b9808599158f965ded1e561f, 
SIP/2.0/UDP 63.98.126.237:5060;branch=z9hG4bK0363b7f6980e531b
Max-Forwards: 69
Contact: <sip:[EMAIL PROTECTED]:5060;user=phone>
Supported: replaces
Content-Type: application/sdp
Accept: application/sdp
Accept-Encoding: 
Accept-Language: en
User-Agent: Lucent-Universal-Gateway
Content-Length: 241

v=0
o=voicecore1 509633866 509633866 IN IP4 63.98.126.237
s=Session SDP
c=IN IP4 63.98.126.237
t=0 0
m=audio 44044 RTP/AVP 0 101
a=silenceSupp:off
a=ecan:b on g168
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=rtpmap:0 PCMU/8000

--- (16 headers 11 lines)---
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to 67.137.28.10 : 5060 (non-NAT)
Found peer 'dialout'
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 63.98.126.237:44044
Found description format telephone-event
Found description format PCMU
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
Looking for 8019324299 in from-pstn (domain 132.1.42.180)
list_route: hop: <sip:[EMAIL PROTECTED]:5060;user=phone>
Transmitting (no NAT) to 67.137.28.10:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK3b9808599158f965ded1e561f, 
SIP/2.0/UDP 
63.98.126.237:5060;branch=z9hG4bK0363b7f6980e531b;received=67.137.28.10
From: "MODESITT,CHRIS " <sip:[EMAIL 
PROTECTED]:5060;user=phone>;tag=7262fc9a-1e60654a-ed7e623f
To: <sip:[EMAIL PROTECTED]:5060;user=phone>
Call-ID: [EMAIL PROTECTED]
CSeq: 5686393 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0


---
Transmitting (no NAT) to 67.137.28.10:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK3b9808599158f965ded1e561f, 
SIP/2.0/UDP 
63.98.126.237:5060;branch=z9hG4bK0363b7f6980e531b;received=67.137.28.10
From: "MODESITT,CHRIS " <sip:[EMAIL 
PROTECTED]:5060;user=phone>;tag=7262fc9a-1e60654a-ed7e623f
To: <sip:[EMAIL PROTECTED]:5060;user=phone>;tag=as7f6bdccd
Call-ID: [EMAIL PROTECTED]
CSeq: 5686393 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0


---
amp-development*CLI> 
<-- SIP read from 67.137.28.10:5060: 
CANCEL sip:[EMAIL PROTECTED] SIP/2.0
To: <sip:[EMAIL PROTECTED]:5060;user=phone>
From: "MODESITT,CHRIS " <sip:[EMAIL 
PROTECTED]:5060;user=phone>;tag=7262fc9a-1e60654a-ed7e623f
Call-ID: [EMAIL PROTECTED]
CSeq: 5686393 CANCEL
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK3b9808599158f965ded1e561f, 
SIP/2.0/UDP 63.98.126.237:5060;branch=z9hG4bK0363b7f6980e531b
Max-Forwards: 69
Reason: Q.850;cause=0;text="mapping-not-set"
User-Agent: Lucent-Universal-Gateway
Content-Length: 0


--- (10 headers 0 lines)---
Sending to 67.137.28.10 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 67.137.28.10:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK3b9808599158f965ded1e561f, 
SIP/2.0/UDP 
63.98.126.237:5060;branch=z9hG4bK0363b7f6980e531b;received=67.137.28.10
From: "MODESITT,CHRIS " <sip:[EMAIL 
PROTECTED]:5060;user=phone>;tag=7262fc9a-1e60654a-ed7e623f
To: <sip:[EMAIL PROTECTED]:5060;user=phone>;tag=as7f6bdccd
Call-ID: [EMAIL PROTECTED]
CSeq: 5686393 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0


---
Transmitting (no NAT) to 67.137.28.10:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK3b9808599158f965ded1e561f, 
SIP/2.0/UDP 
63.98.126.237:5060;branch=z9hG4bK0363b7f6980e531b;received=67.137.28.10
From: "MODESITT,CHRIS " <sip:[EMAIL 
PROTECTED]:5060;user=phone>;tag=7262fc9a-1e60654a-ed7e623f
To: <sip:[EMAIL PROTECTED]:5060;user=phone>;tag=as7f6bdccd
Call-ID: [EMAIL PROTECTED]
CSeq: 5686393 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0


---
amp-development*CLI> 
<-- SIP read from 67.137.28.10:5060: 
ACK sip:[EMAIL PROTECTED] SIP/2.0
To: <sip:[EMAIL PROTECTED]:5060;user=phone>;tag=as7f6bdccd
From: "MODESITT,CHRIS " <sip:[EMAIL 
PROTECTED]:5060;user=phone>;tag=7262fc9a-1e60654a-ed7e623f
Call-ID: [EMAIL PROTECTED]
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK3b9808599158f965ded1e561f
CSeq: 5686393 ACK
Content-Length: 0


--- (7 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'
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