[Asterisk-Users] RE: [asterisk-dev] Possible Bug in SIP Stack.

[Chris Modesitt]

I have included two files, one from asterisk 1.0.10 and one from 1.2.4. 

Thanks Olle

Chris Modesitt


-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Olle E Johansson
Sent: Friday, February 24, 2006 7:53 AM
To: Asterisk Developers Mailing List
Cc: asterisk-users at 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:8013793000 at 200.200.200.200:5060;user=phone>;tag=4fdc9d0e-1e600f94-ed7e6
23f
> 
> To: <sip:8014377860 at 67.137.28.10:5060;user=phone>;tag=as4fc8aa8a
> 
> Call-ID: 3a8530f4-43cb1-1e600f94 at 200.200.200.200
> 
> 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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-------------- next part --------------
amp-development*CLI> sip debug peer 3000
SIP Debugging Enabled for IP: 208.187.197.66:16945
  dialparties.agi: callerid = MODESITT,CHRIS
We're at 132.1.42.180 port 12018
Answering/Requesting with root capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 11 lines
Reliably Transmitting:
INVITE sip:3000 at 208.187.197.66:16945 SIP/2.0
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK02b961b2
From: "MODESITT,CHRIS " <sip:8017874906 at 132.1.42.180>;tag=as6a3d9b11
To: <sip:3000 at 208.187.197.66:16945>
Contact: <sip:8017874906 at 132.1.42.180>
Call-ID: 0f88bf276afd18a31cb50e9b3843bbab at 132.1.42.180
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 24 Feb 2006 19:41:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 2197 2197 IN IP4 132.1.42.180
s=session
c=IN IP4 132.1.42.180
t=0 0
m=audio 12018 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 - - - -
 (no NAT) to 208.187.197.66:16945
amp-development*CLI> 

Sip read: 
SIP/2.0 100 Trying
To: <sip:3000 at 208.187.197.66:16945>
From: "MODESITT,CHRIS " <sip:8017874906 at 132.1.42.180>;tag=as6a3d9b11
Call-ID: 0f88bf276afd18a31cb50e9b3843bbab at 132.1.42.180
CSeq: 102 INVITE
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK02b961b2
Server: Sipura/SPA2002-3.1.2(a)
Content-Length: 0


8 headers, 0 lines
amp-development*CLI> 

Sip read: 
SIP/2.0 180 Ringing
To: <sip:3000 at 208.187.197.66:16945>;tag=28364ccb72fa8d15i0
From: "MODESITT,CHRIS " <sip:8017874906 at 132.1.42.180>;tag=as6a3d9b11
Call-ID: 0f88bf276afd18a31cb50e9b3843bbab at 132.1.42.180
CSeq: 102 INVITE
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK02b961b2
Server: Sipura/SPA2002-3.1.2(a)
Content-Length: 0


8 headers, 0 lines
amp-development*CLI> 

Sip read: 
SIP/2.0 200 OK
To: <sip:3000 at 208.187.197.66:16945>;tag=28364ccb72fa8d15i0
From: "MODESITT,CHRIS " <sip:8017874906 at 132.1.42.180>;tag=as6a3d9b11
Call-ID: 0f88bf276afd18a31cb50e9b3843bbab at 132.1.42.180
CSeq: 102 INVITE
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK02b961b2
Contact: 3000 <sip:3000 at 208.187.197.66:16945>
Server: Sipura/SPA2002-3.1.2(a)
Content-Length: 237
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

=0p-development*CLI> 
o=- 375013 375013 IN IP4 208.187.197.66
s=-
c=IN IP4 208.187.197.66
t=0 0
m=audio 18412 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

12 headers, 12 lines
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 208.187.197.66:18412
Found description format PCMU
Found description format NSE
Found description format telephone-event
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)
list_route: hop: <sip:3000 at 208.187.197.66:16945>
set_destination: Parsing <sip:3000 at 208.187.197.66:16945> for address/port to send to
set_destination: set destination to 208.187.197.66, port 16945
Transmitting:
ACK sip:3000 at 208.187.197.66:16945 SIP/2.0
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK2559f914
From: "MODESITT,CHRIS " <sip:8017874906 at 132.1.42.180>;tag=as6a3d9b11
To: <sip:3000 at 208.187.197.66:16945>;tag=28364ccb72fa8d15i0
Contact: <sip:8017874906 at 132.1.42.180>
Call-ID: 0f88bf276afd18a31cb50e9b3843bbab at 132.1.42.180
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 208.187.197.66:16945
set_destination: Parsing <sip:3000 at 208.187.197.66:16945> for address/port to send to
set_destination: set destination to 208.187.197.66, port 16945
Reliably Transmitting:
BYE sip:3000 at 208.187.197.66:16945 SIP/2.0
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK05081e12
From: "MODESITT,CHRIS " <sip:8017874906 at 132.1.42.180>;tag=as6a3d9b11
To: <sip:3000 at 208.187.197.66:16945>;tag=28364ccb72fa8d15i0
Contact: <sip:8017874906 at 132.1.42.180>
Call-ID: 0f88bf276afd18a31cb50e9b3843bbab at 132.1.42.180
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 208.187.197.66:16945
amp-development*CLI> 

Sip read: 
SIP/2.0 200 OK
To: <sip:3000 at 208.187.197.66:16945>;tag=28364ccb72fa8d15i0
From: "MODESITT,CHRIS " <sip:8017874906 at 132.1.42.180>;tag=as6a3d9b11
Call-ID: 0f88bf276afd18a31cb50e9b3843bbab at 132.1.42.180
CSeq: 103 BYE
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK05081e12
Server: Sipura/SPA2002-3.1.2(a)
Content-Length: 0


8 headers, 0 lines
Destroying call '0f88bf276afd18a31cb50e9b3843bbab at 132.1.42.180'
-------------- next part --------------
We're at 132.1.42.180 port 18822
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines 
Reliably Transmitting (no NAT) to 208.187.197.66:16945:
INVITE sip:3000 at 208.187.197.66:16945 SIP/2.0
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK024a36ca
From: "MODESITT,CHRIS " <sip:8017874906 at 132.1.42.180>;tag=as7331df60
To: <sip:3000 at 208.187.197.66:16945>
Contact: <sip:8017874906 at 132.1.42.180>
Call-ID: 7153759722734b33355a3611290a3225 at 132.1.42.180
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 24 Feb 2006 18:41:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 4078 4078 IN IP4 132.1.42.180
s=session
c=IN IP4 132.1.42.180
t=0 0
m=audio 18822 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 - - - -

---
amp-development*CLI> 
<-- SIP read from 208.187.197.66:16945: 
SIP/2.0 100 Trying
To: <sip:3000 at 208.187.197.66:16945>
From: "MODESITT,CHRIS " <sip:8017874906 at 132.1.42.180>;tag=as7331df60
Call-ID: 7153759722734b33355a3611290a3225 at 132.1.42.180
CSeq: 102 INVITE
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK024a36ca
Server: Sipura/SPA2002-3.1.2(a)
Content-Length: 0


--- (8 headers 0 lines)---
amp-development*CLI> 
<-- SIP read from 208.187.197.66:16945: 
SIP/2.0 180 Ringing
To: <sip:3000 at 208.187.197.66:16945>;tag=f188f1d62b1a2b6ei0
From: "MODESITT,CHRIS " <sip:8017874906 at 132.1.42.180>;tag=as7331df60
Call-ID: 7153759722734b33355a3611290a3225 at 132.1.42.180
CSeq: 102 INVITE
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK024a36ca
Server: Sipura/SPA2002-3.1.2(a)
Content-Length: 0


--- (8 headers 0 lines)---
Reliably Transmitting (no NAT) to 208.187.197.66:16945:
CANCEL sip:3000 at 208.187.197.66:16945 SIP/2.0
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK024a36ca
From: "MODESITT,CHRIS " <sip:8017874906 at 132.1.42.180>;tag=as7331df60
To: <sip:3000 at 208.187.197.66:16945>
Contact: <sip:8017874906 at 132.1.42.180>
Call-ID: 7153759722734b33355a3611290a3225 at 132.1.42.180
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of call '7153759722734b33355a3611290a3225 at 132.1.42.180' in 15000 ms
amp-development*CLI> 
<-- SIP read from 208.187.197.66:16945: 
SIP/2.0 487 Request Terminated
To: <sip:3000 at 208.187.197.66:16945>;tag=f188f1d62b1a2b6ei0
From: "MODESITT,CHRIS " <sip:8017874906 at 132.1.42.180>;tag=as7331df60
Call-ID: 7153759722734b33355a3611290a3225 at 132.1.42.180
CSeq: 102 INVITE
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK024a36ca
Server: Sipura/SPA2002-3.1.2(a)
Content-Length: 0


--- (8 headers 0 lines)---
Transmitting (no NAT) to 208.187.197.66:16945:
ACK sip:3000 at 208.187.197.66:16945 SIP/2.0
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK024a36ca
From: "MODESITT,CHRIS " <sip:8017874906 at 132.1.42.180>;tag=as7331df60
To: <sip:3000 at 208.187.197.66:16945>;tag=f188f1d62b1a2b6ei0
Contact: <sip:8017874906 at 132.1.42.180>
Call-ID: 7153759722734b33355a3611290a3225 at 132.1.42.180
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Destroying call '7153759722734b33355a3611290a3225 at 132.1.42.180'
amp-development*CLI> 
<-- SIP read from 208.187.197.66:16945: 
SIP/2.0 200 OK
To: <sip:3000 at 208.187.197.66:16945>;tag=f188f1d62b1a2b6ei0
From: "MODESITT,CHRIS " <sip:8017874906 at 132.1.42.180>;tag=as7331df60
Call-ID: 7153759722734b33355a3611290a3225 at 132.1.42.180
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK024a36ca
Server: Sipura/SPA2002-3.1.2(a)
Content-Length: 0


--- (8 headers 0 lines)---
Destroying call '7153759722734b33355a3611290a3225 at 132.1.42.180'