[Asterisk-Users] SIP debugging

Klaus Darilion klaus.mailinglists at pernau.at
Thu May 18 02:43:33 MST 2006 

Kevin P. Fleming wrote:
> Klaus Darilion wrote:
> 
>> Shouldn't there be some error indication if Asterisk discards a response?
> 
> Probably, although it's not clear here that Asterisk actually discarded
> anything. Without seeing the entire dialog, there's no way to be sure
> whether there were multiple Call-IDs, multiple tags, etc. involved.

The problem is caused be a forked call with pedantic=yes.

Asterisk --SIP--> Proxy ---SIP----> Sipura
                               \
                                ---> Cisco phone

The SIPURA sends the first 180 Ringing back. Then, Asterisk ignores the 
responses from the Cisco phone (180+200).

When setting pedantic=no, it works (I guess with pedantic=no Asterisk 
does not check the To tag (ugly)).

Is Asterisk not able of handling multiple early dialogs with pedantic=yes?

regards
Klaus

PS: Following the call flows

pedantic=yes:

     -- Executing Set("Zap/50-1", "enumresult=e001-366102 at enum.at43.at") 
in new stack
     -- Executing GotoIf("Zap/50-1", "0?103:3") in new stack
     -- Goto (frompbx,059966366102,3)
     -- Executing SetCIDNum("Zap/50-1", "00431234600265") in new stack
     -- Executing Dial("Zap/50-1", "SIP/e001-366102 at enum.at43.at|90") in 
new stack
     -- parse_srv: SRV mapped to host sip.at43.at, port 5060
We're at 213.174.230.213 port 10392
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 13 lines
Reliably Transmitting (NAT) to 83.136.32.160:5060:
INVITE sip:e001-366102 at enum.at43.at SIP/2.0
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport
From: "00431234600265" <sip:00431234600265 at 213.174.230.213>;tag=as6ce265a8
To: <sip:e001-366102 at enum.at43.at>
Contact: <sip:00431234600265 at 213.174.230.213>
Call-ID: 0baddae44d88ca6771babfc27bab2587 at 213.174.230.213
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 18 May 2006 09:31:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 293

v=0
o=root 9803 9803 IN IP4 213.174.230.213
s=session
c=IN IP4 213.174.230.213
t=0 0
m=audio 10392 RTP/AVP 8 0 3 97 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
     -- Called e001-366102 at enum.at43.at
poeast01*CLI>
<-- SIP read from 83.136.32.160:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060
From: "00431234600265" <sip:00431234600265 at 213.174.230.213>;tag=as6ce265a8
To: <sip:e001-366102 at enum.at43.at>
Call-ID: 0baddae44d88ca6771babfc27bab2587 at 213.174.230.213
CSeq: 102 INVITE
Server: OpenSer (1.0.0-tls (i386/linux))
Content-Length: 0

--- (8 headers 0 lines)---
poeast01*CLI>
<-- SIP read from 83.136.32.160:5060:
SIP/2.0 180 Ringing
t: <sip:e001-366102 at enum.at43.at>;tag=f1d48eba29dc7f4i0
f: "00431234600265" <sip:00431234600265 at 213.174.230.213>;tag=as6ce265a8
i: 0baddae44d88ca6771babfc27bab2587 at 213.174.230.213
CSeq: 102 INVITE
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060
Record-Route: 
<sip:PPC228685 at 83.136.32.162:5065>,<sip:83.136.32.160;ftag=as6ce265a8;lr=on>
Server: Sipura/SPA2000-3.1.2(NTb)
Contact: <sip:PPC228685 at 83.136.32.162:5065>
Content-Length: 0

--- (10 headers 0 lines)---
     -- SIP/enum.at43.at-3323 is ringing
poeast01*CLI>
<-- SIP read from 83.136.32.160:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060
From: "00431234600265" <sip:00431234600265 at 213.174.230.213>;tag=as6ce265a8
To: <sip:e001-366102 at enum.at43.at>;tag=000cce3a7bf804ac335c54e3-740eb2e1
Call-ID: 0baddae44d88ca6771babfc27bab2587 at 213.174.230.213
Date: Thu, 18 May 2006 09:31:25 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:klaus.darilion at 83.136.33.21:5060>
Record-Route: <sip:83.136.32.160;ftag=as6ce265a8;lr=on>
ontent-Length: 0

--- (11 headers 0 lines)---
Destroying call '0baddae44d88ca6771babfc27bab2587 at 213.174.230.213'


poeast01*CLI>
<-- SIP read from 83.136.32.160:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060
From: "00431234600265" <sip:00431234600265 at 213.174.230.213>;tag=as6ce265a8
To: <sip:e001-366102 at enum.at43.at>;tag=000cce3a7bf804ac335c54e3-740eb2e1
Call-ID: 0baddae44d88ca6771babfc27bab2587 at 213.174.230.213
Date: Thu, 18 May 2006 09:31:36 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:klaus.darilion at 83.136.33.21:5060>
Record-Route: <sip:83.136.32.160;ftag=as6ce265a8;lr=on>
Content-Type: application/sdp
Content-Length: 196

v=0
o=Cisco-SIPUA 14377 7540 IN IP4 83.136.33.21
s=SIP Call
c=IN IP4 83.136.33.21
t=0 0
m=audio 21174 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (12 headers 9 lines)---
Destroying call '0baddae44d88ca6771babfc27bab2587 at 213.174.230.213'
poeast01*CLI>
<-- SIP read from 83.136.32.160:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060
From: "00431234600265" <sip:00431234600265 at 213.174.230.213>;tag=as6ce265a8
To: <sip:e001-366102 at enum.at43.at>;tag=000cce3a7bf804ac335c54e3-740eb2e1
Call-ID: 0baddae44d88ca6771babfc27bab2587 at 213.174.230.213
Date: Thu, 18 May 2006 09:31:36 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:klaus.darilion at 83.136.33.21:5060>
Record-Route: <sip:83.136.32.160;ftag=as6ce265a8;lr=on>
Content-Type: application/sdp
Content-Length: 196

v=0
o=Cisco-SIPUA 14377 7540 IN IP4 83.136.33.21
s=SIP Call
c=IN IP4 83.136.33.21
t=0 0
m=audio 21174 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (12 headers 9 lines)---
Destroying call '0baddae44d88ca6771babfc27bab2587 at 213.174.230.213'
poeast01*CLI>
<-- SIP read from 83.136.32.160:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060
From: "00431234600265" <sip:00431234600265 at 213.174.230.213>;tag=as6ce265a8
To: <sip:e001-366102 at enum.at43.at>;tag=000cce3a7bf804ac335c54e3-740eb2e1
Call-ID: 0baddae44d88ca6771babfc27bab2587 at 213.174.230.213
Date: Thu, 18 May 2006 09:31:36 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:klaus.darilion at 83.136.33.21:5060>
Record-Route: <sip:83.136.32.160;ftag=as6ce265a8;lr=on>
Content-Type: application/sdp
Content-Length: 196

v=0
o=Cisco-SIPUA 14377 7540 IN IP4 83.136.33.21
s=SIP Call
c=IN IP4 83.136.33.21
t=0 0
m=audio 21174 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (12 headers 9 lines)---
Destroying call '0baddae44d88ca6771babfc27bab2587 at 213.174.230.213'

poeast01*CLI>
<-- SIP read from 83.136.32.160:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060
From: "00431234600265" <sip:00431234600265 at 213.174.230.213>;tag=as6ce265a8
To: <sip:e001-366102 at enum.at43.at>;tag=000cce3a7bf804ac335c54e3-740eb2e1
Call-ID: 0baddae44d88ca6771babfc27bab2587 at 213.174.230.213
Date: Thu, 18 May 2006 09:31:36 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:klaus.darilion at 83.136.33.21:5060>
Record-Route: <sip:83.136.32.160;ftag=as6ce265a8;lr=on>
Content-Type: application/sdp
Content-Length: 196

v=0
o=Cisco-SIPUA 14377 7540 IN IP4 83.136.33.21
s=SIP Call
c=IN IP4 83.136.33.21
t=0 0
m=audio 21174 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (12 headers 9 lines)---
Destroying call '0baddae44d88ca6771babfc27bab2587 at 213.174.230.213'






pedantic=no


     -- Executing Set("Zap/57-1", "enumresult=e001-366102 at enum.at43.at") 
in new stack
     -- Executing GotoIf("Zap/57-1", "0?103:3") in new stack
     -- Goto (frompbx,059966366102,3)
     -- Executing SetCIDNum("Zap/57-1", "00431234600265") in new stack
     -- Executing Dial("Zap/57-1", "SIP/e001-366102 at enum.at43.at|90") in 
new stack
     -- parse_srv: SRV mapped to host sip.at43.at, port 5060
We're at 213.174.230.213 port 11884
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 13 lines
Reliably Transmitting (NAT) to 83.136.32.160:5060:
INVITE sip:e001-366102 at enum.at43.at SIP/2.0
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK5743cf8d;rport
From: "00431234600265" <sip:00431234600265 at 213.174.230.213>;tag=as2d5cf8e9
To: <sip:e001-366102 at enum.at43.at>
Contact: <sip:00431234600265 at 213.174.230.213>
Call-ID: 74d41c6f2977d6544731c5be14843464 at 213.174.230.213
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 18 May 2006 09:33:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 295

v=0
o=root 10055 10055 IN IP4 213.174.230.213
s=session
c=IN IP4 213.174.230.213
t=0 0
m=audio 11884 RTP/AVP 8 0 3 97 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
     -- Called e001-366102 at enum.at43.at
poeast01*CLI>
<-- SIP read from 83.136.32.160:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK5743cf8d;rport=5060
From: "00431234600265" <sip:00431234600265 at 213.174.230.213>;tag=as2d5cf8e9
To: <sip:e001-366102 at enum.at43.at>
Call-ID: 74d41c6f2977d6544731c5be14843464 at 213.174.230.213
CSeq: 102 INVITE
Server: OpenSer (1.0.0-tls (i386/linux))
Content-Length: 0


--- (8 headers 0 lines)---
poeast01*CLI>
<-- SIP read from 83.136.32.160:5060:
SIP/2.0 180 Ringing
t: <sip:e001-366102 at enum.at43.at>;tag=cfbf759c15bc72d9i0
f: "00431234600265" <sip:00431234600265 at 213.174.230.213>;tag=as2d5cf8e9
i: 74d41c6f2977d6544731c5be14843464 at 213.174.230.213
CSeq: 102 INVITE
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK5743cf8d;rport=5060
Record-Route: 
<sip:PPC228685 at 83.136.32.162:5065>,<sip:83.136.32.160;ftag=as2d5cf8e9;lr=on>
Server: Sipura/SPA2000-3.1.2(NTb)
Contact: <sip:PPC228685 at 83.136.32.162:5065>
Content-Length: 0


--- (10 headers 0 lines)---
     -- SIP/enum.at43.at-a538 is ringing
poeast01*CLI>
<-- SIP read from 83.136.32.160:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK5743cf8d;rport=5060
From: "00431234600265" <sip:00431234600265 at 213.174.230.213>;tag=as2d5cf8e9
To: <sip:e001-366102 at enum.at43.at>;tag=000cce3a7bf804ad3f45745b-0ef87057
Call-ID: 74d41c6f2977d6544731c5be14843464 at 213.174.230.213
Date: Thu, 18 May 2006 09:33:39 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:klaus.darilion at 83.136.33.21:5060>
Record-Route: <sip:83.136.32.160;ftag=as2d5cf8e9;lr=on>
Content-Length: 0


--- (11 headers 0 lines)---
     -- SIP/enum.at43.at-a538 is ringing
     -- Zap/17-1 is making progress passing it to Zap/48-1

poeast01*CLI>
<-- SIP read from 83.136.32.160:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK5743cf8d;rport=5060
From: "00431234600265" <sip:00431234600265 at 213.174.230.213>;tag=as2d5cf8e9
To: <sip:e001-366102 at enum.at43.at>;tag=000cce3a7bf804ad3f45745b-0ef87057
Call-ID: 74d41c6f2977d6544731c5be14843464 at 213.174.230.213
Date: Thu, 18 May 2006 09:33:43 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:klaus.darilion at 83.136.33.21:5060>
Record-Route: <sip:83.136.32.160;ftag=as2d5cf8e9;lr=on>
Content-Type: application/sdp
Content-Length: 197

v=0
o=Cisco-SIPUA 19966 18440 IN IP4 83.136.33.21
s=SIP Call
c=IN IP4 83.136.33.21
t=0 0
m=audio 21176 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (12 headers 9 lines)---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 83.136.33.21:21176
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x8 
(alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:83.136.32.160;ftag=as2d5cf8e9;lr=on>
set_destination: Parsing <sip:83.136.32.160;ftag=as2d5cf8e9;lr=on> for 
address/port to send to
set_destination: set destination to 83.136.32.160, port 5060
Transmitting (NAT) to 83.136.32.160:5060:
ACK sip:klaus.darilion at 83.136.33.21:5060 SIP/2.0
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK2b1bb014;rport
Route: <sip:83.136.32.160;ftag=as2d5cf8e9;lr=on>
From: "00431234600265" <sip:00431234600265 at 213.174.230.213>;tag=as2d5cf8e9
To: <sip:e001-366102 at enum.at43.at>;tag=000cce3a7bf804ad3f45745b-0ef87057
Contact: <sip:00431234600265 at 213.174.230.213>
Call-ID: 74d41c6f2977d6544731c5be14843464 at 213.174.230.213
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
     -- SIP/enum.at43.at-a538 answered Zap/57-1

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