Hi,<br><br>Not using username parameter in gateway is causing following OPTION message being published-<br><br>OPTIONS sip:192.168.1.30;transport=udp SIP/2.0<br>Via: SIP/2.0/UDP 59.90.193.54:5080;rport;branch=z9hG4bKZ5D672rZgta9N<br>
Max-Forwards: 70<br>From: <b><sip:(null)@<a href="http://192.168.1.30">192.168.1.30</a>;</b>transport=udp>;tag=ZS9p1ZN164S4Q<br>To: <b><sip:(null)@<a href="http://192.168.1.30">192.168.1.30</a>;</b>transport=udp><br>
Call-ID: a78e7d46-c2c1-122f-d9a3-0013a9866b3c<br>CSeq: 23455584 OPTIONS<br>Contact: <sip:gw+192.168.1.50@59.92.192.53:5080;transport=udp;gw=192.168.1.50><br>User-Agent: FreeSwitch<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY<br>
Supported: timer, precondition, path, replaces<br>Allow-Events: talk, hold, refer<br>Content-Length: 0<br><br><br><div class="gmail_quote">On Thu, Jan 26, 2012 at 4:33 PM, <span dir="ltr"><<a href="mailto:freeswitch-users-request@lists.freeswitch.org">freeswitch-users-request@lists.freeswitch.org</a>></span> wrote:<br>
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<br>Today's Topics:<br>
<br>
1. Gateway Configuration with No Authentication (Vishal Kakkar)<br>
2. Re: Gateway Configuration with No Authentication<br>
(<a href="mailto:georg@riseup.net">georg@riseup.net</a>)<br>
3. Re: Gateway Configuration with No Authentication (Avi Marcus)<br>
4. Re: Registration Failed with status DNS Error (Antonio)<br>
5. mod_callcenter - ring tones when phone is ringing (Weigel, Stefan)<br>
<br><br>---------- Forwarded message ----------<br>From: Vishal Kakkar <<a href="mailto:vishal.kakkar@gmail.com">vishal.kakkar@gmail.com</a>><br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
Cc: <br>Date: Thu, 26 Jan 2012 14:04:40 +0530<br>Subject: [Freeswitch-users] Gateway Configuration with No Authentication<br>Hi all,<br>
<br>
We are trying to use FS to handle incoming call from SIP provider. Provider
doesnt have any authentication because its all inbound traffic. They
have configured their SIP trunk to point to our FreeSwitch Box.<br>
<br>
We have also configured one gateway in eternal/provider.xml with the IP of
provider and disable the registration as provider is not expecting the
same(they dont have any user configured for us, as they are using their
box in trunk mode).<br>
<br>
Issue is that when an incoming call reaches to provider, we are not getting any INVITE from them and its being dropped at the IP that we have added as gateway. Reason shared is that they are expecting us to have
heartbeat with their IP.<br>
<br>
When we turn ping option, it sends OPTION message but username is
getting appended to TO and FROM uri's so that cause 404 No User found
response.<br>
<br>
Please help to configure FS to send option something like this-<br>
<br>
<br>
<span style="font-weight:bold"><b>OPTIONS sip:10.129.43.154:5060;ttl=0 SIP/2.0</b></span>
<p> Via:
SIP/2.0/UDP <a href="http://10.129.43.23:5060/" target="_blank">10.129.43.23:5060</a>; rport;branch=z9hG4bK-</p>
<p> 6df4-1156281802-19999-423</p>
<p> Call-ID:
<a href="mailto:5aa9-1e61-7222006212322-chiloe-0@10.129.43.23" target="_blank">5aa9-1e61-7222006212322-chiloe-0@10.129.43.23</a></p>
<p> CSeq: 1 OPTIONS</p>
<p> Max-Forwards:
70</p>
<p> <b>To:
<sip:10.129.43.154;ttl=0></b></p>
<p> <b>From:
<sip:10.129.43.23></b>;tag=95ffcd055e0f78f7d5d397020e8</p>
<p> 9288db5f2</p>
<p> User-Agent:
Dialogic-SIP/<a href="http://10.3.2.57/" target="_blank">10.3.2.57</a> chiloe 0</p>
<p> Contact:
<sip:<a href="http://10.129.43.23:5060/" target="_blank">10.129.43.23:5060</a>></p>
<p> Accept:
application/ sdp</p>
<p> Content-Length:
0</p>
<br>
i.e. They want OPTION with only IP address not user@IP. So that on recieving any call they can transfer it to our FS.<br><br>Please Help. <br>Thanks<br>
-Manav<br>
<br><br>---------- Forwarded message ----------<br>From: <a href="mailto:georg@riseup.net">georg@riseup.net</a><br>To: "FreeSWITCH Users Help" <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br>Date: Thu, 26 Jan 2012 10:08:12 +0100<br>Subject: Re: [Freeswitch-users] Gateway Configuration with No Authentication<br>He Vishal,<br>
<br>
> We are trying to use FS to handle incoming call from SIP provider.<br>
> Provider<br>
> doesnt have any authentication because its all inbound traffic. They have<br>
> configured their SIP trunk to point to our FreeSwitch Box.<br>
<br>
I don't think you have to use a username. I've got a similar setup, my<br>
external.xml config is looking like:<br>
<br>
<include><br>
<gateway name="xxx"><br>
<param name="proxy" value="THE.I.R.IP"/><br>
<param name="register" value="false"/><br>
<param name="extension" value="extension_that<br>
_should_be_called_in_your_dialplan"/><br>
<param name="rtp-autofix-timing" value="false"/><br>
<param name="caller-id-in-from" value="true"/><br>
</gateway><br>
</include><br>
<br>
HTH,<br>
Georg<br>
<br>
<br>
<br>
<br><br>---------- Forwarded message ----------<br>From: Avi Marcus <<a href="mailto:avi@avimarcus.net">avi@avimarcus.net</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br>Date: Thu, 26 Jan 2012 11:58:56 +0200<br>Subject: Re: [Freeswitch-users] Gateway Configuration with No Authentication<br><div dir="ltr">I usually have my origination providers send to $number@$myserver:5080 so it goes to external and hits the public context.<div>
Or, if they won't, I create a user with their ACL and set the context on the user to public.</div>
<div><br></div><div>Hmm, perhaps external gateway that doesn't register would work and be cleaner.. <br clear="all"><div dir="ltr"><span style="font-family:Verdana,Arial,Helvetica,sans-serif"><span style="font-size:small">-Avi</span></span></div>
<br><br><div class="gmail_quote">On Thu, Jan 26, 2012 at 11:08 AM, <span dir="ltr"><<a href="mailto:georg@riseup.net" target="_blank">georg@riseup.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
He Vishal,<br>
<div><br>
> We are trying to use FS to handle incoming call from SIP provider.<br>
> Provider<br>
> doesnt have any authentication because its all inbound traffic. They have<br>
> configured their SIP trunk to point to our FreeSwitch Box.<br>
<br>
</div>I don't think you have to use a username. I've got a similar setup, my<br>
external.xml config is looking like:<br>
<br>
<include><br>
<gateway name="xxx"><br>
<param name="proxy" value="THE.I.R.IP"/><br>
<param name="register" value="false"/><br>
<param name="extension" value="extension_that<br>
_should_be_called_in_your_dialplan"/><br>
<param name="rtp-autofix-timing" value="false"/><br>
<param name="caller-id-in-from" value="true"/><br>
</gateway><br>
</include><br>
<br>
HTH,<br>
Georg<br>
<br>
<br>
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</blockquote></div><br></div></div>
<br><br>---------- Forwarded message ----------<br>From: Antonio <<a href="mailto:asilva@wirelessmundi.com">asilva@wirelessmundi.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br>Date: Thu, 26 Jan 2012 11:05:06 +0100<br>Subject: Re: [Freeswitch-users] Registration Failed with status DNS Error<br><u></u>
<div>
Mike, thanks for the answer. <br>
<br>
Actually the problem is the DNS server doesn't have an A entry for the destination domain. When i add this entry to the domain, the register works nice.<br>
<br>
Shouldn't freeswitch try to resolve first to the SRV entry and then if none is found, try to resolve the A entry?<br>
<br>
you can find my logs in <a href="http://pastebin.freeswitch.org/18231" target="_blank">http://pastebin.freeswitch.org/18231</a><br>
<br>
<br>
I read something in the jira for some options to disable SRV in the profile, i look at the code and it seams to be enabled by default. <br>
Just to be sure i also try to put in the profile all the possibilities to disable-srv and disable-srv503, without luck. <br>
<br>
<br>
<br>
Thanks,<br>
António<br>
<br>
<br>
<br>
On Wed, 2012-01-25 at 13:53 -0500, Michael Jerris wrote:
<blockquote type="CITE">
<pre>Is it even sending the request? try turning on nua and sresolv debug in sofia to see more about what is going on?
Mike
On Jan 25, 2012, at 11:37 AM, Antonio wrote:
>
> Hi
>
> I have a problem registering a gateway, the return error is DNS error [503].
>
> My DNS is OK, i can dig to the destination domain in question, and even when o do sofia_dig in fresswith it resolves the domain without problems.
>
>
> how does freeswitch resolves the destination domain?
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</pre>
</blockquote>
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<br><br>---------- Forwarded message ----------<br>From: "Weigel, Stefan" <<a href="mailto:Stefan.Weigel@allianz-warranty.com">Stefan.Weigel@allianz-warranty.com</a>><br>To: "'<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>'" <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br>Date: Thu, 26 Jan 2012 12:03:16 +0100<br>Subject: [Freeswitch-users] mod_callcenter - ring tones when phone is ringing<br>
<div link="blue" vlink="purple" lang="DE">
<div>
<p class="MsoNormal">Hi all,<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal"><span lang="EN-US">we’re using successfully mod_callcenter. If a caller enters the queue, he/she is getting default moh sound. The sound is also playing when the call is passed to an agent and the phone is ringing. Is it possible to play
a normal ringtone when a call is passed to an agent ?<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p>
<p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p>
<p class="MsoNormal"><span lang="EN-US">Thanks and best regards<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p>
<p class="MsoNormal"><span lang="EN-US">Stefan<u></u><u></u></span></p>
</div>
</div>
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