rendered paste bodyPeer audio RTP is at port 89.35.79.69:30498
Found description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 89.35.79.69:30498
list_route: hop: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>
set_destination: Parsing <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab> for address/port to send to
set_destination: set destination to 89.35.79.69, port 16088
Transmitting (no NAT) to 89.35.79.69:16088:
ACK sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab SIP/2.0
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK3d73da63;rport
From: "ORAS" <sip:748881658@89.35.79.5>;tag=as5e1926f2
To: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>;tag=1b3d7009
Contact: <sip:748881658@89.35.79.5>
Call-ID: 693ae16d42b918627a4ad5da07bb9361@89.35.79.5
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- Call on SIP/23003-0823b6c0 left from hold
-- SIP/23003-0823b6c0 answered SIP/5060-0822ef78
Audio is at 10.0.0.2 port 11524
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 10.0.0.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-163db-56e2fa4-3108;received=10.0.0.1
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-163db-56e2fa4-56ab
To: <sip:23003@10.0.0.2>;tag=as6db5b353
Call-ID: 97949eec-a000001-13c4-163db-56e2fa4-1725@10.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:23003@10.0.0.2>
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 24756 24756 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 11524 RTP/AVP 8 0 18 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
-- Packet2Packet bridging SIP/5060-0822ef78 and SIP/23003-0823b6c0
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
ACK sip:23003@10.0.0.2 SIP/2.0
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-163db-56e2fa4-56ab
To: <sip:23003@10.0.0.2>;tag=as6db5b353
Call-ID: 97949eec-a000001-13c4-163db-56e2fa4-1725@10.0.0.1
CSeq: 1 ACK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-163e3-56e4e72-a1
Max-Forwards: 70
Contact: <sip:748881658@10.0.0.1:5060;transport=UDP>
Content-Length:0
<------------->
--- (9 headers 0 lines) ---
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
BYE sip:23003@10.0.0.2;TCID-29 SIP/2.0
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-163db-56e2fa4-56ab
To: <sip:23003@10.0.0.2>;tag=as6db5b353
Call-ID: 97949eec-a000001-13c4-163db-56e2fa4-1725@10.0.0.1
CSeq: 2 BYE
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-163e3-56e4e7a-3287
Max-Forwards: 70
Content-Length:0
<------------->
--- (8 headers 0 lines) ---
Sending to 10.0.0.1 : 5060 (NAT)
<--- Transmitting (NAT) to 10.0.0.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-163e3-56e4e7a-3287;received=10.0.0.1
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-163db-56e2fa4-56ab
To: <sip:23003@10.0.0.2>;tag=as6db5b353
Call-ID: 97949eec-a000001-13c4-163db-56e2fa4-1725@10.0.0.1
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:23003@10.0.0.2>
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '693ae16d42b918627a4ad5da07bb9361@89.35.79.5' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab> for address/port to send to
set_destination: set destination to 89.35.79.69, port 16088
Reliably Transmitting (no NAT) to 89.35.79.69:16088:
BYE sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab SIP/2.0
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK55851551;rport
From: "ORAS" <sip:748881658@89.35.79.5>;tag=as5e1926f2
To: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>;tag=1b3d7009
Call-ID: 693ae16d42b918627a4ad5da07bb9361@89.35.79.5
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (default, 23003, 1) exited non-zero on 'SIP/5060-0822ef78'
voip*CLI>
<--- SIP read from 89.35.79.69:16088 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.35.79.5:5060;branch=z9hG4bK55851551;rport=5060
Contact: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>
To: <sip:23003@89.35.79.69:16088;rinstance=7ed7b782ad482eab>;tag=1b3d7009
From: "ORAS"<sip:748881658@89.35.79.5>;tag=as5e1926f2
Call-ID: 693ae16d42b918627a4ad5da07bb9361@89.35.79.5
CSeq: 103 BYE
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '693ae16d42b918627a4ad5da07bb9361@89.35.79.5' Method: INVITE
Really destroying SIP dialog '97949eec-a000001-13c4-163db-56e2fa4-1725@10.0.0.1' Method: BYE
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=z9hG4bKt8463hHZe
Max-Forwards: 70
User-Agent: Wu Chuan
From: "23001" <sip:23001@voip.telefonet.ro>;tag=VWtuxzrHvah1XTdZ
To: "23001" <sip:23001@voip.telefonet.ro>
Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247
CSeq: 10990 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)
<--- 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=z9hG4bKt8463hHZe;received=89.35.79.247
From: "23001" <sip:23001@voip.telefonet.ro>;tag=VWtuxzrHvah1XTdZ
To: "23001" <sip:23001@voip.telefonet.ro>
Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247
CSeq: 10990 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
<------------>
<--- 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=z9hG4bKt8463hHZe;received=89.35.79.247
From: "23001" <sip:23001@voip.telefonet.ro>;tag=VWtuxzrHvah1XTdZ
To: "23001" <sip:23001@voip.telefonet.ro>;tag=as4f469b8f
Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247
CSeq: 10990 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="4476e144"
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=z9hG4bKvTVL4kPSj
Max-Forwards: 70
User-Agent: Wu Chuan
From: "23001" <sip:23001@voip.telefonet.ro>;tag=vZUv2gTxnjXNfR5q
To: "23001" <sip:23001@voip.telefonet.ro>
Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247
CSeq: 10991 REGISTER
Contact: <sip:23001@89.35.79.247:5060>
Expires: 60
Authorization: Digest username="23001", realm="voip.telefonet.ro", nonce="4476e144", uri="sip:voip.telefonet.ro", response="5f9e1f6376d57706d89409a92d1754fc"
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 89.35.79.247 : 5060 (no NAT)
<--- 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=z9hG4bKvTVL4kPSj;received=89.35.79.247
From: "23001" <sip:23001@voip.telefonet.ro>;tag=vZUv2gTxnjXNfR5q
To: "23001" <sip:23001@voip.telefonet.ro>
Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247
CSeq: 10991 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=z9hG4bKvTVL4kPSj;received=89.35.79.247
From: "23001" <sip:23001@voip.telefonet.ro>;tag=vZUv2gTxnjXNfR5q
To: "23001" <sip:23001@voip.telefonet.ro>;tag=as4f469b8f
Call-ID: XfLu8uaWvRwQUWiM@89.35.79.247
CSeq: 10991 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: Tue, 06 Mar 2007 03:05:54 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'XfLu8uaWvRwQUWiM@89.35.79.247' in 32000 ms (Method: REGISTER)
voip*CLI>