I set: sip debug peer 3000 (my test extension) and dialed #3643873<div>Here is the output:</div><div><br></div><div><div><div><-- SIP read from <a href="http://192.168.1.59:17456">192.168.1.59:17456</a>:</div><div>INVITE <a href="mailto:sip%3A%25233643873@192.168.2.10">sip:%233643873@192.168.2.10</a> SIP/2.0</div>
<div>Via: SIP/2.0/UDP 192.168.1.59:17456;branch=z9hG4bK-d8754z-e210db3acd1ed62b-1---d8754z-;rport</div><div>Max-Forwards: 70</div><div>Contact: <<a href="http://sip:3000@192.168.1.59:17456">sip:3000@192.168.1.59:17456</a>></div>
<div>To: "#3643873"<<a href="mailto:sip%3A%25233643873@192.168.2.10">sip:%233643873@192.168.2.10</a>></div><div>From: "Test"<<a href="mailto:sip%3A3000@192.168.2.10">sip:3000@192.168.2.10</a>>;tag=76126b35</div>
<div>Call-ID: NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE.</div><div>CSeq: 2 INVITE</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO</div><div>Content-Type: application/sdp</div>
<div>Proxy-Authorization: Digest username="3000",realm="asterisk",nonce="6a7a2c99",uri="<a href="mailto:sip%3A%25233643873@192.168.2.10">sip:%233643873@192.168.2.10</a>",response="7bcf9339c154ef939bd575aeaaef1860",algorithm=MD5</div>
<div>User-Agent: X-Lite release 1104o stamp 56125</div><div>Content-Length: 317</div><div><br></div><div>v=0</div><div>o=- 8 2 IN IP4 192.168.1.59</div><div>s=CounterPath X-Lite 3.0</div><div>c=IN IP4 192.168.1.59</div><div>
t=0 0</div><div>m=audio 34194 RTP/AVP 107 0 8 101</div><div>a=alt:1 2 : MsRCET/S fNqrHReN 192.168.200.113 34194</div><div>a=alt:2 1 : pf8wX3Si UdjtGUj2 192.168.1.59 34194</div><div>a=fmtp:101 0-15</div><div>a=rtpmap:107 BV32/16000</div>
<div>a=rtpmap:101 telephone-event/8000</div><div>a=sendrecv</div><div><br></div><div>--- (13 headers 12 lines)---</div><div>Using INVITE request as basis request - NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE.</div><div>Sending to 192.168.1.59 : 17456 (non-NAT)</div>
<div>Found user '3000'</div><div>Found RTP audio format 107</div><div>Found RTP audio format 0</div><div>Found RTP audio format 8</div><div>Found RTP audio format 101</div><div>Peer audio RTP is at port <a href="http://192.168.1.59:34194">192.168.1.59:34194</a></div>
<div>Found description format BV32</div><div>Found description format telephone-event</div><div>Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)</div><div>Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)</div>
<div>Looking for %233643873 in from-internal (domain 192.168.2.10)</div><div>Reliably Transmitting (no NAT) to <a href="http://192.168.1.59:17456">192.168.1.59:17456</a>:</div><div>SIP/2.0 404 Not Found</div><div>Via: SIP/2.0/UDP 192.168.1.59:17456;branch=z9hG4bK-d8754z-e210db3acd1ed62b-1---d8754z-;rport;received=192.168.1.59</div>
<div>From: "Test"<<a href="mailto:sip%3A3000@192.168.2.10">sip:3000@192.168.2.10</a>>;tag=76126b35</div><div>To: "#3643873"<<a href="mailto:sip%3A%25233643873@192.168.2.10">sip:%233643873@192.168.2.10</a>>;tag=as3d020428</div>
<div>Call-ID: NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE.</div><div>CSeq: 2 INVITE</div><div>User-Agent: Asterisk PBX</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</div><div>Max-Forwards: 70</div>
<div>Contact: <<a href="mailto:sip%3A%25233643873@192.168.2.10">sip:%233643873@192.168.2.10</a>></div><div>Content-Length: 0</div><div><br></div><div><br></div><div>---</div><div>aikphone*CLI></div><div><-- SIP read from <a href="http://192.168.1.59:17456">192.168.1.59:17456</a>:</div>
<div>ACK <a href="mailto:sip%3A%25233643873@192.168.2.10">sip:%233643873@192.168.2.10</a> SIP/2.0</div><div>Via: SIP/2.0/UDP 192.168.1.59:17456;branch=z9hG4bK-d8754z-e210db3acd1ed62b-1---d8754z-;rport</div><div>To: "#3643873"<<a href="mailto:sip%3A%25233643873@192.168.2.10">sip:%233643873@192.168.2.10</a>>;tag=as3d020428</div>
<div>From: "Hull Barrett"<<a href="mailto:sip%3A3000@192.168.2.10">sip:3000@192.168.2.10</a>>;tag=76126b35</div><div>Call-ID: NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE.</div><div>CSeq: 2 ACK</div><div>Content-Length: 0</div>
<div><br></div><div><br></div><div>--- (7 headers 0 lines)---</div><div>Destroying call 'NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE.'</div></div><div><br></div><br><div class="gmail_quote">On Wed, May 5, 2010 at 12:40 PM, Danny Nicholas <span dir="ltr"><<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div lang="EN-US" link="blue" vlink="blue">
<div>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy">Ok. I’m confused. I was interpreting
what you wrote to say that you are doing this:</span></font></p>
<ol style="margin-top:0in" start="1" type="1">
<li class="MsoNormal" style="color:navy"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial">pick
up sip phone attached to pbx1 (1.2 box)</span></font></li>
<li class="MsoNormal" style="color:navy"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial">dial
#5551212</span></font></li>
<li class="MsoNormal" style="color:navy"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial">command
dial(iax2/trunk/5551212,30,r) gets executed on 1.2 box </span></font></li>
<li class="MsoNormal" style="color:navy"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial">1.4
box should fall into _XXXXXXX and do DAHDI dial?</span></font></li>
</ol>
<p class="MsoNormal" style="margin-left:.25in"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy"> </span></font></p>
<p class="MsoNormal" style="margin-left:.25in"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy">If this
is correct, where is the IAX command in your CLI output.</span></font></p>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy"> </span></font></p>
<div>
<div class="MsoNormal" align="center" style="text-align:center"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">
<hr size="2" width="100%" align="center">
</span></font></div>
<p class="MsoNormal"><b><font size="2" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma;font-weight:bold">From:</span></font></b><font size="2" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma">
<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b><span style="font-weight:bold">On Behalf Of </span></b>David Nickel<br>
<b><span style="font-weight:bold">Sent:</span></b> Wednesday, May 05, 2010
10:11 AM</span></font></p><font size="2" face="Tahoma"><div><div></div><div class="h5"><br>
<b><span style="font-weight:bold">To:</span></b> Asterisk
Users Mailing List - Non-Commercial Discussion<br>
<b><span style="font-weight:bold">Subject:</span></b> Re: [asterisk-users] Hash
Dial Pattern Problems</div></div></font><p></p>
</div><div><div></div><div class="h5">
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt"> </span></font></p>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">I am on the 1.2 box and see nothing with the verbose cranked up. I do
see the following when tailing the asterisk full log during the calls:</span></font></p>
<div>
<div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">May 5 11:09:46 DEBUG[26538] chan_sip.c: Setting NAT on RTP to 0</span></font></p>
</div>
<div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">May 5 11:09:46 DEBUG[26538] chan_sip.c: Checking SIP call limits
for device 3000</span></font></p>
</div>
<div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">May 5 11:09:46 DEBUG[26538] chan_sip.c: Stopping retransmission
on 'Njg3MjI5N2IzNDk3NWYxZTMzMzFmMzEwNzc2ZDE1NTE.' of Response 2: Match Found</span></font></p>
</div>
<div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt"> </span></font></p>
</div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt"> </span></font></p>
<div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">On Wed, May 5, 2010 at 10:39 AM, Danny Nicholas <<a href="mailto:danny@debsinc.com" target="_blank">danny@debsinc.com</a>> wrote:</span></font></p>
<div link="blue" vlink="blue">
<div>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy">Ok – you have to be getting something or you wouldn’t get that
message. You are looking at CLI on the 1.2 or 1.4 box? If you’re looking
at the 1.4 side, you won’t see anything until a connection is made (although
you should see some kind of credential reject or something??)</span></font></p>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy"> </span></font></p>
<div>
<div class="MsoNormal" align="center" style="text-align:center"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">
<hr size="2" width="100%" align="center">
</span></font></div>
<p class="MsoNormal"><b><font size="2" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma;font-weight:bold">From:</span></font></b><font size="2" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma"> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>]
<b><span style="font-weight:bold">On Behalf Of </span></b>David Nickel<br>
<b><span style="font-weight:bold">Sent:</span></b> Wednesday, May 05, 2010 9:31
AM</span></font></p>
<div>
<div>
<p class="MsoNormal"><font size="2" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma"><br>
<b><span style="font-weight:bold">To:</span></b> Asterisk
Users Mailing List - Non-Commercial Discussion<br>
<b><span style="font-weight:bold">Subject:</span></b> Re: [asterisk-users] Hash
Dial Pattern Problems</span></font></p>
</div>
</div>
</div>
<div>
<div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt"> </span></font></p>
<p class="MsoNormal" style="margin-bottom:12.0pt"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">Nothing..goes
directly to "The person you are calling is unavailable". </span></font></p>
<div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">On Wed, May 5,
2010 at 9:46 AM, Danny Nicholas <<a href="mailto:danny@debsinc.com" target="_blank">danny@debsinc.com</a>> wrote:</span></font></p>
<div link="blue" vlink="blue">
<div>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy">Set verbose to 5 and see if you get a CLI output.</span></font></p>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy"> </span></font></p>
<div>
<div class="MsoNormal" align="center" style="text-align:center"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">
<hr size="2" width="100%" align="center">
</span></font></div>
<p class="MsoNormal"><b><font size="2" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma;font-weight:bold">From:</span></font></b><font size="2" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma"> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>]
<b><span style="font-weight:bold">On Behalf Of </span></b>David Nickel<br>
<b><span style="font-weight:bold">Sent:</span></b> Wednesday, May 05, 2010 8:39
AM<br>
<b><span style="font-weight:bold">To:</span></b> Asterisk
Users Mailing List - Non-Commercial Discussion<br>
<b><span style="font-weight:bold">Subject:</span></b> Re: [asterisk-users] Hash
Dial Pattern Problems</span></font></p>
</div>
<div>
<div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt"> </span></font></p>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">I am sorry it is
not 1.6 but 1.4.30 (argh.. to many boxes)</span></font></p>
<div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">The other box is
running 1.2.1</span></font></p>
</div>
<div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">Thanks,</span></font></p>
</div>
<div>
<p class="MsoNormal" style="margin-bottom:12.0pt"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">David</span></font></p>
<div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">On Wed, May 5,
2010 at 9:28 AM, Danny Nicholas <<a href="mailto:danny@debsinc.com" target="_blank">danny@debsinc.com</a>> wrote:</span></font></p>
<div link="blue" vlink="purple">
<div>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy">Which 1.6 are you running? I dropped my 1.6.1.6 back to
1.4.30 because my other 2 1.4.30 boxes wouldn’t talk to it properly.</span></font></p>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy"> </span></font></p>
<div>
<div class="MsoNormal" align="center" style="text-align:center"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">
<hr size="2" width="100%" align="center">
</span></font></div>
<p class="MsoNormal"><b><font size="2" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma;font-weight:bold">From:</span></font></b><font size="2" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma"> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>]
<b><span style="font-weight:bold">On Behalf Of </span></b>David Nickel<br>
<b><span style="font-weight:bold">Sent:</span></b> Wednesday, May 05, 2010 8:23
AM<br>
<b><span style="font-weight:bold">To:</span></b> <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>
<b><span style="font-weight:bold">Subject:</span></b> [asterisk-users] Hash
Dial Pattern Problems</span></font></p>
</div>
<div>
<div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt"> </span></font></p>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">I have two
Asterisk boxe. One is running 1.6 and the other 1.2 <br>
<br>
The users on the 1.2 system press # plus a local 7 digit number to place local
calls through the trunk to the 1.6 box. <br>
<br>
For some reason this dial pattern fails right away with
"unavailable". There is no activity in the CLI. Other patterns for
the trunk work just fine. <br>
<br>
Dial pattern: <br>
#|. or #|NXXXXXX <br>
<br>
exten => _#.,1,Dial(IAX2/trunk/${EXTEN:1},30,r) <br>
exten => _#.,2,Congestion <br>
<br>
I have been beating my end with the problem for three days. Any suggestions
would be much appreciated. </span></font></p>
</div>
</div>
</div>
</div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt"><br>
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