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The extensions I've created in AAH, when dialed, always go straight to
voicemail.<br>
<br>
I may be missing a step... I'm simply adding it in the "Extensions"
part of AAH.<br>
<br>
I can dial out with my extension, and recieve the voicemail
notification, so I know i'm logged in, or so I thought...<br>
<br>
This is SIP 210 logging in and 220 making a call to 210<br>
<br>
-------------------<br>
asterisk1*CLI> <br>
<br>
Sip read: <br>
REGISTER sip:192.168.0.50 SIP/2.0 <br>
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-6681167b <br>
From: Jeremi Bergman <sip:<a
href="mailto:210@192.168.0.50%3E;tag=49a471fb8d817603o0" target="_new">210@192.168.0.50>;tag=49a471fb8d817603o0</a> <br>
To: Jeremi Bergman <sip:<a href="mailto:210@192.168.0.50"
target="_new">210@192.168.0.50</a>> <br>
Call-ID: <a href="mailto:28f98bdb-a62c25a3@192.168.0.8" target="_new">28f98bdb-a62c25a3@192.168.0.8</a> <br>
CSeq: 1 REGISTER <br>
Max-Forwards: 70 <br>
Contact: Jeremi Bergman <sip:<a
href="mailto:210@192.168.0.8:5060%3E;expires=3600" target="_new">210@192.168.0.8:5060>;expires=3600</a> <br>
User-Agent: Sipura/SPA3000-2.0.10(GWc) <br>
Content-Length: 0 <br>
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER <br>
Supported: x-sipura <br>
<br>
<br>
12 headers, 0 lines <br>
Using latest request as basis request <br>
Sending to 192.168.0.8 : 5060 (non-NAT) <br>
Transmitting (no NAT): <br>
SIP/2.0 100 Trying <br>
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-6681167b <br>
From: Jeremi Bergman <sip:<a
href="mailto:210@192.168.0.50%3E;tag=49a471fb8d817603o0" target="_new">210@192.168.0.50>;tag=49a471fb8d817603o0</a> <br>
To: Jeremi Bergman <sip:<a
href="mailto:210@192.168.0.50%3E;tag=as0373da7c" target="_new">210@192.168.0.50>;tag=as0373da7c</a> <br>
Call-ID: <a href="mailto:28f98bdb-a62c25a3@192.168.0.8" target="_new">28f98bdb-a62c25a3@192.168.0.8</a> <br>
CSeq: 1 REGISTER <br>
User-Agent: Asterisk PBX <br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER <br>
Contact: <sip:<a href="mailto:210@192.168.0.50" target="_new">210@192.168.0.50</a>> <br>
Content-Length: 0 <br>
<br>
asterisk1*CLI> <br>
to 192.168.0.8:5060 <br>
Transmitting (no NAT): <br>
SIP/2.0 401 Unauthorized <br>
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-6681167b <br>
From: Jeremi Bergman <sip:<a
href="mailto:210@192.168.0.50%3E;tag=49a471fb8d817603o0" target="_new">210@192.168.0.50>;tag=49a471fb8d817603o0</a> <br>
To: Jeremi Bergman <sip:<a
href="mailto:210@192.168.0.50%3E;tag=as0373da7c" target="_new">210@192.168.0.50>;tag=as0373da7c</a> <br>
Call-ID: <a href="mailto:28f98bdb-a62c25a3@192.168.0.8" target="_new">28f98bdb-a62c25a3@192.168.0.8</a> <br>
CSeq: 1 REGISTER <br>
User-Agent: Asterisk PBX <br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER <br>
Contact: <sip:<a href="mailto:210@192.168.0.50" target="_new">210@192.168.0.50</a>> <br>
WWW-Authenticate: Digest realm="asterisk", nonce="04ab00ad" <br>
Content-Length: 0 <br>
<br>
<br>
to 192.168.0.8:5060 <br>
Scheduling destruction of call '<a
href="mailto:28f98bdb-a62c25a3@192.168.0.8" target="_new">28f98bdb-a62c25a3@192.168.0.8</a>'
in 15000 ms <br>
asterisk1*CLI> <br>
<br>
Sip read: <br>
REGISTER sip:192.168.0.50 SIP/2.0 <br>
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-bde1320f <br>
From: Jeremi Bergman <sip:<a
href="mailto:210@192.168.0.50%3E;tag=49a471fb8d817603o0" target="_new">210@192.168.0.50>;tag=49a471fb8d817603o0</a> <br>
To: Jeremi Bergman <sip:<a href="mailto:210@192.168.0.50"
target="_new">210@192.168.0.50</a>> <br>
Call-ID: <a href="mailto:28f98bdb-a62c25a3@192.168.0.8" target="_new">28f98bdb-a62c25a3@192.168.0.8</a> <br>
CSeq: 2 REGISTER <br>
Max-Forwards: 70 <br>
Authorization: Digest
username="210",realm="asterisk",nonce="04ab00ad",uri="sip:<a
href="mailto:210@192.168.0.50%22,algorithm=MD5,response=%224b5484b65bc24fc38c8cdff7684a9452%22"
target="_new">210@192.168.0.50",algorithm=MD5,response="4b5484b65bc24fc38c8cdff7684a9452"</a>; <br>
Contact: Jeremi Bergman <sip:<a
href="mailto:210@192.168.0.8:5060%3E;expires=3600" target="_new">210@192.168.0.8:5060>;expires=3600</a> <br>
User-Agent: Sipura/SPA3000-2.0.10(GWc) <br>
Content-Length: 0 <br>
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER <br>
Supported: x-sipura <br>
<br>
<br>
13 headers, 0 lines <br>
Using latest request as basis request <br>
Sending to 192.168.0.8 : 5060 (non-NAT) <br>
Transmitting (no NAT): <br>
SIP/2.0 100 Trying <br>
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-bde1320f <br>
From: Jeremi Bergman <sip:<a
href="mailto:210@192.168.0.50%3E;tag=49a471fb8d817603o0" target="_new">210@192.168.0.50>;tag=49a471fb8d817603o0</a> <br>
To: Jeremi Bergman <sip:<a
href="mailto:210@192.168.0.50%3E;tag=as0373da7c" target="_new">210@192.168.0.50>;tag=as0373da7c</a> <br>
Call-ID: <a href="mailto:28f98bdb-a62c25a3@192.168.0.8" target="_new">28f98bdb-a62c25a3@192.168.0.8</a> <br>
CSeq: 2 REGISTER <br>
User-Agent: Asterisk PBX <br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER <br>
Contact: <sip:<a href="mailto:210@192.168.0.50" target="_new">210@192.168.0.50</a>> <br>
Content-Length: 0 <br>
<br>
<br>
to 192.168.0.8:5060 <br>
Transmitting (no NAT): <br>
SIP/2.0 200 OK <br>
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-bde1320f <br>
From: Jeremi Bergman <sip:<a
href="mailto:210@192.168.0.50%3E;tag=49a471fb8d817603o0" target="_new">210@192.168.0.50>;tag=49a471fb8d817603o0</a> <br>
To: Jeremi Bergman <sip:<a
href="mailto:210@192.168.0.50%3E;tag=as0373da7c" target="_new">210@192.168.0.50>;tag=as0373da7c</a> <br>
Call-ID: <a href="mailto:28f98bdb-a62c25a3@192.168.0.8" target="_new">28f98bdb-a62c25a3@192.168.0.8</a> <br>
CSeq: 2 REGISTER <br>
User-Agent: Asterisk PBX <br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER <br>
Expires: 3600 <br>
Contact: <sip:<a href="mailto:210@192.168.0.8:5060%3E;expires=3600"
target="_new">210@192.168.0.8:5060>;expires=3600</a> <br>
Date: Wed, 06 Jul 2005 15:53:45 GMT <br>
Content-Length: 0 <br>
<br>
<br>
to 192.168.0.8:5060 <br>
Scheduling destruction of call '<a
href="mailto:28f98bdb-a62c25a3@192.168.0.8" target="_new">28f98bdb-a62c25a3@192.168.0.8</a>'
in 15000 ms <br>
11 headers, 2 lines <br>
Reliably Transmitting: <br>
NOTIFY sip:<a href="mailto:210@192.168.0.8:5060" target="_new">210@192.168.0.8:5060</a>
SIP/2.0 <br>
Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK70c85742 <br>
From: "Unknown" <sip:<a
href="mailto:Unknown@192.168.0.50%3E;tag=as60a8b668" target="_new">Unknown@192.168.0.50>;tag=as60a8b668</a> <br>
To: <sip:<a href="mailto:210@192.168.0.8:5060" target="_new">210@192.168.0.8:5060</a>> <br>
Contact: <sip:<a href="mailto:Unknown@192.168.0.50" target="_new">Unknown@192.168.0.50</a>> <br>
Call-ID: <a href="mailto:0bf93c977233508342c7e42f7fa0cec1@192.168.0.50"
target="_new">0bf93c977233508342c7e42f7fa0cec1@192.168.0.50</a> <br>
CSeq: 102 NOTIFY <br>
User-Agent: Asterisk PBX <br>
Event: message-summary <br>
Content-Type: application/simple-message-summary <br>
Content-Length: 42 <br>
<br>
Messages-Waiting: no <br>
Voice-Message: 0/0 <br>
(no NAT) to 192.168.0.8:5060 <br>
Scheduling destruction of call '<a
href="mailto:0bf93c977233508342c7e42f7fa0cec1@192.168.0.50"
target="_new">0bf93c977233508342c7e42f7fa0cec1@192.168.0.50</a>' in
15000 ms <br>
asterisk1*CLI> <br>
<br>
Sip read: <br>
SIP/2.0 200 OK <br>
To: <sip:<a
href="mailto:210@192.168.0.8:5060%3E;tag=e365bb5ba561bc23i0"
target="_new">210@192.168.0.8:5060>;tag=e365bb5ba561bc23i0</a> <br>
From: "Unknown" <sip:<a
href="mailto:Unknown@192.168.0.50%3E;tag=as60a8b668" target="_new">Unknown@192.168.0.50>;tag=as60a8b668</a> <br>
Call-ID: <a href="mailto:0bf93c977233508342c7e42f7fa0cec1@192.168.0.50"
target="_new">0bf93c977233508342c7e42f7fa0cec1@192.168.0.50</a> <br>
CSeq: 102 NOTIFY <br>
Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK70c85742 <br>
Server: Sipura/SPA3000-2.0.10(GWc) <br>
Content-Length: 0 <br>
<br>
<br>
8 headers, 0 lines <br>
Destroying call '<a
href="mailto:0bf93c977233508342c7e42f7fa0cec1@192.168.0.50"
target="_new">0bf93c977233508342c7e42f7fa0cec1@192.168.0.50</a>' <br>
Destroying call '<a href="mailto:28f98bdb-a62c25a3@192.168.0.8"
target="_new">28f98bdb-a62c25a3@192.168.0.8</a>' <br>
asterisk1*CLI> <br>
asterisk1*CLI> <br>
<br>
Sip read: <br>
INVITE sip:<a href="mailto:210@192.168.0.50" target="_new">210@192.168.0.50</a>
SIP/2.0 <br>
Via: SIP/2.0/UDP
192.168.0.2:5060;rport;branch=z9hG4bKCF6A42E8A4F84FF1B1FE8F2D8EA78724 <br>
From: Jeremi Bergman <sip:<a
href="mailto:220@192.168.0.50%3E;tag=2051996763" target="_new">220@192.168.0.50>;tag=2051996763</a> <br>
To: <sip:<a href="mailto:210@192.168.0.50" target="_new">210@192.168.0.50</a>> <br>
Contact: <sip:<a href="mailto:220@192.168.0.2:5060" target="_new">220@192.168.0.2:5060</a>> <br>
Call-ID: <a
href="mailto:24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2"
target="_new">24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2</a> <br>
CSeq: 44887 INVITE <br>
Max-Forwards: 70 <br>
Content-Type: application/sdp <br>
User-Agent: X-Lite release 1103m <br>
Content-Length: 292 <br>
<br>
v=0 <br>
o=220 782441328 782441343 IN IP4 192.168.0.2 <br>
s=X-Lite <br>
c=IN IP4 192.168.0.2 <br>
t=0 0 <br>
m=audio 8000 RTP/AVP 0 8 3 98 97 101 <br>
a=rtpmap:0 pcmu/8000 <br>
a=rtpmap:8 pcma/8000 <br>
a=rtpmap:3 gsm/8000 <br>
a=rtpmap:98 iLBC/8000 <br>
a=rtpmap:97 speex/8000 <br>
a=rtpmap:101 telephone-event/8000 <br>
a=fmtp:101 0-15 <br>
<br>
11 headers, 13 lines <br>
Using latest request as basis request <br>
Sending to 192.168.0.2 : 5060 (non-NAT) <br>
Reliably Transmitting (no NAT): <br>
SIP/2.0 407 Proxy Authentication Required <br>
Via: SIP/2.0/UDP
192.168.0.2:5060;branch=z9hG4bKCF6A42E8A4F84FF1B1FE8F2D8EA78724 <br>
From: Jeremi Bergman <sip:<a
href="mailto:220@192.168.0.50%3E;tag=2051996763" target="_new">220@192.168.0.50>;tag=2051996763</a> <br>
To: <sip:<a href="mailto:210@192.168.0.50%3E;tag=as063c6cef"
target="_new">210@192.168.0.50>;tag=as063c6cef</a> <br>
Call-ID: <a
href="mailto:24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2"
target="_new">24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2</a> <br>
CSeq: 44887 INVITE <br>
User-Agent: Asterisk PBX <br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER <br>
Contact: <sip:<a href="mailto:210@192.168.0.50" target="_new">210@192.168.0.50</a>> <br>
Proxy-Authenticate: Digest realm="asterisk", nonce="71694d20" <br>
Content-Length: 0 <br>
<br>
<br>
to 192.168.0.2:5060 <br>
Scheduling destruction of call '<a
href="mailto:24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2"
target="_new">24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2</a>' in
15000 ms <br>
Found user '220' <br>
asterisk1*CLI> <br>
<br>
Sip read: <br>
ACK sip:<a href="mailto:210@192.168.0.50" target="_new">210@192.168.0.50</a>
SIP/2.0 <br>
Via: SIP/2.0/UDP
192.168.0.2:5060;rport;branch=z9hG4bKCF6A42E8A4F84FF1B1FE8F2D8EA78724 <br>
From: Jeremi Bergman <sip:<a
href="mailto:220@192.168.0.50%3E;tag=2051996763" target="_new">220@192.168.0.50>;tag=2051996763</a> <br>
To: <sip:<a href="mailto:210@192.168.0.50%3E;tag=as063c6cef"
target="_new">210@192.168.0.50>;tag=as063c6cef</a> <br>
Contact: <sip:<a href="mailto:220@192.168.0.2:5060" target="_new">220@192.168.0.2:5060</a>> <br>
Call-ID: <a
href="mailto:24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2"
target="_new">24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2</a> <br>
CSeq: 44887 ACK <br>
Max-Forwards: 70 <br>
Content-Length: 0 <br>
<br>
<br>
9 headers, 0 lines <br>
<br>
<br>
Sip read: <br>
INVITE sip:<a href="mailto:210@192.168.0.50" target="_new">210@192.168.0.50</a>
SIP/2.0 <br>
Via: SIP/2.0/UDP
192.168.0.2:5060;rport;branch=z9hG4bK2A632B189EC04A0CBAEB17D1B3342931 <br>
From: Jeremi Bergman <sip:<a
href="mailto:220@192.168.0.50%3E;tag=2051996763" target="_new">220@192.168.0.50>;tag=2051996763</a> <br>
To: <sip:<a href="mailto:210@192.168.0.50" target="_new">210@192.168.0.50</a>> <br>
Contact: <sip:<a href="mailto:220@192.168.0.2:5060" target="_new">220@192.168.0.2:5060</a>> <br>
Call-ID: <a
href="mailto:24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2"
target="_new">24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2</a> <br>
CSeq: 44888 INVITE <br>
Proxy-Authorization: Digest
username="220",realm="asterisk",nonce="71694d20",response="d1ee9ed08d97c4b058f97a24f64ba5a4",uri="sip:<a
href="mailto:210@192.168.0.50%22" target="_new">210@192.168.0.50"</a>; <br>
Max-Forwards: 70 <br>
Content-Type: application/sdp <br>
User-Agent: X-Lite release 1103m <br>
Content-Length: 292 <br>
<br>
v=0 <br>
o=220 782441328 782441343 IN IP4 192.168.0.2 <br>
s=X-Lite <br>
c=IN IP4 192.168.0.2 <br>
t=0 0 <br>
m=audio 8000 RTP/AVP 0 8 3 98 97 101 <br>
a=rtpmap:0 pcmu/8000 <br>
a=rtpmap:8 pcma/8000 <br>
a=rtpmap:3 gsm/8000 <br>
a=rtpmap:98 iLBC/8000 <br>
a=rtpmap:97 speex/8000 <br>
a=rtpmap:101 telephone-event/8000 <br>
a=fmtp:101 0-15 <br>
<br>
12 headers, 13 lines <br>
Using latest request as basis request <br>
Sending to 192.168.0.2 : 5060 (non-NAT) <br>
Found user '220' <br>
Found RTP audio format 0 <br>
Found RTP audio format 8 <br>
Found RTP audio format 3 <br>
Found RTP audio format 98 <br>
Found RTP audio format 97 <br>
Found RTP audio format 101 <br>
Peer audio RTP is at port 192.168.0.2:8000 <br>
Found description format pcmu <br>
Found description format pcma <br>
Found description format gsm <br>
Found description format iLBC <br>
Found description format speex <br>
Found description format telephone-event <br>
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x60e
(gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xc
(ulaw|alaw) <br>
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723) <br>
Looking for 210 in from-internal <br>
list_route: hop: <sip:<a href="mailto:220@192.168.0.2:5060"
target="_new">220@192.168.0.2:5060</a>> <br>
Transmitting (no NAT): <br>
SIP/2.0 100 Trying <br>
Via: SIP/2.0/UDP
192.168.0.2:5060;branch=z9hG4bK2A632B189EC04A0CBAEB17D1B3342931 <br>
From: Jeremi Bergman <sip:<a
href="mailto:220@192.168.0.50%3E;tag=2051996763" target="_new">220@192.168.0.50>;tag=2051996763</a> <br>
To: <sip:<a href="mailto:210@192.168.0.50%3E;tag=as3714a6d8"
target="_new">210@192.168.0.50>;tag=as3714a6d8</a> <br>
Call-ID: <a
href="mailto:24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2"
target="_new">24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2</a> <br>
CSeq: 44888 INVITE <br>
User-Agent: Asterisk PBX <br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER <br>
Contact: <sip:<a href="mailto:210@192.168.0.50" target="_new">210@192.168.0.50</a>> <br>
Content-Length: 0 <br>
<br>
<br>
to 192.168.0.2:5060 <br>
-- Executing Macro("SIP/220-0f69", "exten-vm|<a
href="mailto:210@default%7C210%22" target="_new">210@default|210"</a>;)
in new stack <br>
-- Executing SetVar("SIP/220-0f69", "FROMCONTEXT=exten-vm") in new
stack <br>
-- Executing Macro("SIP/220-0f69", "record-enable|210|IN") in new stack <br>
-- Executing GotoIf("SIP/220-0f69", "0 > 0?2:4") in new stack <br>
-- Goto (macro-record-enable,s,4) <br>
-- Executing GotoIf("SIP/220-0f69", "0?5:8") in new stack <br>
-- Goto (macro-record-enable,s,8) <br>
-- Executing GotoIf("SIP/220-0f69", "0?9:12") in new stack <br>
-- Goto (macro-record-enable,s,12) <br>
-- Executing DBget("SIP/220-0f69", "RecEnable=RECORD-IN/210") in new
stack <br>
-- DBget: varname=RecEnable, family=RECORD-IN, key=210 <br>
-- DBget: Value not found in database. <br>
-- Executing SetVar("SIP/220-0f69",
"CALLFILENAME=20050706-115857-1120665537.5") in new stack <br>
-- Executing GotoIf("SIP/220-0f69", "0?15:99") in new stack <br>
-- Goto (macro-record-enable,s,99) <br>
-- Executing NoOp("SIP/220-0f69", "NO RECORDING NEEDED") in new stack <br>
-- Executing Macro("SIP/220-0f69", "dial|15|tr|210") in new stack <br>
-- Executing GotoIf("SIP/220-0f69", "0?4:2") in new stack <br>
-- Goto (macro-dial,s,2) <br>
-- Executing GotoIf("SIP/220-0f69", "0?4:3") in new stack <br>
-- Goto (macro-dial,s,3) <br>
-- Executing SetCIDName("SIP/220-0f69", "David Johnson") in new stack <br>
-- Executing AGI("SIP/220-0f69", "dialparties.agi") in new stack <br>
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi <br>
-- AGI Script dialparties.agi completed, returning 0 <br>
-- Executing NoOp("SIP/220-0f69", "Returned from dialparties with no
extensions to call") in new stack <br>
-- Executing SetVar("SIP/220-0f69", "DIALSTATUS=BUSY") in new stack <br>
-- Executing GotoIf("SIP/220-0f69", "0?s-BUSY|1") in new stack <br>
-- Executing GotoIf("SIP/220-0f69", "0?s-BUSY|1") in new stack <br>
-- Executing NoOp("SIP/220-0f69", "Sending to Voicemail box <a
href="mailto:210@default%22" target="_new">210@default"</a>;) in new
stack <br>
-- Executing Macro("SIP/220-0f69", "vm|<a
href="mailto:210@default%7CBUSY%22" target="_new">210@default|BUSY"</a>;)
in new stack <br>
-- Executing Goto("SIP/220-0f69", "s-BUSY|1") in new stack <br>
-- Goto (macro-vm,s-BUSY,1) <br>
-- Executing VoiceMail("SIP/220-0f69", "<a href="mailto:b210@default%22"
target="_new">b210@default"</a>;) in new stack <br>
We're at 192.168.0.50 port 17334 <br>
Answering with preferred capability 0x4 (ulaw) <br>
Answering with preferred capability 0x8 (alaw) <br>
Answering with non-codec capability 0x1 (telephone-event) <br>
Reliably Transmitting (no NAT): <br>
SIP/2.0 200 OK <br>
Via: SIP/2.0/UDP
192.168.0.2:5060;branch=z9hG4bK2A632B189EC04A0CBAEB17D1B3342931 <br>
From: Jeremi Bergman <sip:<a
href="mailto:220@192.168.0.50%3E;tag=2051996763" target="_new">220@192.168.0.50>;tag=2051996763</a> <br>
To: <sip:<a href="mailto:210@192.168.0.50%3E;tag=as3714a6d8"
target="_new">210@192.168.0.50>;tag=as3714a6d8</a> <br>
Call-ID: <a
href="mailto:24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2"
target="_new">24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2</a> <br>
CSeq: 44888 INVITE <br>
User-Agent: Asterisk PBX <br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER <br>
Contact: <sip:<a href="mailto:210@192.168.0.50" target="_new">210@192.168.0.50</a>> <br>
Content-Type: application/sdp <br>
Content-Length: 238 <br>
<br>
v=0 <br>
o=root 1983 1983 IN IP4 192.168.0.50 <br>
s=session <br>
c=IN IP4 192.168.0.50 <br>
t=0 0 <br>
m=audio 17334 RTP/AVP 0 8 101 <br>
a=rtpmap:0 PCMU/8000 <br>
a=rtpmap:8 PCMA/8000 <br>
a=rtpmap:101 telephone-event/8000 <br>
a=fmtp:101 0-16 <br>
a=silenceSupp:off - - - - <br>
<br>
to 192.168.0.2:5060 <br>
-- Playing 'vm-theperson' (language 'en') <br>
asterisk1*CLI> <br>
<br>
Sip read: <br>
ACK sip:<a href="mailto:210@192.168.0.50" target="_new">210@192.168.0.50</a>
SIP/2.0 <br>
Via: SIP/2.0/UDP
192.168.0.2:5060;rport;branch=z9hG4bK39CF2EA3F9794B68A722E58AB6B681D1 <br>
From: Jeremi Bergman <sip:<a
href="mailto:220@192.168.0.50%3E;tag=2051996763" target="_new">220@192.168.0.50>;tag=2051996763</a> <br>
To: <sip:<a href="mailto:210@192.168.0.50%3E;tag=as3714a6d8"
target="_new">210@192.168.0.50>;tag=as3714a6d8</a> <br>
Contact: <sip:<a href="mailto:220@192.168.0.2:5060" target="_new">220@192.168.0.2:5060</a>> <br>
Call-ID: <a
href="mailto:24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2"
target="_new">24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2</a> <br>
CSeq: 44888 ACK <br>
Max-Forwards: 70 <br>
Content-Length: 0 <br>
<br>
<br>
9 headers, 0 lines <br>
-- Playing 'digits/2' (language 'en') <br>
-- Playing 'digits/1' (language 'en') <br>
-- Playing 'digits/0' (language 'en') <br>
-- Playing 'vm-isonphone' (language 'en') <br>
asterisk1*CLI> <br>
<br>
Sip read: <br>
<br>
<br>
0 headers, 0 lines <br>
-- Playing 'vm-intro' (language 'en') <br>
-- Playing 'beep' (language 'en') <br>
-- Recording the message <br>
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/210/INBOX/msg0001 format: wav49,
0x8615f78 <br>
-- x=1, open writing:
/var/spool/asterisk/voicemail/default/210/INBOX/msg0001 format: wav,
0x8605ea0 <br>
asterisk1*CLI> <br>
<br>
Sip read: <br>
BYE sip:<a href="mailto:210@192.168.0.50" target="_new">210@192.168.0.50</a>
SIP/2.0 <br>
Via: SIP/2.0/UDP
192.168.0.2:5060;rport;branch=z9hG4bKBB5910FF6B464AA2AC22A34DAC7CE76F <br>
From: Jeremi Bergman <sip:<a
href="mailto:220@192.168.0.50%3E;tag=2051996763" target="_new">220@192.168.0.50>;tag=2051996763</a> <br>
To: <sip:<a href="mailto:210@192.168.0.50%3E;tag=as3714a6d8"
target="_new">210@192.168.0.50>;tag=as3714a6d8</a> <br>
Contact: <sip:<a href="mailto:220@192.168.0.2:5060" target="_new">220@192.168.0.2:5060</a>> <br>
Call-ID: <a
href="mailto:24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2"
target="_new">24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2</a> <br>
CSeq: 44889 BYE <br>
Max-Forwards: 70 <br>
User-Agent: X-Lite release 1103m <br>
Content-Length: 0 <br>
<br>
<br>
10 headers, 0 lines <br>
Sending to 192.168.0.2 : 5060 (non-NAT) <br>
Transmitting (no NAT): <br>
SIP/2.0 200 OK <br>
Via: SIP/2.0/UDP
192.168.0.2:5060;branch=z9hG4bKBB5910FF6B464AA2AC22A34DAC7CE76F <br>
From: Jeremi Bergman <sip:<a
href="mailto:220@192.168.0.50%3E;tag=2051996763" target="_new">220@192.168.0.50>;tag=2051996763</a> <br>
To: <sip:<a href="mailto:210@192.168.0.50%3E;tag=as3714a6d8"
target="_new">210@192.168.0.50>;tag=as3714a6d8</a> <br>
Call-ID: <a
href="mailto:24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2"
target="_new">24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2</a> <br>
CSeq: 44889 BYE <br>
User-Agent: Asterisk PBX <br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER <br>
Contact: <sip:<a href="mailto:210@192.168.0.50" target="_new">210@192.168.0.50</a>> <br>
Content-Length: 0 <br>
<br>
<br>
to 192.168.0.2:5060 <br>
-- User hung up <br>
-- Recording was 2 seconds long but needs to be at least 3 - abandoning <br>
== Spawn extension (macro-vm, s-BUSY, 1) exited non-zero on
'SIP/220-0f69' in macro 'vm' <br>
== Spawn extension (macro-exten-vm, s, 7) exited non-zero on
'SIP/220-0f69' in macro 'exten-vm' <br>
== Spawn extension (from-internal, 210, 1) exited non-zero on
'SIP/220-0f69' <br>
-- Executing Macro("SIP/220-0f69", "hangupcall") in new stack <br>
-- Executing ResetCDR("SIP/220-0f69", "w") in new stack <br>
-- Executing NoCDR("SIP/220-0f69", "") in new stack <br>
-- Executing Wait("SIP/220-0f69", "5") in new stack <br>
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'SIP/220-0f69' in macro 'hangupcall' <br>
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/220-0f69' <br>
Destroying call '<a
href="mailto:24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2"
target="_new">24FFA722-455A-4022-9D89-119784F97A90@192.168.0.2</a>' <br>
asterisk1*CLI>
<br>
---------------<br>
<br>
Thanks in advanced!<br>
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