<div dir="ltr">Maybe stupid solution but, when Mitel phone i called, why dont you pickup put the person on hold, call Mitel phone, and connect them, what i want to say, add some delay.<br><br><div class="gmail_quote">2008/7/19 Mark Wiater <<a href="mailto:mwiater@cablespeed.com">mwiater@cablespeed.com</a>>:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div class="Ih2E3d">Matt Watson wrote:<br>
> On July 19, 2008 11:22:08 am Mark Wiater wrote:<br>
>> Hi,<br>
>><br>
>> I have a client using Mitel 5212 phones in SIP mode with a <a href="http://1.4.21.1" target="_blank">1.4.21.1</a><br>
>> Asterisk server (and a couple of previous 1.4 versions). They're<br>
>> mostly happy with the combination except for this one issue.<br>
>><br>
>> For incoming calls only, either originating from other local SIP<br>
>> phones or from a PRI, calls won't get bridged (remote party get's<br>
>> hung up) if the call is answer too quickly on the Mitel. Or so it<br>
>> seems. The receiving Mitel phone thinks the call is in session though.<br>
><br>
>> Asterisk is reporting errors like:<br>
>><br>
>> [Jul 19 10:46:41] NOTICE[2466]: chan_sip.c:8068<br>
>> set_address_from_contact: '"<a href="http://72.16.1.20" target="_blank">72.16.1.20</a>>;tag=as7b9f4bfb' is not a<br>
>> valid SIP contact (missing sip:) trying to use anyway<br>
>> [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:8097<br>
>> set_address_from_contact: Invalid host name in Contact: (can't<br>
>> resolve in DNS) : '"<a href="http://72.16.1.20" target="_blank">72.16.1.20</a>>'<br>
>> [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination:<br>
>> Can't find address for host '"<a href="http://72.16.1.20" target="_blank">72.16.1.20</a>'<br>
>> [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination:<br>
>> Can't find address for host '"<a href="http://72.16.1.20" target="_blank">72.16.1.20</a>'<br>
>><br>
><br>
> Might want to post a sip debug of one of the sessions from the Mitel phone.<br>
><br>
><br>
<br>
</div>Thanks Matt<br>
<br>
I was also able to test this with Mitel's firmware version <a href="http://7.0.0.8" target="_blank">7.0.0.8</a><br>
with the same results.<br>
<br>
Mitel phone still acts like it's on a call, Asterisk does not nor<br>
does the originating phone.<br>
<br>
PBX*CLI> sip set debug peer 517<br>
SIP Debugging Enabled for IP: <a href="http://172.16.1.174:5060" target="_blank">172.16.1.174:5060</a><br>
Audio is at <a href="http://172.16.1.20" target="_blank">172.16.1.20</a> port 15594<br>
Adding codec 0x4 (ulaw) to SDP<br>
Adding non-codec 0x1 (telephone-event) to SDP<br>
Reliably Transmitting (no NAT) to <a href="http://172.16.1.174:5060" target="_blank">172.16.1.174:5060</a>:<br>
INVITE <a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a> SIP/2.0<br>
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK38581b5a;rport<br>
From: "512" <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>>;tag=as7ec9e8af<br>
To: <<a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a>><br>
Contact: <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>><br>
Call-ID: <a href="mailto:2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20">2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20</a><br>
CSeq: 102 INVITE<br>
User-Agent: Asterisk PBX<br>
Max-Forwards: 70<br>
Date: Sat, 19 Jul 2008 17:20:54 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces<br>
Content-Type: application/sdp<br>
Content-Length: 236<br>
<br>
v=0<br>
o=root 2247 2247 IN IP4 <a href="http://172.16.1.20" target="_blank">172.16.1.20</a><br>
s=session<br>
c=IN IP4 <a href="http://172.16.1.20" target="_blank">172.16.1.20</a><br>
t=0 0<br>
m=audio 15594 RTP/AVP 0 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=silenceSupp:off - - - -<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
---<br>
PBX*CLI><br>
<--- SIP read from <a href="http://172.16.1.174:5060" target="_blank">172.16.1.174:5060</a> ---><br>
SIP/2.0 100 Trying<br>
Via:SIP/2.0/UDP<br>
172.16.1.20:5060;rport=5060;received=<a href="http://172.16.1.20" target="_blank">172.16.1.20</a>;branch=z9hG4bK38581b5a<br>
From:"512" <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>>;tag=as7ec9e8af<br>
To:<<a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965<br>
CSeq:102 INVITE<br>
User-Agent:Mitel-5212-SIP-Phone <a href="http://07.00.00.08" target="_blank">07.00.00.08</a> 08000F32A6C1<br>
<a href="mailto:Call-ID%3A2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20">Call-ID:2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20</a><br>
Content-Length:0<br>
<br>
<br>
<-------------><br>
--- (8 headers 0 lines) ---<br>
PBX*CLI><br>
<--- SIP read from <a href="http://172.16.1.174:5060" target="_blank">172.16.1.174:5060</a> ---><br>
SIP/2.0 180 Ringing<br>
Via:SIP/2.0/UDP<br>
172.16.1.20:5060;rport=5060;received=<a href="http://172.16.1.20" target="_blank">172.16.1.20</a>;branch=z9hG4bK38581b5a<br>
From:"512" <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>>;tag=as7ec9e8af<br>
To:<<a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965<br>
CSeq:102 INVITE<br>
User-Agent:Mitel-5212-SIP-Phone <a href="http://07.00.00.08" target="_blank">07.00.00.08</a> 08000F32A6C1<br>
<a href="mailto:Call-ID%3A2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20">Call-ID:2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20</a><br>
Allow-Events:talk,hold,conference<br>
Content-Length:0<br>
<br>
<br>
<-------------><br>
--- (9 headers 0 lines) ---<br>
PBX*CLI><br>
<--- SIP read from <a href="http://172.16.1.174:5060" target="_blank">172.16.1.174:5060</a> ---><br>
SIP/2.0 200 OK<br>
Via:SIP/2.0/UDP<br>
172.16.1.20:5060;rport=5060;received=<a href="http://172.16.1.20" target="_blank">172.16.1.20</a>;branch=z9hG4bK38581b5a<br>
From:"512" <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>>;tag=as7ec9e8af<br>
To:<<a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965<br>
CSeq:102 INVITE<br>
User-Agent:Mitel-5212-SIP-Phone <a href="http://07.00.00.08" target="_blank">07.00.00.08</a> 08000F32A6C1<br>
<a href="mailto:Call-ID%3A2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20">Call-ID:2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20</a><br>
Contact:"<a href="mailto:p%3A517@172.16.1.174">p:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965<br>
CSeq:102 INVITE<br>
User" <<a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a>><br>
Allow-Events:talk,hold,conference<br>
Allow:INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,NOTIFY,PRACK,UPDATE<br>
Supported:timer,100rel,replaces<br>
Content-Type:application/sdp<br>
Content-Length:182<br>
<br>
v=0<br>
o=517 1216473942 1216473941 IN IP4 <a href="http://172.16.1.174" target="_blank">172.16.1.174</a><br>
s=SIP Call<br>
c=IN IP4 <a href="http://172.16.1.174" target="_blank">172.16.1.174</a><br>
t=0 0<br>
m=audio 20012 RTP/AVP 0 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:101 telephone-event/8000<br>
<br>
<-------------><br>
--- (15 headers 8 lines) ---<br>
Found RTP audio format 0<br>
Found RTP audio format 101<br>
Peer audio RTP is at port <a href="http://172.16.1.174:20012" target="_blank">172.16.1.174:20012</a><br>
Found audio description format PCMU for ID 0<br>
Found audio description format telephone-event for ID 101<br>
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0<br>
(nothing), combined - 0x4 (ulaw)<br>
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer -<br>
0x1 (telephone-event), combined - 0x1 (telephone-event)<br>
Peer audio RTP is at port <a href="http://172.16.1.174:20012" target="_blank">172.16.1.174:20012</a><br>
[Jul 19 13:20:56] NOTICE[2466]: chan_sip.c:8068<br>
set_address_from_contact:<br>
'"<a href="mailto:p%3A517@172.16.1.174">p:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965' is not a valid SIP<br>
<div class="Ih2E3d">contact (missing sip:) trying to use anyway<br>
</div>[Jul 19 13:20:56] WARNING[2466]: chan_sip.c:8097<br>
<div class="Ih2E3d">set_address_from_contact: Invalid host name in Contact: (can't<br>
</div>resolve in DNS) : '<a href="http://172.16.1.174" target="_blank">172.16.1.174</a>>'<br>
list_route: hop: <"<a href="mailto:p%3A517@172.16.1.174">p:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965><br>
set_destination: Parsing<br>
<"<a href="mailto:p%3A517@172.16.1.174">p:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965> for address/port to<br>
send to<br>
set_destination: set destination to <a href="http://172.16.1.174" target="_blank">172.16.1.174</a>, port 5060<br>
Transmitting (no NAT) to <a href="http://172.16.1.174:5060" target="_blank">172.16.1.174:5060</a>:<br>
ACK "<a href="mailto:p%3A517@172.16.1.174">p:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965 SIP/2.0<br>
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK100c41c2;rport<br>
From: "512" <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>>;tag=as7ec9e8af<br>
To: <<a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965<br>
Contact: <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>><br>
Call-ID: <a href="mailto:2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20">2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20</a><br>
CSeq: 102 ACK<br>
User-Agent: Asterisk PBX<br>
Max-Forwards: 70<br>
Content-Length: 0<br>
<br>
<br>
---<br>
set_destination: Parsing<br>
<"<a href="mailto:p%3A517@172.16.1.174">p:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965> for address/port to<br>
send to<br>
set_destination: set destination to <a href="http://172.16.1.174" target="_blank">172.16.1.174</a>, port 5060<br>
Reliably Transmitting (no NAT) to <a href="http://172.16.1.174:5060" target="_blank">172.16.1.174:5060</a>:<br>
BYE "<a href="mailto:p%3A517@172.16.1.174">p:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965 SIP/2.0<br>
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK45d6127c;rport<br>
From: "512" <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>>;tag=as7ec9e8af<br>
To: <<a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965<br>
Call-ID: <a href="mailto:2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20">2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20</a><br>
CSeq: 103 BYE<br>
User-Agent: Asterisk PBX<br>
Max-Forwards: 70<br>
X-Asterisk-HangupCause: Normal Clearing<br>
X-Asterisk-HangupCauseCode: 16<br>
Content-Length: 0<br>
<br>
<br>
---<br>
Scheduling destruction of SIP dialog<br>
'<a href="mailto:2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20">2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20</a>' in 32000 ms (Method:<br>
INVITE)<br>
set_destination: Parsing<br>
<"<a href="mailto:p%3A517@172.16.1.174">p:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965> for address/port to<br>
send to<br>
set_destination: set destination to <a href="http://172.16.1.174" target="_blank">172.16.1.174</a>, port 5060<br>
Audio is at <a href="http://172.16.1.20" target="_blank">172.16.1.20</a> port 15594<br>
Adding codec 0x4 (ulaw) to SDP<br>
Adding non-codec 0x1 (telephone-event) to SDP<br>
Reliably Transmitting (no NAT) to <a href="http://172.16.1.174:5060" target="_blank">172.16.1.174:5060</a>:<br>
INVITE "<a href="mailto:p%3A517@172.16.1.174">p:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965 SIP/2.0<br>
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK46ec2f8b;rport<br>
From: "512" <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>>;tag=as7ec9e8af<br>
To: <<a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965<br>
Contact: <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>><br>
Call-ID: <a href="mailto:2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20">2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20</a><br>
CSeq: 104 INVITE<br>
User-Agent: Asterisk PBX<br>
Max-Forwards: 70<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces<br>
X-asterisk-Info: SIP re-invite (External RTP bridge)<br>
Content-Type: application/sdp<br>
Content-Length: 237<br>
<br>
v=0<br>
o=root 2247 2248 IN IP4 <a href="http://172.16.1.156" target="_blank">172.16.1.156</a><br>
s=session<br>
c=IN IP4 <a href="http://172.16.1.156" target="_blank">172.16.1.156</a><br>
t=0 0<br>
m=audio 2224 RTP/AVP 0 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=silenceSupp:off - - - -<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
---<br>
PBX*CLI><br>
<--- SIP read from <a href="http://172.16.1.174:5060" target="_blank">172.16.1.174:5060</a> ---><br>
SIP/2.0 416 Unsupported URI Scheme<br>
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK45d6127c;rport<br>
From:"512" <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>>;tag=as7ec9e8af<br>
To:<<a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965<br>
CSeq:103 BYE<br>
User-Agent:Mitel-5212-SIP-Phone <a href="http://07.00.00.08" target="_blank">07.00.00.08</a> 08000F32A6C1<br>
<a href="mailto:Call-ID%3A2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20">Call-ID:2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20</a><br>
Content-Length:0<br>
<br>
<br>
<-------------><br>
--- (8 headers 0 lines) ---<br>
PBX*CLI><br>
<--- SIP read from <a href="http://172.16.1.174:5060" target="_blank">172.16.1.174:5060</a> ---><br>
SIP/2.0 416 Unsupported URI Scheme<br>
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK46ec2f8b;rport<br>
From:"512" <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>>;tag=as7ec9e8af<br>
To:<<a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965<br>
CSeq:104 INVITE<br>
User-Agent:Mitel-5212-SIP-Phone <a href="http://07.00.00.08" target="_blank">07.00.00.08</a> 08000F32A6C1<br>
<a href="mailto:Call-ID%3A2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20">Call-ID:2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20</a><br>
Content-Length:0<br>
<br>
<br>
<-------------><br>
--- (8 headers 0 lines) ---<br>
set_destination: Parsing<br>
<"<a href="mailto:p%3A517@172.16.1.174">p:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965> for address/port to<br>
send to<br>
set_destination: set destination to <a href="http://172.16.1.174" target="_blank">172.16.1.174</a>, port 5060<br>
Transmitting (no NAT) to <a href="http://172.16.1.174:5060" target="_blank">172.16.1.174:5060</a>:<br>
ACK "<a href="mailto:p%3A517@172.16.1.174">p:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965 SIP/2.0<br>
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK46ec2f8b;rport<br>
From: "512" <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>>;tag=as7ec9e8af<br>
To: <<a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965<br>
Contact: <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>><br>
Call-ID: <a href="mailto:2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20">2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20</a><br>
CSeq: 104 ACK<br>
User-Agent: Asterisk PBX<br>
Max-Forwards: 70<br>
Content-Length: 0<br>
<br>
<br>
---<br>
PBX*CLI><br>
<--- SIP read from <a href="http://172.16.1.174:5060" target="_blank">172.16.1.174:5060</a> ---><br>
SIP/2.0 200 OK<br>
Via:SIP/2.0/UDP<br>
172.16.1.20:5060;rport=5060;received=<a href="http://172.16.1.20" target="_blank">172.16.1.20</a>;branch=z9hG4bK38581b5a<br>
From:"512" <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>>;tag=as7ec9e8af<br>
To:<<a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965<br>
CSeq:102 INVITE<br>
User-Agent:Mitel-5212-SIP-Phone <a href="http://07.00.00.08" target="_blank">07.00.00.08</a> 08000F32A6C1<br>
<a href="mailto:Call-ID%3A2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20">Call-ID:2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20</a><br>
Contact:"<a href="mailto:p%3A517@172.16.1.174">p:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965<br>
CSeq:102 INVITE<br>
User" <<a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a>><br>
Allow-Events:talk,hold,conference<br>
Allow:INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,NOTIFY,PRACK,UPDATE<br>
Supported:timer,100rel,replaces<br>
Content-Type:application/sdp<br>
Content-Length:182<br>
<br>
v=0<br>
o=517 1216473942 1216473941 IN IP4 <a href="http://172.16.1.174" target="_blank">172.16.1.174</a><br>
s=SIP Call<br>
c=IN IP4 <a href="http://172.16.1.174" target="_blank">172.16.1.174</a><br>
t=0 0<br>
m=audio 20012 RTP/AVP 0 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:101 telephone-event/8000<br>
<br>
<-------------><br>
--- (15 headers 8 lines) ---<br>
Retransmitting #1 (no NAT) to <a href="http://172.16.1.174:5060" target="_blank">172.16.1.174:5060</a>:<br>
BYE "<a href="mailto:p%3A517@172.16.1.174">p:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965 SIP/2.0<br>
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK45d6127c;rport<br>
From: "512" <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>>;tag=as7ec9e8af<br>
To: <<a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965<br>
Call-ID: <a href="mailto:2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20">2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20</a><br>
CSeq: 103 BYE<br>
User-Agent: Asterisk PBX<br>
Max-Forwards: 70<br>
X-Asterisk-HangupCause: Normal Clearing<br>
X-Asterisk-HangupCauseCode: 16<br>
Content-Length: 0<br>
<br>
<br>
---<br>
PBX*CLI><br>
<--- SIP read from <a href="http://172.16.1.174:5060" target="_blank">172.16.1.174:5060</a> ---><br>
SIP/2.0 416 Unsupported URI Scheme<br>
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK45d6127c;rport<br>
From:"512" <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>>;tag=as7ec9e8af<br>
To:<<a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965<br>
CSeq:103 BYE<br>
User-Agent:Mitel-5212-SIP-Phone <a href="http://07.00.00.08" target="_blank">07.00.00.08</a> 08000F32A6C1<br>
<a href="mailto:Call-ID%3A2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20">Call-ID:2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20</a><br>
Content-Length:0<br>
<br>
<br>
<-------------><br>
--- (8 headers 0 lines) ---<br>
PBX*CLI> sip set debug peer 517<br>
<--- SIP read from <a href="http://172.16.1.174:5060" target="_blank">172.16.1.174:5060</a> ---><br>
SIP/2.0 200 OK<br>
Via:SIP/2.0/UDP<br>
172.16.1.20:5060;rport=5060;received=<a href="http://172.16.1.20" target="_blank">172.16.1.20</a>;branch=z9hG4bK38581b5a<br>
From:"512" <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>>;tag=as7ec9e8af<br>
To:<<a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965<br>
CSeq:102 INVITE<br>
User-Agent:Mitel-5212-SIP-Phone <a href="http://07.00.00.08" target="_blank">07.00.00.08</a> 08000F32A6C1<br>
<a href="mailto:Call-ID%3A2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20">Call-ID:2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20</a><br>
Contact:"<a href="mailto:p%3A517@172.16.1.174">p:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965<br>
CSeq:102 INVITE<br>
User" <<a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a>><br>
Allow-Events:talk,hold,conference<br>
Allow:INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,NOTIFY,PRACK,UPDATE<br>
Supported:timer,100rel,replaces<br>
Content-Type:application/sdp<br>
Content-Length:182<br>
<br>
v=0<br>
o=517 1216473942 1216473941 IN IP4 <a href="http://172.16.1.174" target="_blank">172.16.1.174</a><br>
s=SIP Call<br>
c=IN IP4 <a href="http://172.16.1.174" target="_blank">172.16.1.174</a><br>
t=0 0<br>
m=audio 20012 RTP/AVP 0 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:101 telephone-event/8000<br>
<br>
<-------------><br>
--- (15 headers 8 lines) ---<br>
Retransmitting #2 (no NAT) to <a href="http://172.16.1.174:5060" target="_blank">172.16.1.174:5060</a>:<br>
BYE "<a href="mailto:p%3A517@172.16.1.174">p:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965 SIP/2.0<br>
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK45d6127c;rport<br>
From: "512" <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>>;tag=as7ec9e8af<br>
To: <<a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965<br>
Call-ID: <a href="mailto:2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20">2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20</a><br>
CSeq: 103 BYE<br>
User-Agent: Asterisk PBX<br>
Max-Forwards: 70<br>
X-Asterisk-HangupCause: Normal Clearing<br>
X-Asterisk-HangupCauseCode: 16<br>
Content-Length: 0<br>
<br>
<br>
---<br>
PBX*CLI> sip set debug off<br>
<--- SIP read from <a href="http://172.16.1.174:5060" target="_blank">172.16.1.174:5060</a> ---><br>
SIP/2.0 416 Unsupported URI Scheme<br>
Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK45d6127c;rport<br>
From:"512" <<a href="mailto:sip%3A512@172.16.1.20">sip:512@172.16.1.20</a>>;tag=as7ec9e8af<br>
To:<<a href="mailto:sip%3A517@172.16.1.174">sip:517@172.16.1.174</a>>;tag=4881ea36-2ca-6747d965<br>
CSeq:103 BYE<br>
User-Agent:Mitel-5212-SIP-Phone <a href="http://07.00.00.08" target="_blank">07.00.00.08</a> 08000F32A6C1<br>
<a href="mailto:Call-ID%3A2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20">Call-ID:2cf91dcf4de0aecd2993bc8c6a799efa@172.16.1.20</a><br>
Content-Length:0<br>
<br>
<br>
<-------------><br>
--- (8 headers 0 lines) ---<br>
greybeamPBX*CLI> sip set debug off<br>
<div><div></div><div class="Wj3C7c"><br>
<br>
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