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  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12495">
    <title>Re: Trouble dialing in to SipX from PSTN</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12495</link>
    <description>
You can run multiple instances on the same host; I've run 5 sipXecs's
simultaneously.  In principle it's simple, in practice it's messy.  The
port numbers are controlled by a configuration file.  (The exact method
is being changed, but the capability remains -- down in the bowels,
sipXecs listening ports can be reconfigured.)  The mess is that you have
to give each sipXecs instance a different tree of directories to store
its stuff in.  You can do that by creating a chroot environment, or by
rebuilding the code from source, providing "./configure
--prefix=/root/of/sipXecs/execution/file/tree".

Dale


</description>
    <dc:creator>Dale Worley</dc:creator>
    <dc:date>2008-12-01T22:14:10</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12494">
    <title>Re: Trouble dialing in to SipX from PSTN</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12494</link>
    <description>
On Mon, 2008-12-01 at 10:54 -0800, Michael LeBlanc wrote:

yes, with great care - this really isn't a "supported" thing to do, but
with manual configuration it can be made to work.


no.

What are you trying to do that you need this?

</description>
    <dc:creator>Scott Lawrence</dc:creator>
    <dc:date>2008-12-01T20:24:07</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12493">
    <title>Re: Trouble dialing in to SipX from PSTN</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12493</link>
    <description>Good point -- OpenSIPS is a piece I'd like to remove. I did get 
sipXproxy to start on a different port -- but then I had trouble 
configuring X-Lite to connect to the proxy on anything other than 5060.

Can I run multiple instances of sipXproxy on the same host, or have the 
same instance listen on multiple ports?

-----Original Message-----
From: Scott Lawrence [mailto:scott.lawrence&lt; at &gt;nortel.com]
Sent: December 1, 2008 10:39 AM
To: Michael LeBlanc
Cc: sipx-users&lt; at &gt;list.sipfoundry.org
Subject: Re: [sipx-users] Trouble dialing in to SipX from PSTN


On Mon, 2008-12-01 at 10:01 -0800, Michael LeBlanc wrote:
 &gt; I figured out where I went wrong. I was pointing the gateway directly at
 &gt; the SIP Registrar, rather than the SIP Proxy (5060). Once I had the
 &gt; gateway talking to the SIP Proxy, the autoattendant and extensions with
 &gt; aliases set to DiDs were found by SipX.
 &gt;
 &gt; The complicating factor is that our gateway has to talk to SipX on a
 &gt; non-standard port, say port 6060, rather than 5060. So I had to set u</description>
    <dc:creator>Michael LeBlanc</dc:creator>
    <dc:date>2008-12-01T18:54:21</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12492">
    <title>Re: Trouble dialing in to SipX from PSTN</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12492</link>
    <description>
On Mon, 2008-12-01 at 10:01 -0800, Michael LeBlanc wrote:

It sounds like you've got workarounds on your workarounds

Getting sipXproxy to run on a different port is not a big deal.


</description>
    <dc:creator>Scott Lawrence</dc:creator>
    <dc:date>2008-12-01T18:38:41</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12491">
    <title>SipX Active Directory Sync/SIP Password Storage</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12491</link>
    <description>I was wondering if anyone has gone the route of AD synchronization for 
SipX user information. In particular, is it possible to actually 
synchronize a user's username/Sip-password with their AD 
username/password? I can think of reasons why this wouldn't be desirable 
(especially with passwords being stored on devices), but I was wondering 
if it's a possibility. My understanding is that Active Directory won't 
let you read a password attribute via LDAP, even if you're binding as a 
privileged user.

Also, does SipX store sip-passwords in the clear? If so, are there any 
ways to hash it -- or is it generally accepted that the sip-password is 
a low security token, just by virtue of the way it's used (stored in 
text files on devices, etc...).

Cheers,

Mike
</description>
    <dc:creator>Michael LeBlanc</dc:creator>
    <dc:date>2008-12-01T18:25:51</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12490">
    <title>Re: Trouble dialing in to SipX from PSTN</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12490</link>
    <description/>
    <dc:creator>Tony Graziano</dc:creator>
    <dc:date>2008-12-01T18:05:33</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12489">
    <title>Re: Trouble dialing in to SipX from PSTN</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12489</link>
    <description>I figured out where I went wrong. I was pointing the gateway directly at 
the SIP Registrar, rather than the SIP Proxy (5060). Once I had the 
gateway talking to the SIP Proxy, the autoattendant and extensions with 
aliases set to DiDs were found by SipX.

The complicating factor is that our gateway has to talk to SipX on a 
non-standard port, say port 6060, rather than 5060. So I had to set up 
OpenSIPS to perform some routing, taking the requests on port 6060, and 
relaying them to SipX on 5060 (and rewrite the contact so it was 
recognized by SipX).

Dialing-in from the PSTN to a logged-in SipX user now works perfectly. I 
still can't get to voicemail/aa from the PSTN, but I'm pretty sure 
that's firewall related (the user is found, but RTP traffic isn't 
getting through).

Cheers,

Mike

*
From:* Chris Jones [mailto:cjonesmo&lt; at &gt;gmail.com]
*Sent:* November 28, 2008 8:41 PM
*To:* Michael LeBlanc
*Subject:* Re: [sipx-users] Trouble dialing in to SipX from PSTN

 

Make sure that you created the gateway with th</description>
    <dc:creator>Michael LeBlanc</dc:creator>
    <dc:date>2008-12-01T18:01:28</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12488">
    <title>Re: Trouble dialing in to SipX from PSTN</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12488</link>
    <description>
We'll need at least a proper snapshot (run 'sipx-snapshot') to be able
to help.

Dale


</description>
    <dc:creator>Dale Worley</dc:creator>
    <dc:date>2008-12-01T16:43:48</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12487">
    <title>Re: sipx (HA) + CentOS from single CD</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12487</link>
    <description>
On Mon, 2008-12-01 at 11:16 +0530, Vikas Sharma wrote:

You can use the 'tcpdump' program from the shell; it uses the same
packet-capture library but does not need a gui.

You can also get information on the SIP messages using the sipXecs logs
- see:

http://sipx-wiki.calivia.com/index.php/Display_SIP_message_flow_using_Sipviewer#Getting_SIP_Messages_to_display

</description>
    <dc:creator>Scott Lawrence</dc:creator>
    <dc:date>2008-12-01T13:31:23</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12486">
    <title>Re: sipx (HA) + CentOS from single CD</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12486</link>
    <description/>
    <dc:creator>Picher, Michael</dc:creator>
    <dc:date>2008-12-01T09:31:16</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12485">
    <title>sipx (HA) + CentOS from single CD</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12485</link>
    <description/>
    <dc:creator>Vikas Sharma</dc:creator>
    <dc:date>2008-12-01T05:46:42</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12484">
    <title>compiling on debian (lenny)</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12484</link>
    <description/>
    <dc:creator>zorg</dc:creator>
    <dc:date>2008-11-30T16:17:06</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12483">
    <title>Re: same operator number for each group</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12483</link>
    <description/>
    <dc:creator>Picher, Michael</dc:creator>
    <dc:date>2008-11-30T13:18:47</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12482">
    <title>Re: sipXecs ISO install -- I blew away my named</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12482</link>
    <description>Gracias!

Thanks -- should help.

On Nov 29, 2008, at 8:37 AM, Scott Lawrence wrote:


</description>
    <dc:creator>Chad Leigh - Pengar Enterprises Inc</dc:creator>
    <dc:date>2008-11-29T21:37:50</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12481">
    <title>Re: same operator number for each group</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12481</link>
    <description/>
    <dc:creator>Tony Graziano</dc:creator>
    <dc:date>2008-11-29T15:57:54</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12480">
    <title>Re: same operator number for each group</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12480</link>
    <description/>
    <dc:creator>Tony Graziano</dc:creator>
    <dc:date>2008-11-29T15:56:05</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12479">
    <title>Re: same operator number for each group</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12479</link>
    <description>
On Sat, 2008-11-29 at 17:24 +0200, dimitris(yahoo) wrote:

No - there's no way to do that now.



_______________________________________________
sipx-users mailing list
sipx-users&lt; at &gt;list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users
Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users</description>
    <dc:creator>Scott Lawrence</dc:creator>
    <dc:date>2008-11-29T15:50:33</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12478">
    <title>Re: sipXecs ISO install -- I blew away my named</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12478</link>
    <description>
On Thu, 2008-11-27 at 03:07 -0700, Chad Leigh - Pengar Enterprises Inc
wrote:

See 

  sipx-dns --help

that will generate the DNS records you need - insert them in a standard
bind zone file template.


</description>
    <dc:creator>Scott Lawrence</dc:creator>
    <dc:date>2008-11-29T15:37:22</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12477">
    <title>same operator number for each group</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12477</link>
    <description/>
    <dc:creator>dimitris(yahoo</dc:creator>
    <dc:date>2008-11-29T15:24:20</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12476">
    <title>Trouble dialing in to SipX from PSTN</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12476</link>
    <description>Hi,

I've been working on getting a SipX instance up and running, and I've 
hit a snag with inbound calls to SipX from the PSTN. As a bit of 
background, I've had to change the SIP registrar port to listen on 5062 
rather than 5070 -- but that's really the only deviation from the 
standard SipX RPM install.

The situation is that I can see the SIP invite messages coming from the 
PSTN gateway to SipX, but SipX responds that the user doesn't exist (404 
Not Found). The extension does exist, and works when I call internally 
within SipX from a different account. I've sanitized the sip registrar 
logs to remove the serverIP, but you can see SipX looking for the user 
(72301) and not finding it.

Any thoughts would be much appreciated ... let me know if more details 
(from sipviewer, etc...) would be useful.

Cheers,

Mike

SipRedirectServer-13:B6700B90:SipRegistrar:"SipRedirectServer::handleMessage 
Start processing redirect message 0: 'INVITE' 'sip:72301&lt; at &gt;&lt;IP&gt;:5062'"
SipRedirectServer-13:B6700B90:SipRegistrar:"</description>
    <dc:creator>Michael LeBlanc</dc:creator>
    <dc:date>2008-11-29T00:26:02</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12475">
    <title>Re: mr clueless here -- cannot dial out throughSIPgateway</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/12475</link>
    <description/>
    <dc:creator>Picher, Michael</dc:creator>
    <dc:date>2008-11-27T11:27:01</dc:date>
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