thr3ads.net - asterisk users - [Asterisk-Users] phone comparison matrix [Jul 2005]

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Patrick Fortin

2005-Jul-06 06:08 UTC

[Asterisk-Users] phone comparison matrix

Hi

Is there a phone comparison matrix I could consult

I have a series of features that I would like to evaluate on the most 
common phones on the market

example:

dual-ethernet
POE / direct power / both
number of lines
speed dials programmable buttons
BLF LEDS
Headset plug
conference call built in
hands free operation
display size
codecs
communication protocol (SIP, h.323)
price
availability
reliability
know bugs / limitations
asterisk compatibility

If someone has done this recently that would save me some time

Patrick

GLNX

2005-Jul-07 13:58 UTC

head link

[Asterisk-Users] phone comparison matrix

I was just doing that compilation for a comparative matrix a couple 
weeks ago, to get a snapshot of models/features/price range

It's just a draft toy for personal use but it might be useful if I share 
it and more people help to fill it. If we setup a matrix table in 
voip-info wiki, i'll try my $0.01 releasing it

Patrick Fortin wrote:
> Hi
>
> Is there a phone comparison matrix I could consult
>
> I have a series of features that I would like to evaluate on the most 
> common phones on the market
>
> example:
>
> dual-ethernet
> POE / direct power / both
> number of lines
> speed dials programmable buttons
> BLF LEDS
> Headset plug
> conference call built in
> hands free operation
> display size
> codecs
> communication protocol (SIP, h.323)
> price
> availability
> reliability
> know bugs / limitations
> asterisk compatibility
>
> If someone has done this recently that would save me some time
>
> Patrick
>

Jeremi Bergman

2005-Jul-07 16:12 UTC

head link

[Asterisk-Users] Extension Problems

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
  <meta content="text/html;charset=ISO-8859-1"
http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
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>&nbsp;<br>
&nbsp;<br>
Sip read:&nbsp;<br>
REGISTER sip:192.168.0.50 SIP/2.0&nbsp;<br>
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-6681167b&nbsp;<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>&nbsp;<br>
To: Jeremi Bergman <sip:<a href="mailto:210@192.168.0.50"
 target="_new">210@192.168.0.50</a>>&nbsp;<br>
Call-ID: <a href="mailto:28f98bdb-a62c25a3@192.168.0.8"
target="_new">28f98bdb-a62c25a3@192.168.0.8</a>&nbsp;<br>
CSeq: 1 REGISTER&nbsp;<br>
Max-Forwards: 70&nbsp;<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>&nbsp;<br>
User-Agent: Sipura/SPA3000-2.0.10(GWc)&nbsp;<br>
Content-Length: 0&nbsp;<br>
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS,
REFER&nbsp;<br>
Supported: x-sipura&nbsp;<br>
&nbsp;<br>
&nbsp;<br>
12 headers, 0 lines&nbsp;<br>
Using latest request as basis request&nbsp;<br>
Sending to 192.168.0.8 : 5060 (non-NAT)&nbsp;<br>
Transmitting (no NAT):&nbsp;<br>
SIP/2.0 100 Trying&nbsp;<br>
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-6681167b&nbsp;<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>&nbsp;<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>&nbsp;<br>
Call-ID: <a href="mailto:28f98bdb-a62c25a3@192.168.0.8"
target="_new">28f98bdb-a62c25a3@192.168.0.8</a>&nbsp;<br>
CSeq: 1 REGISTER&nbsp;<br>
User-Agent: Asterisk PBX&nbsp;<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER&nbsp;<br>
Contact: <sip:<a href="mailto:210@192.168.0.50"
target="_new">210@192.168.0.50</a>>&nbsp;<br>
Content-Length: 0&nbsp;<br>
&nbsp;<br>
asterisk1*CLI>&nbsp;<br>
to 192.168.0.8:5060&nbsp;<br>
Transmitting (no NAT):&nbsp;<br>
SIP/2.0 401 Unauthorized&nbsp;<br>
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-6681167b&nbsp;<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>&nbsp;<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>&nbsp;<br>
Call-ID: <a href="mailto:28f98bdb-a62c25a3@192.168.0.8"
target="_new">28f98bdb-a62c25a3@192.168.0.8</a>&nbsp;<br>
CSeq: 1 REGISTER&nbsp;<br>
User-Agent: Asterisk PBX&nbsp;<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER&nbsp;<br>
Contact: <sip:<a href="mailto:210@192.168.0.50"
target="_new">210@192.168.0.50</a>>&nbsp;<br>
WWW-Authenticate: Digest realm="asterisk",
nonce="04ab00ad"&nbsp;<br>
Content-Length: 0&nbsp;<br>
&nbsp;<br>
&nbsp;<br>
to 192.168.0.8:5060&nbsp;<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&nbsp;<br>
asterisk1*CLI>&nbsp;<br>
&nbsp;<br>
Sip read:&nbsp;<br>
REGISTER sip:192.168.0.50 SIP/2.0&nbsp;<br>
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-bde1320f&nbsp;<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>&nbsp;<br>
To: Jeremi Bergman <sip:<a href="mailto:210@192.168.0.50"
 target="_new">210@192.168.0.50</a>>&nbsp;<br>
Call-ID: <a href="mailto:28f98bdb-a62c25a3@192.168.0.8"
target="_new">28f98bdb-a62c25a3@192.168.0.8</a>&nbsp;<br>
CSeq: 2 REGISTER&nbsp;<br>
Max-Forwards: 70&nbsp;<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>;&nbsp;<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>&nbsp;<br>
User-Agent: Sipura/SPA3000-2.0.10(GWc)&nbsp;<br>
Content-Length: 0&nbsp;<br>
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS,
REFER&nbsp;<br>
Supported: x-sipura&nbsp;<br>
&nbsp;<br>
&nbsp;<br>
13 headers, 0 lines&nbsp;<br>
Using latest request as basis request&nbsp;<br>
Sending to 192.168.0.8 : 5060 (non-NAT)&nbsp;<br>
Transmitting (no NAT):&nbsp;<br>
SIP/2.0 100 Trying&nbsp;<br>
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-bde1320f&nbsp;<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>&nbsp;<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>&nbsp;<br>
Call-ID: <a href="mailto:28f98bdb-a62c25a3@192.168.0.8"
target="_new">28f98bdb-a62c25a3@192.168.0.8</a>&nbsp;<br>
CSeq: 2 REGISTER&nbsp;<br>
User-Agent: Asterisk PBX&nbsp;<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER&nbsp;<br>
Contact: <sip:<a href="mailto:210@192.168.0.50"
target="_new">210@192.168.0.50</a>>&nbsp;<br>
Content-Length: 0&nbsp;<br>
&nbsp;<br>
&nbsp;<br>
to 192.168.0.8:5060&nbsp;<br>
Transmitting (no NAT):&nbsp;<br>
SIP/2.0 200 OK&nbsp;<br>
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-bde1320f&nbsp;<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>&nbsp;<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>&nbsp;<br>
Call-ID: <a href="mailto:28f98bdb-a62c25a3@192.168.0.8"
target="_new">28f98bdb-a62c25a3@192.168.0.8</a>&nbsp;<br>
CSeq: 2 REGISTER&nbsp;<br>
User-Agent: Asterisk PBX&nbsp;<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER&nbsp;<br>
Expires: 3600&nbsp;<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>&nbsp;<br>
Date: Wed, 06 Jul 2005 15:53:45 GMT&nbsp;<br>
Content-Length: 0&nbsp;<br>
&nbsp;<br>
&nbsp;<br>
to 192.168.0.8:5060&nbsp;<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&nbsp;<br>
11 headers, 2 lines&nbsp;<br>
Reliably Transmitting:&nbsp;<br>
NOTIFY sip:<a href="mailto:210@192.168.0.8:5060"
target="_new">210@192.168.0.8:5060</a>
SIP/2.0&nbsp;<br>
Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK70c85742&nbsp;<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>&nbsp;<br>
To: <sip:<a href="mailto:210@192.168.0.8:5060"
target="_new">210@192.168.0.8:5060</a>>&nbsp;<br>
Contact: <sip:<a href="mailto:Unknown@192.168.0.50"
target="_new">Unknown@192.168.0.50</a>>&nbsp;<br>
Call-ID: <a
href="mailto:0bf93c977233508342c7e42f7fa0cec1@192.168.0.50"
 target="_new">0bf93c977233508342c7e42f7fa0cec1@192.168.0.50</a>&nbsp;<br>
CSeq: 102 NOTIFY&nbsp;<br>
User-Agent: Asterisk PBX&nbsp;<br>
Event: message-summary&nbsp;<br>
Content-Type: application/simple-message-summary&nbsp;<br>
Content-Length: 42&nbsp;<br>
&nbsp;<br>
Messages-Waiting: no&nbsp;<br>
Voice-Message: 0/0&nbsp;<br>
(no NAT) to 192.168.0.8:5060&nbsp;<br>
Scheduling destruction of call '<a
 href="mailto:0bf93c977233508342c7e42f7fa0cec1@192.168.0.50"
 target="_new">0bf93c977233508342c7e42f7fa0cec1@192.168.0.50</a>'
in
15000 ms&nbsp;<br>
asterisk1*CLI>&nbsp;<br>
&nbsp;<br>
Sip read:&nbsp;<br>
SIP/2.0 200 OK&nbsp;<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>&nbsp;<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>&nbsp;<br>
Call-ID: <a
href="mailto:0bf93c977233508342c7e42f7fa0cec1@192.168.0.50"
 target="_new">0bf93c977233508342c7e42f7fa0cec1@192.168.0.50</a>&nbsp;<br>
CSeq: 102 NOTIFY&nbsp;<br>
Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK70c85742&nbsp;<br>
Server: Sipura/SPA3000-2.0.10(GWc)&nbsp;<br>
Content-Length: 0&nbsp;<br>
&nbsp;<br>
&nbsp;<br>
8 headers, 0 lines&nbsp;<br>
Destroying call '<a
 href="mailto:0bf93c977233508342c7e42f7fa0cec1@192.168.0.50"
 target="_new">0bf93c977233508342c7e42f7fa0cec1@192.168.0.50</a>'&nbsp;<br>
Destroying call '<a href="mailto:28f98bdb-a62c25a3@192.168.0.8"
 target="_new">28f98bdb-a62c25a3@192.168.0.8</a>'&nbsp;<br>
asterisk1*CLI>&nbsp;<br>
asterisk1*CLI>&nbsp;<br>
&nbsp;<br>
Sip read:&nbsp;<br>
INVITE sip:<a href="mailto:210@192.168.0.50"
target="_new">210@192.168.0.50</a>
SIP/2.0&nbsp;<br>
Via: SIP/2.0/UDP
192.168.0.2:5060;rport;branch=z9hG4bKCF6A42E8A4F84FF1B1FE8F2D8EA78724&nbsp;<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>&nbsp;<br>
To: <sip:<a href="mailto:210@192.168.0.50"
target="_new">210@192.168.0.50</a>>&nbsp;<br>
Contact: <sip:<a href="mailto:220@192.168.0.2:5060"
target="_new">220@192.168.0.2:5060</a>>&nbsp;<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>&nbsp;<br>
CSeq: 44887 INVITE&nbsp;<br>
Max-Forwards: 70&nbsp;<br>
Content-Type: application/sdp&nbsp;<br>
User-Agent: X-Lite release 1103m&nbsp;<br>
Content-Length: 292&nbsp;<br>
&nbsp;<br>
v=0&nbsp;<br>
o=220 782441328 782441343 IN IP4 192.168.0.2&nbsp;<br>
s=X-Lite&nbsp;<br>
c=IN IP4 192.168.0.2&nbsp;<br>
t=0 0&nbsp;<br>
m=audio 8000 RTP/AVP 0 8 3 98 97 101&nbsp;<br>
a=rtpmap:0 pcmu/8000&nbsp;<br>
a=rtpmap:8 pcma/8000&nbsp;<br>
a=rtpmap:3 gsm/8000&nbsp;<br>
a=rtpmap:98 iLBC/8000&nbsp;<br>
a=rtpmap:97 speex/8000&nbsp;<br>
a=rtpmap:101 telephone-event/8000&nbsp;<br>
a=fmtp:101 0-15&nbsp;<br>
&nbsp;<br>
11 headers, 13 lines&nbsp;<br>
Using latest request as basis request&nbsp;<br>
Sending to 192.168.0.2 : 5060 (non-NAT)&nbsp;<br>
Reliably Transmitting (no NAT):&nbsp;<br>
SIP/2.0 407 Proxy Authentication Required&nbsp;<br>
Via: SIP/2.0/UDP
192.168.0.2:5060;branch=z9hG4bKCF6A42E8A4F84FF1B1FE8F2D8EA78724&nbsp;<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>&nbsp;<br>
To: <sip:<a href="mailto:210@192.168.0.50%3E;tag=as063c6cef"
 target="_new">210@192.168.0.50>;tag=as063c6cef</a>&nbsp;<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>&nbsp;<br>
CSeq: 44887 INVITE&nbsp;<br>
User-Agent: Asterisk PBX&nbsp;<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER&nbsp;<br>
Contact: <sip:<a href="mailto:210@192.168.0.50"
target="_new">210@192.168.0.50</a>>&nbsp;<br>
Proxy-Authenticate: Digest realm="asterisk",
nonce="71694d20"&nbsp;<br>
Content-Length: 0&nbsp;<br>
&nbsp;<br>
&nbsp;<br>
to 192.168.0.2:5060&nbsp;<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&nbsp;<br>
Found user '220'&nbsp;<br>
asterisk1*CLI>&nbsp;<br>
&nbsp;<br>
Sip read:&nbsp;<br>
ACK sip:<a href="mailto:210@192.168.0.50"
target="_new">210@192.168.0.50</a>
SIP/2.0&nbsp;<br>
Via: SIP/2.0/UDP
192.168.0.2:5060;rport;branch=z9hG4bKCF6A42E8A4F84FF1B1FE8F2D8EA78724&nbsp;<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>&nbsp;<br>
To: <sip:<a href="mailto:210@192.168.0.50%3E;tag=as063c6cef"
 target="_new">210@192.168.0.50>;tag=as063c6cef</a>&nbsp;<br>
Contact: <sip:<a href="mailto:220@192.168.0.2:5060"
target="_new">220@192.168.0.2:5060</a>>&nbsp;<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>&nbsp;<br>
CSeq: 44887 ACK&nbsp;<br>
Max-Forwards: 70&nbsp;<br>
Content-Length: 0&nbsp;<br>
&nbsp;<br>
&nbsp;<br>
9 headers, 0 lines&nbsp;<br>
&nbsp;<br>
&nbsp;<br>
Sip read:&nbsp;<br>
INVITE sip:<a href="mailto:210@192.168.0.50"
target="_new">210@192.168.0.50</a>
SIP/2.0&nbsp;<br>
Via: SIP/2.0/UDP
192.168.0.2:5060;rport;branch=z9hG4bK2A632B189EC04A0CBAEB17D1B3342931&nbsp;<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>&nbsp;<br>
To: <sip:<a href="mailto:210@192.168.0.50"
target="_new">210@192.168.0.50</a>>&nbsp;<br>
Contact: <sip:<a href="mailto:220@192.168.0.2:5060"
target="_new">220@192.168.0.2:5060</a>>&nbsp;<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>&nbsp;<br>
CSeq: 44888 INVITE&nbsp;<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>;&nbsp;<br>
Max-Forwards: 70&nbsp;<br>
Content-Type: application/sdp&nbsp;<br>
User-Agent: X-Lite release 1103m&nbsp;<br>
Content-Length: 292&nbsp;<br>
&nbsp;<br>
v=0&nbsp;<br>
o=220 782441328 782441343 IN IP4 192.168.0.2&nbsp;<br>
s=X-Lite&nbsp;<br>
c=IN IP4 192.168.0.2&nbsp;<br>
t=0 0&nbsp;<br>
m=audio 8000 RTP/AVP 0 8 3 98 97 101&nbsp;<br>
a=rtpmap:0 pcmu/8000&nbsp;<br>
a=rtpmap:8 pcma/8000&nbsp;<br>
a=rtpmap:3 gsm/8000&nbsp;<br>
a=rtpmap:98 iLBC/8000&nbsp;<br>
a=rtpmap:97 speex/8000&nbsp;<br>
a=rtpmap:101 telephone-event/8000&nbsp;<br>
a=fmtp:101 0-15&nbsp;<br>
&nbsp;<br>
12 headers, 13 lines&nbsp;<br>
Using latest request as basis request&nbsp;<br>
Sending to 192.168.0.2 : 5060 (non-NAT)&nbsp;<br>
Found user '220'&nbsp;<br>
Found RTP audio format 0&nbsp;<br>
Found RTP audio format 8&nbsp;<br>
Found RTP audio format 3&nbsp;<br>
Found RTP audio format 98&nbsp;<br>
Found RTP audio format 97&nbsp;<br>
Found RTP audio format 101&nbsp;<br>
Peer audio RTP is at port 192.168.0.2:8000&nbsp;<br>
Found description format pcmu&nbsp;<br>
Found description format pcma&nbsp;<br>
Found description format gsm&nbsp;<br>
Found description format iLBC&nbsp;<br>
Found description format speex&nbsp;<br>
Found description format telephone-event&nbsp;<br>
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x60e
(gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xc
(ulaw|alaw)&nbsp;<br>
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)&nbsp;<br>
Looking for 210 in from-internal&nbsp;<br>
list_route: hop: <sip:<a href="mailto:220@192.168.0.2:5060"
 target="_new">220@192.168.0.2:5060</a>>&nbsp;<br>
Transmitting (no NAT):&nbsp;<br>
SIP/2.0 100 Trying&nbsp;<br>
Via: SIP/2.0/UDP
192.168.0.2:5060;branch=z9hG4bK2A632B189EC04A0CBAEB17D1B3342931&nbsp;<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>&nbsp;<br>
To: <sip:<a href="mailto:210@192.168.0.50%3E;tag=as3714a6d8"
 target="_new">210@192.168.0.50>;tag=as3714a6d8</a>&nbsp;<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>&nbsp;<br>
CSeq: 44888 INVITE&nbsp;<br>
User-Agent: Asterisk PBX&nbsp;<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER&nbsp;<br>
Contact: <sip:<a href="mailto:210@192.168.0.50"
target="_new">210@192.168.0.50</a>>&nbsp;<br>
Content-Length: 0&nbsp;<br>
&nbsp;<br>
&nbsp;<br>
to 192.168.0.2:5060&nbsp;<br>
-- Executing Macro("SIP/220-0f69", "exten-vm|<a
 href="mailto:210@default%7C210%22"
target="_new">210@default|210"</a>;)
in new stack&nbsp;<br>
-- Executing SetVar("SIP/220-0f69", "FROMCONTEXT=exten-vm")
in new
stack&nbsp;<br>
-- Executing Macro("SIP/220-0f69", "record-enable|210|IN")
in new stack&nbsp;<br>
-- Executing GotoIf("SIP/220-0f69", "0 > 0?2:4") in new
stack&nbsp;<br>
-- Goto (macro-record-enable,s,4)&nbsp;<br>
-- Executing GotoIf("SIP/220-0f69", "0?5:8") in new
stack&nbsp;<br>
-- Goto (macro-record-enable,s,8)&nbsp;<br>
-- Executing GotoIf("SIP/220-0f69", "0?9:12") in new
stack&nbsp;<br>
-- Goto (macro-record-enable,s,12)&nbsp;<br>
-- Executing DBget("SIP/220-0f69",
"RecEnable=RECORD-IN/210") in new
stack&nbsp;<br>
-- DBget: varname=RecEnable, family=RECORD-IN, key=210&nbsp;<br>
-- DBget: Value not found in database.&nbsp;<br>
-- Executing SetVar("SIP/220-0f69",
"CALLFILENAME=20050706-115857-1120665537.5") in new
stack&nbsp;<br>
-- Executing GotoIf("SIP/220-0f69", "0?15:99") in new
stack&nbsp;<br>
-- Goto (macro-record-enable,s,99)&nbsp;<br>
-- Executing NoOp("SIP/220-0f69", "NO RECORDING NEEDED") in
new stack&nbsp;<br>
-- Executing Macro("SIP/220-0f69", "dial|15|tr|210") in new
stack&nbsp;<br>
-- Executing GotoIf("SIP/220-0f69", "0?4:2") in new
stack&nbsp;<br>
-- Goto (macro-dial,s,2)&nbsp;<br>
-- Executing GotoIf("SIP/220-0f69", "0?4:3") in new
stack&nbsp;<br>
-- Goto (macro-dial,s,3)&nbsp;<br>
-- Executing SetCIDName("SIP/220-0f69", "David Johnson") in
new stack&nbsp;<br>
-- Executing AGI("SIP/220-0f69", "dialparties.agi") in new
stack&nbsp;<br>
-- Launched AGI Script
/var/lib/asterisk/agi-bin/dialparties.agi&nbsp;<br>
-- AGI Script dialparties.agi completed, returning 0&nbsp;<br>
-- Executing NoOp("SIP/220-0f69", "Returned from dialparties with
no
extensions to call") in new stack&nbsp;<br>
-- Executing SetVar("SIP/220-0f69", "DIALSTATUS=BUSY") in
new stack&nbsp;<br>
-- Executing GotoIf("SIP/220-0f69", "0?s-BUSY|1") in new
stack&nbsp;<br>
-- Executing GotoIf("SIP/220-0f69", "0?s-BUSY|1") in new
stack&nbsp;<br>
-- Executing NoOp("SIP/220-0f69", "Sending to Voicemail box <a
 href="mailto:210@default%22"
target="_new">210@default"</a>;) in new
stack&nbsp;<br>
-- Executing Macro("SIP/220-0f69", "vm|<a
 href="mailto:210@default%7CBUSY%22"
target="_new">210@default|BUSY"</a>;)
in new stack&nbsp;<br>
-- Executing Goto("SIP/220-0f69", "s-BUSY|1") in new
stack&nbsp;<br>
-- Goto (macro-vm,s-BUSY,1)&nbsp;<br>
-- Executing VoiceMail("SIP/220-0f69", "<a
href="mailto:b210@default%22"
 target="_new">b210@default"</a>;) in new
stack&nbsp;<br>
We're at 192.168.0.50 port 17334&nbsp;<br>
Answering with preferred capability 0x4 (ulaw)&nbsp;<br>
Answering with preferred capability 0x8 (alaw)&nbsp;<br>
Answering with non-codec capability 0x1 (telephone-event)&nbsp;<br>
Reliably Transmitting (no NAT):&nbsp;<br>
SIP/2.0 200 OK&nbsp;<br>
Via: SIP/2.0/UDP
192.168.0.2:5060;branch=z9hG4bK2A632B189EC04A0CBAEB17D1B3342931&nbsp;<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>&nbsp;<br>
To: <sip:<a href="mailto:210@192.168.0.50%3E;tag=as3714a6d8"
 target="_new">210@192.168.0.50>;tag=as3714a6d8</a>&nbsp;<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>&nbsp;<br>
CSeq: 44888 INVITE&nbsp;<br>
User-Agent: Asterisk PBX&nbsp;<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER&nbsp;<br>
Contact: <sip:<a href="mailto:210@192.168.0.50"
target="_new">210@192.168.0.50</a>>&nbsp;<br>
Content-Type: application/sdp&nbsp;<br>
Content-Length: 238&nbsp;<br>
&nbsp;<br>
v=0&nbsp;<br>
o=root 1983 1983 IN IP4 192.168.0.50&nbsp;<br>
s=session&nbsp;<br>
c=IN IP4 192.168.0.50&nbsp;<br>
t=0 0&nbsp;<br>
m=audio 17334 RTP/AVP 0 8 101&nbsp;<br>
a=rtpmap:0 PCMU/8000&nbsp;<br>
a=rtpmap:8 PCMA/8000&nbsp;<br>
a=rtpmap:101 telephone-event/8000&nbsp;<br>
a=fmtp:101 0-16&nbsp;<br>
a=silenceSupp:off - - - -&nbsp;<br>
&nbsp;<br>
to 192.168.0.2:5060&nbsp;<br>
-- Playing 'vm-theperson' (language 'en')&nbsp;<br>
asterisk1*CLI>&nbsp;<br>
&nbsp;<br>
Sip read:&nbsp;<br>
ACK sip:<a href="mailto:210@192.168.0.50"
target="_new">210@192.168.0.50</a>
SIP/2.0&nbsp;<br>
Via: SIP/2.0/UDP
192.168.0.2:5060;rport;branch=z9hG4bK39CF2EA3F9794B68A722E58AB6B681D1&nbsp;<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>&nbsp;<br>
To: <sip:<a href="mailto:210@192.168.0.50%3E;tag=as3714a6d8"
 target="_new">210@192.168.0.50>;tag=as3714a6d8</a>&nbsp;<br>
Contact: <sip:<a href="mailto:220@192.168.0.2:5060"
target="_new">220@192.168.0.2:5060</a>>&nbsp;<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>&nbsp;<br>
CSeq: 44888 ACK&nbsp;<br>
Max-Forwards: 70&nbsp;<br>
Content-Length: 0&nbsp;<br>
&nbsp;<br>
&nbsp;<br>
9 headers, 0 lines&nbsp;<br>
-- Playing 'digits/2' (language 'en')&nbsp;<br>
-- Playing 'digits/1' (language 'en')&nbsp;<br>
-- Playing 'digits/0' (language 'en')&nbsp;<br>
-- Playing 'vm-isonphone' (language 'en')&nbsp;<br>
asterisk1*CLI>&nbsp;<br>
&nbsp;<br>
Sip read:&nbsp;<br>
&nbsp;<br>
&nbsp;<br>
0 headers, 0 lines&nbsp;<br>
-- Playing 'vm-intro' (language 'en')&nbsp;<br>
-- Playing 'beep' (language 'en')&nbsp;<br>
-- Recording the message&nbsp;<br>
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/210/INBOX/msg0001 format: wav49,
0x8615f78&nbsp;<br>
-- x=1, open writing:
/var/spool/asterisk/voicemail/default/210/INBOX/msg0001 format: wav,
0x8605ea0&nbsp;<br>
asterisk1*CLI>&nbsp;<br>
&nbsp;<br>
Sip read:&nbsp;<br>
BYE sip:<a href="mailto:210@192.168.0.50"
target="_new">210@192.168.0.50</a>
SIP/2.0&nbsp;<br>
Via: SIP/2.0/UDP
192.168.0.2:5060;rport;branch=z9hG4bKBB5910FF6B464AA2AC22A34DAC7CE76F&nbsp;<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>&nbsp;<br>
To: <sip:<a href="mailto:210@192.168.0.50%3E;tag=as3714a6d8"
 target="_new">210@192.168.0.50>;tag=as3714a6d8</a>&nbsp;<br>
Contact: <sip:<a href="mailto:220@192.168.0.2:5060"
target="_new">220@192.168.0.2:5060</a>>&nbsp;<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>&nbsp;<br>
CSeq: 44889 BYE&nbsp;<br>
Max-Forwards: 70&nbsp;<br>
User-Agent: X-Lite release 1103m&nbsp;<br>
Content-Length: 0&nbsp;<br>
&nbsp;<br>
&nbsp;<br>
10 headers, 0 lines&nbsp;<br>
Sending to 192.168.0.2 : 5060 (non-NAT)&nbsp;<br>
Transmitting (no NAT):&nbsp;<br>
SIP/2.0 200 OK&nbsp;<br>
Via: SIP/2.0/UDP
192.168.0.2:5060;branch=z9hG4bKBB5910FF6B464AA2AC22A34DAC7CE76F&nbsp;<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>&nbsp;<br>
To: <sip:<a href="mailto:210@192.168.0.50%3E;tag=as3714a6d8"
 target="_new">210@192.168.0.50>;tag=as3714a6d8</a>&nbsp;<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>&nbsp;<br>
CSeq: 44889 BYE&nbsp;<br>
User-Agent: Asterisk PBX&nbsp;<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER&nbsp;<br>
Contact: <sip:<a href="mailto:210@192.168.0.50"
target="_new">210@192.168.0.50</a>>&nbsp;<br>
Content-Length: 0&nbsp;<br>
&nbsp;<br>
&nbsp;<br>
to 192.168.0.2:5060&nbsp;<br>
-- User hung up&nbsp;<br>
-- Recording was 2 seconds long but needs to be at least 3 -
abandoning&nbsp;<br>
== Spawn extension (macro-vm, s-BUSY, 1) exited non-zero on
'SIP/220-0f69' in macro 'vm'&nbsp;<br>
== Spawn extension (macro-exten-vm, s, 7) exited non-zero on
'SIP/220-0f69' in macro 'exten-vm'&nbsp;<br>
== Spawn extension (from-internal, 210, 1) exited non-zero on
'SIP/220-0f69'&nbsp;<br>
-- Executing Macro("SIP/220-0f69", "hangupcall") in new
stack&nbsp;<br>
-- Executing ResetCDR("SIP/220-0f69", "w") in new
stack&nbsp;<br>
-- Executing NoCDR("SIP/220-0f69", "") in new
stack&nbsp;<br>
-- Executing Wait("SIP/220-0f69", "5") in new
stack&nbsp;<br>
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'SIP/220-0f69' in macro 'hangupcall'&nbsp;<br>
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/220-0f69'&nbsp;<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>'&nbsp;<br>
asterisk1*CLI>&nbsp;
<br>
---------------<br>
<br>
Thanks in advanced!<br>
</body>
</html>

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