rendered paste bodylocalhost*CLI>
<--- SIP read from 189.180.217.155:55977 --->
INVITE sip:0016044690084@sip.telavip.com.mx SIP/2.0
Via: SIP/2.0/UDP 189.180.217.155:55828;branch=z9hG4bK-d8754z-add2164cc95d1441-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:5512047763@189.180.217.155:55828>
To: <sip:0016044690084@sip.telavip.com.mx>
From: <sip:5512047763@sip.telavip.com.mx>;transport=UDP;tag=d813117c
Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Type: application/sdp
User-Agent: Zoiper rev.417
Content-Length: 250
v=0
o= 0 0 IN IP4 189.180.217.155
s=
c=IN IP4 189.180.217.155
t=0 0
m=audio 56038 RTP/AVP 3 110 97 101
a=fmtp:97 mode=30
a=fmtp:101 0-15
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
<------------->
--- (12 headers 12 lines) ---
Sending to 189.180.217.155 : 55977 (NAT)
Using INVITE request as basis request - ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk.
<--- Reliably Transmitting (no NAT) to 189.180.217.155:55828 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 189.180.217.155:55828;branch=z9hG4bK-d8754z-add2164cc95d1441-1---d8754z-;rport;received=189.180.217.155
From: <sip:5512047763@sip.telavip.com.mx>;transport=UDP;tag=d813117c
To: <sip:0016044690084@sip.telavip.com.mx>;tag=as3066808d
Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7c389d18"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk.' in 32000 ms (Method: INVITE)
Found user '5512047763'
localhost*CLI>
<--- SIP read from 189.180.217.155:55977 --->
ACK sip:0016044690084@sip.telavip.com.mx SIP/2.0
Via: SIP/2.0/UDP 189.180.217.155:55828;branch=z9hG4bK-d8754z-add2164cc95d1441-1---d8754z-;rport
To: <sip:0016044690084@sip.telavip.com.mx>;tag=as3066808d
From: <sip:5512047763@sip.telavip.com.mx>;transport=UDP;tag=d813117c
Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk.
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
localhost*CLI>
<--- SIP read from 189.180.217.155:55977 --->
INVITE sip:0016044690084@sip.telavip.com.mx SIP/2.0
Via: SIP/2.0/UDP 189.180.217.155:55828;branch=z9hG4bK-d8754z-4fe4cc06ff31530d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:5512047763@189.180.217.155:55828>
To: <sip:0016044690084@sip.telavip.com.mx>
From: <sip:5512047763@sip.telavip.com.mx>;transport=UDP;tag=d813117c
Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Type: application/sdp
Proxy-Authorization: Digest username="5512047763",realm="asterisk",nonce="7c389d18",uri="sip:0016044690084@sip.telavip.com.mx",response="c4315cb52245bc9c43111d2b11281f12",algorithm=MD5
User-Agent: Zoiper rev.417
Content-Length: 250
v=0
o= 0 0 IN IP4 189.180.217.155
s=
c=IN IP4 189.180.217.155
t=0 0
m=audio 56038 RTP/AVP 3 110 97 101
a=fmtp:97 mode=30
a=fmtp:101 0-15
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
<------------->
--- (13 headers 12 lines) ---
Sending to 189.180.217.155 : 55977 (NAT)
Using INVITE request as basis request - ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk.
Found user '5512047763'
Found RTP audio format 3
Found RTP audio format 110
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 189.180.217.155:56038
Found description format GSM for ID 3
Found description format speex for ID 110
Found description format iLBC for ID 97
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x602 (gsm|speex|ilbc)/video=0x0 (nothing), combined - 0x2 (gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 189.180.217.155:56038
Looking for 0016044690084 in salida-df (domain sip.telavip.com.mx)
list_route: hop: <sip:5512047763@189.180.217.155:55828>
<--- Transmitting (no NAT) to 189.180.217.155:55828 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 189.180.217.155:55828;branch=z9hG4bK-d8754z-4fe4cc06ff31530d-1---d8754z-;rport;received=189.180.217.155
From: <sip:5512047763@sip.telavip.com.mx>;transport=UDP;tag=d813117c
To: <sip:0016044690084@sip.telavip.com.mx>
Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:0016044690084@200.76.98.137>
Content-Length: 0
<------------>
-- Executing [0016044690084@salida-df:1] Goto("SIP/5512047763-08b1b2b0", "salida-internacional|0016044690084|1") in new stack
-- Goto (salida-internacional,0016044690084,1)
-- Executing [0016044690084@salida-internacional:1] Dial("SIP/5512047763-08b1b2b0", "SIP/16044690084@sip.ipacomm.net|30|r") in new stack
Audio is at 200.76.98.137 port 11802
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 208.239.76.163:5060:
INVITE sip:16044690084@sip.ipacomm.net SIP/2.0
Via: SIP/2.0/UDP 200.76.98.137:5060;branch=z9hG4bK58a79278;rport
From: "Usuario Prueba" <sip:5512047763@200.76.98.137>;tag=as0c351c86
To: <sip:16044690084@sip.ipacomm.net>
Contact: <sip:5512047763@200.76.98.137>
Call-ID: 4e14d2eb06413e2523ba9dbe100e4981@200.76.98.137
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 20 Nov 2007 00:51:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 19922 19922 IN IP4 200.76.98.137
s=session
c=IN IP4 200.76.98.137
t=0 0
m=audio 11802 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
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
---
-- Called 16044690084@sip.ipacomm.net
<--- Transmitting (no NAT) to 189.180.217.155:55828 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 189.180.217.155:55828;branch=z9hG4bK-d8754z-4fe4cc06ff31530d-1---d8754z-;rport;received=189.180.217.155
From: <sip:5512047763@sip.telavip.com.mx>;transport=UDP;tag=d813117c
To: <sip:0016044690084@sip.telavip.com.mx>;tag=as0aba4a06
Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:0016044690084@200.76.98.137>
Content-Length: 0
<------------>
localhost*CLI>
<--- SIP read from 208.239.76.163:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 200.76.98.137:5060;received=200.76.98.137;branch=z9hG4bK58a79278;rport=5060
From: "Usuario Prueba" <sip:5512047763@200.76.98.137>;tag=as0c351c86
To: <sip:16044690084@sip.ipacomm.net>
Call-ID: 4e14d2eb06413e2523ba9dbe100e4981@200.76.98.137
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
localhost*CLI>
<--- SIP read from 208.239.76.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 200.76.98.137:5060;received=200.76.98.137;branch=z9hG4bK58a79278;rport=5060
From: "Usuario Prueba" <sip:5512047763@200.76.98.137>;tag=as0c351c86
To: <sip:16044690084@sip.ipacomm.net>;tag=SD5i84f99-e3736a12
Call-ID: 4e14d2eb06413e2523ba9dbe100e4981@200.76.98.137
CSeq: 102 INVITE
X-UUID: a60162e825054ffe8d5c0e63f7f2c53f
Contact: <sip:16044690084@208.239.76.163:5060;wlsscid=1ae4691c270666;transport=udp>
Content-Length: 267
ontent-Type: application/sdp
Allow: INVITE
v=0
o=root 29482 29482 IN IP4 208.239.76.163
s=session
c=IN IP4 208.239.76.163
t=0 0
m=audio 47970 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
<------------->
--- (11 headers 12 lines) ---
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 208.239.76.163:47970
Found description format GSM for ID 3
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 208.239.76.163:47970
list_route: hop: <sip:16044690084@208.239.76.163:5060;wlsscid=1ae4691c270666;transport=udp>
set_destination: Parsing <sip:16044690084@208.239.76.163:5060;wlsscid=1ae4691c270666;transport=udp> for address/port to send to
set_destination: set destination to 208.239.76.163, port 5060
Transmitting (no NAT) to 208.239.76.163:5060:
ACK sip:16044690084@208.239.76.163:5060;wlsscid=1ae4691c270666;transport=udp SIP/2.0
Via: SIP/2.0/UDP 200.76.98.137:5060;branch=z9hG4bK54f9419e;rport
From: "Usuario Prueba" <sip:5512047763@200.76.98.137>;tag=as0c351c86
To: <sip:16044690084@sip.ipacomm.net>;tag=SD5i84f99-e3736a12
Contact: <sip:5512047763@200.76.98.137>
Call-ID: 4e14d2eb06413e2523ba9dbe100e4981@200.76.98.137
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/sip.ipacomm.net-08bb2a30 answered SIP/5512047763-08b1b2b0
Audio is at 200.76.98.137 port 16566
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 189.180.217.155:55828 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 189.180.217.155:55828;branch=z9hG4bK-d8754z-4fe4cc06ff31530d-1---d8754z-;rport;received=189.180.217.155
From: <sip:5512047763@sip.telavip.com.mx>;transport=UDP;tag=d813117c
To: <sip:0016044690084@sip.telavip.com.mx>;tag=as0aba4a06
Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:0016044690084@200.76.98.137>
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 19922 19922 IN IP4 200.76.98.137
s=session
c=IN IP4 200.76.98.137
t=0 0
m=audio 16566 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
-- Packet2Packet bridging SIP/5512047763-08b1b2b0 and SIP/sip.ipacomm.net-08bb2a30
localhost*CLI>
<--- SIP read from 189.180.217.155:55977 --->
ACK sip:0016044690084@200.76.98.137 SIP/2.0
Via: SIP/2.0/UDP 189.180.217.155:55828;branch=z9hG4bK-d8754z-60c1e71a0d6209b7-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:5512047763@189.180.217.155:55828>
To: <sip:0016044690084@sip.telavip.com.mx>;tag=as0aba4a06
From: <sip:5512047763@sip.telavip.com.mx>;transport=UDP;tag=d813117c
Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk.
CSeq: 2 ACK
Proxy-Authorization: Digest username="5512047763",realm="asterisk",nonce="7c389d18",uri="sip:0016044690084@sip.telavip.com.mx",response="c4315cb52245bc9c43111d2b11281f12",algorithm=MD5
User-Agent: Zoiper rev.417
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
localhost*CLI>
<--- SIP read from 208.239.76.163:5060 --->
BYE sip:5512047763@200.76.98.137 SIP/2.0
Via: SIP/2.0/UDP 208.239.76.163:5060;branch=z9hG4bKu9fupa104o0glf4th7g1sd0000g00.1
To: "Usuario Prueba" <sip:5512047763@200.76.98.137>;tag=as0c351c86
Content-Length: 0
CSeq: 1 BYE
Call-ID: 4e14d2eb06413e2523ba9dbe100e4981@200.76.98.137
From: <sip:16044690084@sip.ipacomm.net>;tag=SD5i84f99-e3736a12
Max-Forwards: 68
<------------->
--- (8 headers 0 lines) ---
Sending to 208.239.76.163 : 5060 (no NAT)
<--- Transmitting (no NAT) to 208.239.76.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.239.76.163:5060;branch=z9hG4bKu9fupa104o0glf4th7g1sd0000g00.1;received=208.239.76.163
From: <sip:16044690084@sip.ipacomm.net>;tag=SD5i84f99-e3736a12
To: "Usuario Prueba" <sip:5512047763@200.76.98.137>;tag=as0c351c86
Call-ID: 4e14d2eb06413e2523ba9dbe100e4981@200.76.98.137
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:5512047763@200.76.98.137>
Content-Length: 0
<------------>
== Spawn extension (salida-internacional, 0016044690084, 1) exited non-zero on 'SIP/5512047763-08b1b2b0'
Scheduling destruction of SIP dialog 'ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk.' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:5512047763@189.180.217.155:55828> for address/port to send to
set_destination: set destination to 189.180.217.155, port 55828
Reliably Transmitting (no NAT) to 189.180.217.155:55828:
BYE sip:5512047763@189.180.217.155:55828 SIP/2.0
Via: SIP/2.0/UDP 200.76.98.137:5060;branch=z9hG4bK3a9c95bf
From: <sip:0016044690084@sip.telavip.com.mx>;tag=as0aba4a06
To: <sip:5512047763@sip.telavip.com.mx>;transport=UDP;tag=d813117c
Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk.
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Really destroying SIP dialog '4e14d2eb06413e2523ba9dbe100e4981@200.76.98.137' Method: BYE
localhost*CLI>
<--- SIP read from 189.180.217.155:55977 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 200.76.98.137:5060;branch=z9hG4bK3a9c95bf
Contact: <sip:5512047763@189.180.217.155:5060>
To: <sip:5512047763@sip.telavip.com.mx>;transport=UDP;tag=d813117c
From: <sip:0016044690084@sip.telavip.com.mx>;tag=as0aba4a06
Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk.
CSeq: 102 BYE
User-Agent: Zoiper rev.417
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk.' Method: ACK
localhost*CLI> sip
<--- SIP read from 189.180.217.155:55977 --->
<------------->
--- (0 headers 1 lines) ---