

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '55ae66469f7f@24.123.23.170' Method: REGISTER
set_destination: Parsing <sip:528@192.168.1.93:5060> for address/port to send to
set_destination: set destination to 192.168.1.93, port 5060
Audio is at 24.123.23.170 port 17286
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.93:5060:
INVITE sip:528@192.168.1.93:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK42c538af
From: <sip:526@24.123.23.170>;tag=as68aa18d6
To: 528 <sip:528@192.168.1.93:5060>;tag=90493555b415b8be1762433957f5a495
Contact: <sip:526@24.123.23.170>
Call-ID: 079076751e0b5ba32413ad190c372a4a@24.123.23.170
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 3998 4000 IN IP4 24.123.23.170
s=session
c=IN IP4 24.123.23.170
t=0 0
m=audio 17286 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=recvonly

---
Scheduling destruction of SIP dialog '079076751e0b5ba32413ad190c372a4a@24.123.23.170' in 32000 ms (Method: ACK)
  == Spawn extension (smvoice-sip, 204, 6) exited non-zero on 'SIP/526-b7720560'
unifiedpaging*CLI>
<--- SIP read from 192.168.1.93:5060 --->
SIP/2.0 200 OK
To: 528 <sip:528@192.168.1.93:5060>;tag=90493555b415b8be1762433957f5a495
Contact: <sip:528@192.168.1.93:5060>
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK42c538af
From: <sip:526@24.123.23.170>;tag=as68aa18d6
Call-ID: 079076751e0b5ba32413ad190c372a4a@24.123.23.170
CSeq: 103 INVITE
Content-Length: 241
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK

v=0
o=- 3998 103875 IN IP4 192.168.1.93
s=session
c=IN IP4 0.0.0.0
t=0 0
m=audio 30028 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16,17,18
a=silenceSupp:off - - - -
a=ptime:20
a=sendonly

<------------->
--- (10 headers 12 lines) ---
set_destination: Parsing <sip:528@192.168.1.93:5060> for address/port to send to
set_destination: set destination to 192.168.1.93, port 5060
Transmitting (no NAT) to 192.168.1.93:5060:
ACK sip:528@192.168.1.93:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK2668c44b
From: <sip:526@24.123.23.170>;tag=as68aa18d6
To: 528 <sip:528@192.168.1.93:5060>;tag=90493555b415b8be1762433957f5a495
Contact: <sip:526@24.123.23.170>
Call-ID: 079076751e0b5ba32413ad190c372a4a@24.123.23.170
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:528@192.168.1.93:5060> for address/port to send to
set_destination: set destination to 192.168.1.93, port 5060
Reliably Transmitting (no NAT) to 192.168.1.93:5060:
BYE sip:528@192.168.1.93:5060 SIP/2.0
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK2b453c9f
From: <sip:526@24.123.23.170>;tag=as68aa18d6
To: 528 <sip:528@192.168.1.93:5060>;tag=90493555b415b8be1762433957f5a495
Call-ID: 079076751e0b5ba32413ad190c372a4a@24.123.23.170
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of SIP dialog '079076751e0b5ba32413ad190c372a4a@24.123.23.170' in 32000 ms (Method: ACK)
Retransmitting #3 (no NAT) to 192.168.1.93:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKx61d03b960da7358c2333efdaf0622dbe;received=192.168.1.93
From: 528 <sip:528@192.168.1.93:5060>;tag=f055e63c670dfe04684b84461a029087
To: <sip:550@24.123.23.170>;tag=as68237d31
Call-ID: 386a40f7388a7a353e329d8124252dda@24.123.23.170
CSeq: 465812 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:550@24.123.23.170>
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 3998 4000 IN IP4 192.168.1.91
s=session
c=IN IP4 192.168.1.91
t=0 0
m=audio 10010 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
unifiedpaging*CLI>
<--- SIP read from 192.168.1.93:5060 --->
SIP/2.0 200 OK
To: 528 <sip:528@192.168.1.93:5060>;tag=90493555b415b8be1762433957f5a495
Via: SIP/2.0/UDP 24.123.23.170:5060;branch=z9hG4bK2b453c9f
From: <sip:526@24.123.23.170>;tag=as68aa18d6
Call-ID: 079076751e0b5ba32413ad190c372a4a@24.123.23.170
CSeq: 104 BYE
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '079076751e0b5ba32413ad190c372a4a@24.123.23.170' Method: ACK
Retransmitting #4 (no NAT) to 192.168.1.93:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKx61d03b960da7358c2333efdaf0622dbe;received=192.168.1.93
From: 528 <sip:528@192.168.1.93:5060>;tag=f055e63c670dfe04684b84461a029087
To: <sip:550@24.123.23.170>;tag=as68237d31
Call-ID: 386a40f7388a7a353e329d8124252dda@24.123.23.170
CSeq: 465812 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:550@24.123.23.170>
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 3998 4000 IN IP4 192.168.1.91
s=session
c=IN IP4 192.168.1.91
t=0 0
m=audio 10010 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #5 (no NAT) to 192.168.1.93:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKx61d03b960da7358c2333efdaf0622dbe;received=192.168.1.93
From: 528 <sip:528@192.168.1.93:5060>;tag=f055e63c670dfe04684b84461a029087
To: <sip:550@24.123.23.170>;tag=as68237d31
Call-ID: 386a40f7388a7a353e329d8124252dda@24.123.23.170
CSeq: 465812 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:550@24.123.23.170>
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 3998 4000 IN IP4 192.168.1.91
s=session
c=IN IP4 192.168.1.91
t=0 0
m=audio 10010 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Feb 15 16:41:49] NOTICE[4029]: chan_sip.c:7055 sip_reregister:    -- Re-registration for  3173241052@sip.broadvoice.com
[Feb 15 16:41:49] NOTICE[4029]: chan_sip.c:11961 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s)
Retransmitting #6 (no NAT) to 192.168.1.93:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKx61d03b960da7358c2333efdaf0622dbe;received=192.168.1.93
From: 528 <sip:528@192.168.1.93:5060>;tag=f055e63c670dfe04684b84461a029087
To: <sip:550@24.123.23.170>;tag=as68237d31
Call-ID: 386a40f7388a7a353e329d8124252dda@24.123.23.170
CSeq: 465812 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:550@24.123.23.170>
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 3998 4000 IN IP4 192.168.1.91
s=session
c=IN IP4 192.168.1.91
t=0 0
m=audio 10010 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Feb 15 16:41:54] WARNING[4029]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission 386a40f7388a7a353e329d8124252dda@24.123.23.170 for seqno 465812 (Critical Response)
[Feb 15 16:41:54] WARNING[4029]: chan_sip.c:1898 retrans_pkt: Hanging up call 386a40f7388a7a353e329d8124252dda@24.123.23.170 - no reply to our critical packet.
  == Spawn extension (smvoice-sip, 204, 6) exited non-zero on 'SIP/550-b7734378'
Really destroying SIP dialog '386a40f7388a7a353e329d8124252dda@24.123.23.170' Method: ACK
unifiedpaging*CLI>
<--- SIP read from 192.168.1.93:5060 --->
BYE sip:550@24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKvdfc86f6d4ca22066c8a692b67e2c3c12
Call-ID: 386a40f7388a7a353e329d8124252dda@24.123.23.170
CSeq: 465813 BYE
From: 528 <sip:528@192.168.1.93:5060>;tag=f055e63c670dfe04684b84461a029087
To: <sip:550@24.123.23.170>;tag=as68237d31
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---

<--- Transmitting (no NAT) to 192.168.1.93:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKvdfc86f6d4ca22066c8a692b67e2c3c12;received=192.168.1.93
From: 528 <sip:528@192.168.1.93:5060>;tag=f055e63c670dfe04684b84461a029087
To: <sip:550@24.123.23.170>;tag=as68237d31
Call-ID: 386a40f7388a7a353e329d8124252dda@24.123.23.170
CSeq: 465813 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
unifiedpaging*CLI>
<--- SIP read from 192.168.1.93:5060 --->
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKd79f74b3cb3cbbf05a003f0e07299adab
Call-ID: 55ae66469f7f@24.123.23.170
CSeq: 37380 REGISTER
From: <sip:528@24.123.23.170>;tag=94d2aaa50f89390af2cf3ace3ef31f76
To: <sip:528@24.123.23.170>
Contact: <sip:528@192.168.1.93:5060>;expires=60;q=0.50
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Authorization: Digest realm="asterisk",   nonce="7273933d",   algorithm=MD5,   uri="sip:24.123.23.170",   username="528",   response="b81b38903c6cb734d1cb18f7e251e60a"


<------------->
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.93 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.1.93:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKd79f74b3cb3cbbf05a003f0e07299adab;received=192.168.1.93
From: <sip:528@24.123.23.170>;tag=94d2aaa50f89390af2cf3ace3ef31f76
To: <sip:528@24.123.23.170>
Call-ID: 55ae66469f7f@24.123.23.170
CSeq: 37380 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:528@24.123.23.170>
Content-Length: 0


<------------>
unifiedpaging*CLI>
<--- Transmitting (no NAT) to 192.168.1.93:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKd79f74b3cb3cbbf05a003f0e07299adab;received=192.168.1.93
From: <sip:528@24.123.23.170>;tag=94d2aaa50f89390af2cf3ace3ef31f76
To: <sip:528@24.123.23.170>;tag=as0e0dd9da
Call-ID: 55ae66469f7f@24.123.23.170
CSeq: 37380 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="21a1c665"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '55ae66469f7f@24.123.23.170' in 32000 ms (Method: REGISTER)
unifiedpaging*CLI>
<--- SIP read from 192.168.1.93:5060 --->
REGISTER sip:24.123.23.170 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKua02f3e059cfd16f0ca668dfc24bc03f6
CSeq: 37381 REGISTER
Call-ID: 55ae66469f7f@24.123.23.170
From: <sip:528@24.123.23.170>;tag=94d2aaa50f89390af2cf3ace3ef31f76
To: <sip:528@24.123.23.170>
Contact: <sip:528@192.168.1.93:5060>;expires=60;q=0.50
User-Agent: Uniden SIP Phone p2 Ver BS4.77
Max-Forwards: 70
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK
Authorization: Digest realm="asterisk",   nonce="21a1c665",   algorithm=MD5,   uri="sip:24.123.23.170",   username="528",   response="5606d9374716cefc019277ae5e024a07"


<------------->
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.93 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.1.93:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKua02f3e059cfd16f0ca668dfc24bc03f6;received=192.168.1.93
From: <sip:528@24.123.23.170>;tag=94d2aaa50f89390af2cf3ace3ef31f76
To: <sip:528@24.123.23.170>
Call-ID: 55ae66469f7f@24.123.23.170
CSeq: 37381 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:528@24.123.23.170>
Content-Length: 0


<------------>
unifiedpaging*CLI>
<--- Transmitting (no NAT) to 192.168.1.93:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bKua02f3e059cfd16f0ca668dfc24bc03f6;received=192.168.1.93
From: <sip:528@24.123.23.170>;tag=94d2aaa50f89390af2cf3ace3ef31f76
To: <sip:528@24.123.23.170>;tag=as0e0dd9da
Call-ID: 55ae66469f7f@24.123.23.170
CSeq: 37381 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: <sip:528@192.168.1.93:5060>;expires=60
Date: Thu, 15 Feb 2007 21:42:05 GMT
Content-Length: 0