[Asterisk-Users] RE: [asterisk-dev] Possible Bug in SIP Stack.
Chris Modesitt
chris at octelecom.net
Fri Feb 24 13:01:34 MST 2006
Sorry Olle, I bet you wanted this from the SIP Proxy:)
Chris
-----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 --------------
Sip read:
INVITE sip:8019324299 at 132.1.42.180 SIP/2.0
To: <sip:8019324299 at 67.137.28.10:5060;user=phone>
From: "MODESITT,CHRIS " <sip:8017874906 at 63.98.126.237:5060;user=phone>;tag=5a670be4-1e606464-ed7e623f
Remote-Party-Id: "MODESITT,CHRIS " <sip:8017874906 at 63.98.126.237:5060;user=phone>;screen=yes;id-type=subscriber;party=calling;privacy=off
Call-ID: e3add72c-49ad4-1e606464 at 63.98.126.237
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:8017874906 at 63.98.126.237:5060;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:8017874906 at 63.98.126.237:5060;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:8017874906 at 63.98.126.237:5060;user=phone>;tag=5a670be4-1e606464-ed7e623f
To: <sip:8019324299 at 67.137.28.10:5060;user=phone>
Call-ID: e3add72c-49ad4-1e606464 at 63.98.126.237
CSeq: 5684092 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8019324299 at 132.1.42.180>
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:8017874906 at 63.98.126.237:5060;user=phone>;tag=5a670be4-1e606464-ed7e623f
To: <sip:8019324299 at 67.137.28.10:5060;user=phone>;tag=as1eb1110f
Call-ID: e3add72c-49ad4-1e606464 at 63.98.126.237
CSeq: 5684092 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8019324299 at 132.1.42.180>
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:8017874906 at 63.98.126.237:5060;user=phone>;tag=5a670be4-1e606464-ed7e623f
To: <sip:8019324299 at 67.137.28.10:5060;user=phone>;tag=as1eb1110f
Call-ID: e3add72c-49ad4-1e606464 at 63.98.126.237
CSeq: 5684092 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8019324299 at 132.1.42.180>
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:8017874906 at 63.98.126.237:5060;user=phone>;tag=5a670be4-1e606464-ed7e623f
To: <sip:8019324299 at 67.137.28.10:5060;user=phone>;tag=as1eb1110f
Call-ID: e3add72c-49ad4-1e606464 at 63.98.126.237
CSeq: 5684092 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8019324299 at 132.1.42.180>
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
-------------- next part --------------
<-- SIP read from 67.137.28.10:5060:
INVITE sip:8019324299 at 132.1.42.180 SIP/2.0
To: <sip:8019324299 at 67.137.28.10:5060;user=phone>
From: "MODESITT,CHRIS " <sip:8017874906 at 63.98.126.237:5060;user=phone>;tag=7262fc9a-1e60654a-ed7e623f
Remote-Party-Id: "MODESITT,CHRIS " <sip:8017874906 at 63.98.126.237:5060;user=phone>;screen=yes;id-type=subscriber;party=calling;privacy=off
Call-ID: b48e49c5-49beb-1e60654a at 63.98.126.237
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:8017874906 at 63.98.126.237:5060;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 - b48e49c5-49beb-1e60654a at 63.98.126.237
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:8017874906 at 63.98.126.237:5060;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:8017874906 at 63.98.126.237:5060;user=phone>;tag=7262fc9a-1e60654a-ed7e623f
To: <sip:8019324299 at 67.137.28.10:5060;user=phone>
Call-ID: b48e49c5-49beb-1e60654a at 63.98.126.237
CSeq: 5686393 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8019324299 at 132.1.42.180>
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:8017874906 at 63.98.126.237:5060;user=phone>;tag=7262fc9a-1e60654a-ed7e623f
To: <sip:8019324299 at 67.137.28.10:5060;user=phone>;tag=as7f6bdccd
Call-ID: b48e49c5-49beb-1e60654a at 63.98.126.237
CSeq: 5686393 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8019324299 at 132.1.42.180>
Content-Length: 0
---
amp-development*CLI>
<-- SIP read from 67.137.28.10:5060:
CANCEL sip:8019324299 at 132.1.42.180 SIP/2.0
To: <sip:8019324299 at 67.137.28.10:5060;user=phone>
From: "MODESITT,CHRIS " <sip:8017874906 at 63.98.126.237:5060;user=phone>;tag=7262fc9a-1e60654a-ed7e623f
Call-ID: b48e49c5-49beb-1e60654a at 63.98.126.237
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:8017874906 at 63.98.126.237:5060;user=phone>;tag=7262fc9a-1e60654a-ed7e623f
To: <sip:8019324299 at 67.137.28.10:5060;user=phone>;tag=as7f6bdccd
Call-ID: b48e49c5-49beb-1e60654a at 63.98.126.237
CSeq: 5686393 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8019324299 at 132.1.42.180>
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:8017874906 at 63.98.126.237:5060;user=phone>;tag=7262fc9a-1e60654a-ed7e623f
To: <sip:8019324299 at 67.137.28.10:5060;user=phone>;tag=as7f6bdccd
Call-ID: b48e49c5-49beb-1e60654a at 63.98.126.237
CSeq: 5686393 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8019324299 at 132.1.42.180>
Content-Length: 0
---
amp-development*CLI>
<-- SIP read from 67.137.28.10:5060:
ACK sip:8019324299 at 132.1.42.180 SIP/2.0
To: <sip:8019324299 at 67.137.28.10:5060;user=phone>;tag=as7f6bdccd
From: "MODESITT,CHRIS " <sip:8017874906 at 63.98.126.237:5060;user=phone>;tag=7262fc9a-1e60654a-ed7e623f
Call-ID: b48e49c5-49beb-1e60654a at 63.98.126.237
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK3b9808599158f965ded1e561f
CSeq: 5686393 ACK
Content-Length: 0
--- (7 headers 0 lines)---
Destroying call 'b48e49c5-49beb-1e60654a at 63.98.126.237'
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