[Asterisk-Users] SIP

Concent Telecom igor at concent.net
Wed Dec 17 10:44:59 MST 2003 

Hi,

Could somebody help me this SIP trasport?
I'm receiving SIP "invite" with CLI of calling party from the SIP gateway, aster that my IVR has to answer the call.

sip.conf:
=========
[general]
port = 5060
bindaddr = 0.0.0.0
context = incomingsip
videosupport=yes                ; Turn on support for SIP video
disallow=all                    ; Disallow all codecs
allow=g729
allow=ulaw                      ; Allow codecs in order of preference
allow=alaw

extensions.conf:
================
[incomingsip]
exten => _X.,1,AGI,ivr.pl

sip debug:
==========

Sip read: >
INVITE sip:8005095690 at x.x.x.x SIP/2.0
Max-Forwards: 5
To: 8005095690 <sip:8005095690 at x.x.x.x:5060;user=phone>
From: <sip:002125095690 at x.x.x.x>;tag=839C4028-E83
Call-ID: 8780-3280658280-475041 at x.x.x.x
CSeq: 1 INVITE
Via: SIP/2.0/UDP x.x.x.x:5060
Contact: sip:002125095690 at x.x.x.x:5060
Content-Type: application/sdp
Content-Length: 267
  
v=0
o=labis01 1234 5795 IN IP4 x.x.x.x
s=sip call
c=IN IP4 x.x.x.x
t=0 0
m=audio 17184 RTP/AVP 18 0 4 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
  
10 headers, 12 lines
Using latest request as basis request
Sending to x.x.x.x : 5060 (non-NAT)
Found audio format UNKN
Found audio format UNKN
Found audio format ULAW
Found audio format ALAW
Found audio format UNKN
Found description format G729
Found description format PCMU
Found description format G723
Found description format PCMA
Found description format telephone-event
Capabilities: us - 268, them - 269/854015, combined - 268
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 8005095690 in incomingsip
list_route: hop: <sip:002125095690 at x.x.x.x:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP x.x.x.x:5060
From: <sip:002125095690 at x.x.x.x>;tag=839C4028-E83
To: 8005095690 <sip:8005095690 at x.x.x.x:5060;user=phone>;tag=as6e87bf67
Call-ID: 8780-3280658280-475041 at x.x.x.x
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8005095690 at x.x.x.x>
Content-Length: 0
  
 
 to x.x.x.x:5060
    -- Executing AGI("SIP/-0830f3d0", "ivr-main.pl") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/ivr-main.pl
We're at x.x.x.x port 11038
Video is at x.x.x.x port 10436
Answering with preferred capability 256
Answering with preferred capability 4
Answering with preferred capability 8
Answering with non-codec capability 1
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.x.x:5060
From: <sip:002125095690 at x.x.x.x>;tag=839C4028-E83
To: 8005095690 <sip:8005095690 at x.x.x.x:5060;user=phone>;tag=as6e87bf67
Call-ID: 8780-3280658280-475041 at x.x.x.x
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8005095690 at x.x.x.x>
Content-Type: application/sdp
Content-Length: 260
  
v=0
o=root 9017 9017 IN IP4 x.x.x.x
s=session
c=IN IP4 x.x.x.x
t=0 0
m=audio 11038 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 10436 RTP/AVP
  
 to x.x.x.x:5060
Sip read: >
BYE sip:8005095690 at x.x.x.x SIP/2.0
Max-Forwards: 5
To: 8005095690 <sip:8005095690 at x.x.x.x:5060;user=phone>;tag=as6e87bf67
From: <sip:002125095690 at x.x.x.x>;tag=839C4028-E83
Call-ID: 8780-3280658280-475041 at x.x.x.x
CSeq: 2 BYE
Via: SIP/2.0/UDP x.x.x.x:5060
Contact: sip:002125095690 at x.x.x.x:5060
Content-Length: 0
  
 
9 headers, 0 lines
Sending to x.x.x.x : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.x.x:5060
From: <sip:002125095690 at x.x.x.x>;tag=839C4028-E83
To: 8005095690 <sip:8005095690 at x.x.x.x:5060;user=phone>;tag=as6e87bf67
Call-ID: 8780-3280658280-475041 at x.x.x.x
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8005095690 at x.x.x.x>
Content-Length: 0
  
 
 to x.x.x.x:5060
  == Spawn extension (incomingsip, 8005095690, 1) exited non-zero on 'SIP/-0830f3d0'
Retransmitting #1 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.x.x:5060
From: <sip:002125095690 at x.x.x.x>;tag=839C4028-E83
To: 8005095690 <sip:8005095690 at x.x.x.x:5060;user=phone>;tag=as6e87bf67
Call-ID: 8780-3280658280-475041 at x.x.x.x
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8005095690 at x.x.x.x>
Content-Type: application/sdp
Content-Length: 260
  
v=0
o=root 9017 9017 IN IP4 x.x.x.x
s=session
c=IN IP4 x.x.x.x
t=0 0
m=audio 11038 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 10436 RTP/AVP
835.3e
 to x.x.x.x:5060
Retransmitting #2 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.x.x:5060
From: <sip:002125095690 at x.x.x.x>;tag=839C4028-E83
To: 8005095690 <sip:8005095690 at x.x.x.x:5060;user=phone>;tag=as6e87bf67
Call-ID: 8780-3280658280-475041 at x.x.x.x
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8005095690 at x.x.x.x>
Content-Type: application/sdp
Content-Length: 260
  
v=0
o=root 9017 9017 IN IP4 x.x.x.x
s=session
c=IN IP4 x.x.x.x
t=0 0
m=audio 11038 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 10436 RTP/AVP
835.3e
 to x.x.x.x:5060
Retransmitting #3 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.x.x:5060
From: <sip:002125095690 at x.x.x.x>;tag=839C4028-E83
To: 8005095690 <sip:8005095690 at x.x.x.x:5060;user=phone>;tag=as6e87bf67
Call-ID: 8780-3280658280-475041 at x.x.x.x
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8005095690 at x.x.x.x>
Content-Type: application/sdp
Content-Length: 260
  
v=0
o=root 9017 9017 IN IP4 x.x.x.x
s=session
c=IN IP4 x.x.x.x
t=0 0
m=audio 11038 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 10436 RTP/AVP
835.3e
 to x.x.x.x:5060
Retransmitting #4 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.x.x:5060
From: <sip:002125095690 at x.x.x.x>;tag=839C4028-E83
To: 8005095690 <sip:8005095690 at x.x.x.x:5060;user=phone>;tag=as6e87bf67
Call-ID: 8780-3280658280-475041 at x.x.x.x
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8005095690 at x.x.x.x>
Content-Type: application/sdp
Content-Length: 260
  
v=0
o=root 9017 9017 IN IP4 x.x.x.x
s=session
c=IN IP4 x.x.x.x
t=0 0
m=audio 11038 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 10436 RTP/AVP
835.3e
 to x.x.x.x:5060
Retransmitting #5 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.x.x:5060
From: <sip:002125095690 at x.x.x.x>;tag=839C4028-E83
To: 8005095690 <sip:8005095690 at x.x.x.x:5060;user=phone>;tag=as6e87bf67
Call-ID: 8780-3280658280-475041 at x.x.x.x
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8005095690 at x.x.x.x>
Content-Type: application/sdp
Content-Length: 260
  
v=0
o=root 9017 9017 IN IP4 x.x.x.x
s=session
c=IN IP4 x.x.x.x
t=0 0
m=audio 11038 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 10436 RTP/AVP
835.3e
 to x.x.x.x:5060

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