Installation Asterisk avec IVR pour une preuve de concept

**survivant** · 05/11/2010, 15h08

suite #2

Code:

---
    -- Called BabyTEL/5141111234
[Nov  3 20:09:58] NOTICE[2625]: chan_sip.c:12147 sip_reregister:    -- Re-registration for  1XXXXXXXXXX@216.18.125.12
       > doing dnsmgr_lookup for '216.18.125.12'
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 216.18.125.12:5065:
REGISTER sip:216.18.125.12:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.130:15060;branch=z9hG4bK16b51376
Max-Forwards: 70
From: <sip:1XXXXXXXXXX@216.18.125.12>;tag=as79b6ed58
To: <sip:1XXXXXXXXXX@216.18.125.12>
Call-ID: 121f677f2238fd64085b6dec58a69206@127.0.1.1
CSeq: 112 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r293530
Authorization: Digest username="1XXXXXXXXXX", realm="sip.babytel.ca", algorithm=MD5, uri="sip:216.18.125.12:5065", nonce="4cd33f8b180931c526ea05e41a9b5130ef068253", response="d951d801b9e4ec0d9315fc5827aa4f8b"
Expires: 120
Contact: <sip:s@192.168.10.130:15060>
Content-Length: 0


---
Retransmitting #1 (NAT) to 216.18.125.12:5060:
INVITE sip:15141111234@216.18.125.12:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.130:15060;branch=z9hG4bK190b332f;rport
Max-Forwards: 70
From: "SoftPhone" <sip:1001@216.18.125.12>;tag=as63526dd5
To: <sip:5141111234@216.18.125.12:5065>
Contact: <sip:1001@192.168.10.130:15060>
Call-ID: 64c18d840ecdc7f124098b426c7cfd08@216.18.125.12
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r293530
Date: Thu, 04 Nov 2010 03:09:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 1215646287 1215646287 IN IP4 192.168.10.130
s=Asterisk PBX SVN-branch-1.8-r293530
c=IN IP4 192.168.10.130
t=0 0
m=audio 7018 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #2 (NAT) to 216.18.125.12:5060:
INVITE sip:15141111234@216.18.125.12:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.130:15060;branch=z9hG4bK190b332f;rport
Max-Forwards: 70
From: "SoftPhone" <sip:1001@216.18.125.12>;tag=as63526dd5
To: <sip:5141111234@216.18.125.12:5065>
Contact: <sip:1001@192.168.10.130:15060>
Call-ID: 64c18d840ecdc7f124098b426c7cfd08@216.18.125.12
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r293530
Date: Thu, 04 Nov 2010 03:09:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 1215646287 1215646287 IN IP4 192.168.10.130
s=Asterisk PBX SVN-branch-1.8-r293530
c=IN IP4 192.168.10.130
t=0 0
m=audio 7018 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:216.18.125.12:5065 --->
SIP/2.0 200 OK
To: <sip:1XXXXXXXXXX@216.18.125.12>;tag=115ba56c
From: <sip:1XXXXXXXXXX@216.18.125.12>;tag=as79b6ed58
Via: SIP/2.0/UDP 192.168.10.130:15060;branch=z9hG4bK16b51376
Call-ID: 121f677f2238fd64085b6dec58a69206@127.0.1.1
CSeq: 112 REGISTER
Contact: <sip:s@192.168.10.130:15060>;expires=41
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog '121f677f2238fd64085b6dec58a69206@127.0.1.1' in 32000 ms (Method: REGISTER)
[Nov  3 20:09:59] NOTICE[2625]: chan_sip.c:19496 handle_response_register: Outbound Registration: Expiry for 216.18.125.12 is 41 sec (Scheduling reregistration in 26 s)
Retransmitting #3 (NAT) to 216.18.125.12:5060:
INVITE sip:5141111234@216.18.125.12:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.130:15060;branch=z9hG4bK190b332f;rport
Max-Forwards: 70
From: "SoftPhone" <sip:1001@216.18.125.12>;tag=as63526dd5
To: <sip:5141111234@216.18.125.12:5065>
Contact: <sip:1001@192.168.10.130:15060>
Call-ID: 64c18d840ecdc7f124098b426c7cfd08@216.18.125.12
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r293530
Date: Thu, 04 Nov 2010 03:09:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 1215646287 1215646287 IN IP4 192.168.10.130
s=Asterisk PBX SVN-branch-1.8-r293530
c=IN IP4 192.168.10.130
t=0 0
m=audio 7018 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #4 (NAT) to 216.18.125.12:5060:
INVITE sip:15141111234@216.18.125.12:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.130:15060;branch=z9hG4bK190b332f;rport
Max-Forwards: 70
From: "SoftPhone" <sip:1001@216.18.125.12>;tag=as63526dd5
To: <sip:15141111234@216.18.125.12:5065>
Contact: <sip:1001@192.168.10.130:15060>
Call-ID: 64c18d840ecdc7f124098b426c7cfd08@216.18.125.12
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r293530
Date: Thu, 04 Nov 2010 03:09:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 1215646287 1215646287 IN IP4 192.168.10.130
s=Asterisk PBX SVN-branch-1.8-r293530
c=IN IP4 192.168.10.130
t=0 0
m=audio 7018 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #5 (NAT) to 216.18.125.12:5060:
INVITE sip:15141111234@216.18.125.12:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.130:15060;branch=z9hG4bK190b332f;rport
Max-Forwards: 70
From: "SoftPhone" <sip:1001@216.18.125.12>;tag=as63526dd5
To: <sip:15141111234@216.18.125.12:5065>
Contact: <sip:1001@192.168.10.130:15060>
Call-ID: 64c18d840ecdc7f124098b426c7cfd08@216.18.125.12
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r293530
Date: Thu, 04 Nov 2010 03:09:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 1215646287 1215646287 IN IP4 192.168.10.130
s=Asterisk PBX SVN-branch-1.8-r293530
c=IN IP4 192.168.10.130
t=0 0
m=audio 7018 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #6 (NAT) to 216.18.125.12:5060:
INVITE sip:15141111234@216.18.125.12:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.130:15060;branch=z9hG4bK190b332f;rport
Max-Forwards: 70
From: "SoftPhone" <sip:1001@216.18.125.12>;tag=as63526dd5
To: <sip:5141111234@216.18.125.12:5065>
Contact: <sip:1001@192.168.10.130:15060>
Call-ID: 64c18d840ecdc7f124098b426c7cfd08@216.18.125.12
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r293530
Date: Thu, 04 Nov 2010 03:09:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 1215646287 1215646287 IN IP4 192.168.10.130
s=Asterisk PBX SVN-branch-1.8-r293530
c=IN IP4 192.168.10.130
t=0 0
m=audio 7018 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Nov  3 20:10:05] WARNING[2625]: chan_sip.c:3383 retrans_pkt: Retransmission timeout reached on transmission 64c18d840ecdc7f124098b426c7cfd08@216.18.125.12 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt.

Packet timed out after 6400ms with no response
[Nov  3 20:10:05] WARNING[2625]: chan_sip.c:3412 retrans_pkt: Hanging up call 64c18d840ecdc7f124098b426c7cfd08@216.18.125.12 - no reply to our critical packet (see doc/sip-retransmit.txt).

  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [915141111234@Interne:3] Hangup("SIP/1001-00000019", "") in new stack
  == Spawn extension (Interne, 915141111234, 3) exited non-zero on 'SIP/1001-00000019'
Scheduling destruction of SIP dialog '5259' in 6400 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to 192.168.10.1:6666 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.0.17:6666;branch=z9hG4bK23771;received=192.168.10.1;rport=6666
From: <sip:1001@192.168.10.130:15060>;tag=30407
To: <sip:1915141111234@192.168.10.130:15060>;tag=as3a6cee68
Call-ID: 5259
CSeq: 21 INVITE
Server: Asterisk PBX SVN-branch-1.8-r293530
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

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