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", "[EMAIL PROTECTED]")
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/[EMAIL PROTECTED]|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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport
From: "00431234600265" <sip:[EMAIL PROTECTED]>;tag=as6ce265a8
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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 [EMAIL PROTECTED]
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:[EMAIL PROTECTED]>;tag=as6ce265a8
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]>;tag=f1d48eba29dc7f4i0
f: "00431234600265" <sip:[EMAIL PROTECTED]>;tag=as6ce265a8
i: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK44b15398;rport=5060
Record-Route:
<sip:[EMAIL PROTECTED]:5065>,<sip:83.136.32.160;ftag=as6ce265a8;lr=on>
Server: Sipura/SPA2000-3.1.2(NTb)
Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=as6ce265a8
To: <sip:[EMAIL PROTECTED]>;tag=000cce3a7bf804ac335c54e3-740eb2e1
Call-ID: [EMAIL PROTECTED]
Date: Thu, 18 May 2006 09:31:25 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:[EMAIL PROTECTED]:5060>
Record-Route: <sip:83.136.32.160;ftag=as6ce265a8;lr=on>
ontent-Length: 0
--- (11 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'
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:[EMAIL PROTECTED]>;tag=as6ce265a8
To: <sip:[EMAIL PROTECTED]>;tag=000cce3a7bf804ac335c54e3-740eb2e1
Call-ID: [EMAIL PROTECTED]
Date: Thu, 18 May 2006 09:31:36 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:[EMAIL PROTECTED]: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 '[EMAIL PROTECTED]'
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:[EMAIL PROTECTED]>;tag=as6ce265a8
To: <sip:[EMAIL PROTECTED]>;tag=000cce3a7bf804ac335c54e3-740eb2e1
Call-ID: [EMAIL PROTECTED]
Date: Thu, 18 May 2006 09:31:36 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:[EMAIL PROTECTED]: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 '[EMAIL PROTECTED]'
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:[EMAIL PROTECTED]>;tag=as6ce265a8
To: <sip:[EMAIL PROTECTED]>;tag=000cce3a7bf804ac335c54e3-740eb2e1
Call-ID: [EMAIL PROTECTED]
Date: Thu, 18 May 2006 09:31:36 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:[EMAIL PROTECTED]: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 '[EMAIL PROTECTED]'
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:[EMAIL PROTECTED]>;tag=as6ce265a8
To: <sip:[EMAIL PROTECTED]>;tag=000cce3a7bf804ac335c54e3-740eb2e1
Call-ID: [EMAIL PROTECTED]
Date: Thu, 18 May 2006 09:31:36 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:[EMAIL PROTECTED]: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 '[EMAIL PROTECTED]'
pedantic=no
-- Executing Set("Zap/57-1", "[EMAIL PROTECTED]")
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/[EMAIL PROTECTED]|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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK5743cf8d;rport
From: "00431234600265" <sip:[EMAIL PROTECTED]>;tag=as2d5cf8e9
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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 [EMAIL PROTECTED]
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:[EMAIL PROTECTED]>;tag=as2d5cf8e9
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]>;tag=cfbf759c15bc72d9i0
f: "00431234600265" <sip:[EMAIL PROTECTED]>;tag=as2d5cf8e9
i: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 213.174.230.213:5060;branch=z9hG4bK5743cf8d;rport=5060
Record-Route:
<sip:[EMAIL PROTECTED]:5065>,<sip:83.136.32.160;ftag=as2d5cf8e9;lr=on>
Server: Sipura/SPA2000-3.1.2(NTb)
Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=as2d5cf8e9
To: <sip:[EMAIL PROTECTED]>;tag=000cce3a7bf804ad3f45745b-0ef87057
Call-ID: [EMAIL PROTECTED]
Date: Thu, 18 May 2006 09:33:39 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=as2d5cf8e9
To: <sip:[EMAIL PROTECTED]>;tag=000cce3a7bf804ad3f45745b-0ef87057
Call-ID: [EMAIL PROTECTED]
Date: Thu, 18 May 2006 09:33:43 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=as2d5cf8e9
To: <sip:[EMAIL PROTECTED]>;tag=000cce3a7bf804ad3f45745b-0ef87057
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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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