rendered paste bodyNon-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 89.35.79.69:56404
Looking for 0748881658 in numberplan-custom-1 (domain voip.telefonet.ro)
list_route: hop: <sip:23003@89.35.79.69:61286>
voip*CLI>
<--- Transmitting (no NAT) to 89.35.79.69:61286 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 89.35.79.69:61286;branch=z9hG4bK-d87543-41418932c6173f26-1--d87543-;received=89.35.79.69;rport=61286
From: "Znuff"<sip:23003@voip.telefonet.ro>;tag=856ae020
To: "0748881658"<sip:0748881658@voip.telefonet.ro>
Call-ID: ZjhkNjdlYTFiN2NkNWNmNDJhZTgxMzZiYWJkZDEwMTU.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:0748881658@89.35.79.5>
Content-Length: 0
<------------>
-- Executing [0748881658@numberplan-custom-1:1] Dial("SIP/23003-b5a1c108", "SIP/0748881658@10.0.0.1") in new stack
Audio is at 10.0.0.2 port 15592
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.1:5060:
INVITE sip:0748881658@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK3542a8a3;rport
From: "Znuff" <sip:23003@10.0.0.2>;tag=as69b93c0a
To: <sip:0748881658@10.0.0.1>
Contact: <sip:23003@10.0.0.2>
Call-ID: 565304bb089b4fff4719b03c311ec869@10.0.0.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 05 Mar 2007 04:21:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 236
v=0
o=root 8856 8856 IN IP4 89.35.79.69
s=session
c=IN IP4 89.35.79.69
t=0 0
m=audio 56404 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 0748881658@10.0.0.1
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
SIP/2.0 100 Trying
From: "Znuff"<sip:23003@10.0.0.2>;tag=as69b93c0a
To: <sip:0748881658@10.0.0.1>;tag=a000001-13c4-2413-8d07c6-36c1
Call-ID: 565304bb089b4fff4719b03c311ec869@10.0.0.2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.2:5060;received=10.0.0.2;branch=z9hG4bK3542a8a3;rport
Contact: <sip:0748881658@10.0.0.1>
Content-Length:0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'XfLu8uaWvRwQUWiM@89.35.79.247' Method: REGISTER
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
SIP/2.0 415 Unsupported Media Type
From: "Znuff"<sip:23003@10.0.0.2>;tag=as69b93c0a
To: <sip:0748881658@10.0.0.1>;tag=a000001-13c4-2413-8d07c6-36c1
Call-ID: 565304bb089b4fff4719b03c311ec869@10.0.0.2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.2:5060;received=10.0.0.2;branch=z9hG4bK3542a8a3;rport
Content-Length:0
<------------->
--- (7 headers 0 lines) ---
-- Got SIP response 415 "Unsupported Media Type" back from 10.0.0.1
Transmitting (no NAT) to 10.0.0.1:5060:
ACK sip:0748881658@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK3542a8a3;rport
From: "Znuff" <sip:23003@10.0.0.2>;tag=as69b93c0a
To: <sip:0748881658@10.0.0.1>;tag=a000001-13c4-2413-8d07c6-36c1
Contact: <sip:23003@10.0.0.2>
Call-ID: 565304bb089b4fff4719b03c311ec869@10.0.0.2
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [0748881658@numberplan-custom-1:2] Congestion("SIP/23003-b5a1c108", "") in new stack
voip*CLI>
<--- Transmitting (no NAT) to 89.35.79.69:61286 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 89.35.79.69:61286;branch=z9hG4bK-d87543-41418932c6173f26-1--d87543-;received=89.35.79.69;rport=61286
From: "Znuff"<sip:23003@voip.telefonet.ro>;tag=856ae020
To: "0748881658"<sip:0748881658@voip.telefonet.ro>;tag=as6f87b067
Call-ID: ZjhkNjdlYTFiN2NkNWNmNDJhZTgxMzZiYWJkZDEwMTU.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:0748881658@89.35.79.5>
Content-Length: 0
X-Asterisk-HangupCause: Interworking, unspecified
X-Asterisk-HangupCauseCode: 127
<------------>
== Spawn extension (numberplan-custom-1, 0748881658, 2) exited non-zero on 'SIP/23003-b5a1c108'
voip*CLI>
<--- SIP read from 89.35.79.69:61286 --->
ACK sip:0748881658@voip.telefonet.ro SIP/2.0
Via: SIP/2.0/UDP 89.35.79.69:61286;branch=z9hG4bK-d87543-41418932c6173f26-1--d87543-;rport
To: "0748881658"<sip:0748881658@voip.telefonet.ro>;tag=as6f87b067
From: "Znuff"<sip:23003@voip.telefonet.ro>;tag=856ae020
Call-ID: ZjhkNjdlYTFiN2NkNWNmNDJhZTgxMzZiYWJkZDEwMTU.
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '565304bb089b4fff4719b03c311ec869@10.0.0.2' Method: INVITE
Really destroying SIP dialog 'ZjhkNjdlYTFiN2NkNWNmNDJhZTgxMzZiYWJkZDEwMTU.' Method: ACK
voip*CLI>
<--- SIP read from 89.35.79.69:61286 --->
<------------->
--- (0 headers 1 lines) ---
Really destroying SIP dialog 'owiz95Hz9XJxvP8b@89.35.79.94' Method: REGISTER
voip*CLI>
<--- SIP read from 89.35.79.247:5060 --->
REGISTER sip:voip.telefonet.ro SIP/2.0
Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKIzCib2opx
Max-Forwards: 70
User-Agent: Wu Chuan
From: "23001" <sip:23001@voip.telefonet.ro>;tag=hOC1H8UNFOCvPuLu
To: "23001" <sip:23001@voip.telefonet.ro>
Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247
CSeq: 9806 REGISTER
Contact: <sip:23001@89.35.79.247:5060>
Expires: 60
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 89.35.79.247 : 5060 (no NAT)
voip*CLI>
<--- Transmitting (no NAT) to 89.35.79.247:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKIzCib2opx;received=89.35.79.247
From: "23001" <sip:23001@voip.telefonet.ro>;tag=hOC1H8UNFOCvPuLu
To: "23001" <sip:23001@voip.telefonet.ro>
Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247
CSeq: 9806 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:23001@89.35.79.5>
Content-Length: 0
<------------>
voip*CLI>
<--- Transmitting (no NAT) to 89.35.79.247:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKIzCib2opx;received=89.35.79.247
From: "23001" <sip:23001@voip.telefonet.ro>;tag=hOC1H8UNFOCvPuLu
To: "23001" <sip:23001@voip.telefonet.ro>;tag=as70d42adb
Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247
CSeq: 9806 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="voip.telefonet.ro", nonce="31195ebf"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'XfLu8uaWvRwQUWiM@89.35.79.247' in 32000 ms (Method: REGISTER)
voip*CLI>
<--- SIP read from 89.35.79.247:5060 --->
REGISTER sip:voip.telefonet.ro SIP/2.0
Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKdNux7SGxR
Max-Forwards: 70
User-Agent: Wu Chuan
From: "23001" <sip:23001@voip.telefonet.ro>;tag=YhLcODA3FkkSvvdi
To: "23001" <sip:23001@voip.telefonet.ro>
Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247
CSeq: 9807 REGISTER
Contact: <sip:23001@89.35.79.247:5060>
Expires: 60
Authorization: Digest username="23001", realm="voip.telefonet.ro", nonce="31195ebf", uri="sip:voip.telefonet.ro", response="b7406079f2061834b4245da55f0a0a26"
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 89.35.79.247 : 5060 (no NAT)
voip*CLI>
<--- Transmitting (no NAT) to 89.35.79.247:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKdNux7SGxR;received=89.35.79.247
From: "23001" <sip:23001@voip.telefonet.ro>;tag=YhLcODA3FkkSvvdi
To: "23001" <sip:23001@voip.telefonet.ro>
Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247
CSeq: 9807 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:23001@89.35.79.5>
Content-Length: 0
<------------>
voip*CLI>
<--- Transmitting (no NAT) to 89.35.79.247:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.35.79.247:5060;branch=z9hG4bKdNux7SGxR;received=89.35.79.247
From: "23001" <sip:23001@voip.telefonet.ro>;tag=YhLcODA3FkkSvvdi
To: "23001" <sip:23001@voip.telefonet.ro>;tag=as70d42adb
Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247
CSeq: 9807 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: <sip:23001@89.35.79.247:5060>;expires=60
Date: Mon, 05 Mar 2007 04:21:47 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'XfLu8uaWvRwQUWiM@89.35.79.247' in 32000 ms (Method: REGISTER)
voip*CLI>