[Freeswitch-users] Gateway Configuration with No Authentication
Vishal Kakkar
vishal.kakkar at gmail.com
Thu Jan 26 16:19:22 MSK 2012
Hi,
Not using username parameter in gateway is causing following OPTION message
being published-
OPTIONS sip:192.168.1.30;transport=udp SIP/2.0
Via: SIP/2.0/UDP 59.90.193.54:5080;rport;branch=z9hG4bKZ5D672rZgta9N
Max-Forwards: 70
From: *<sip:(null)@192.168.1.30;*transport=udp>;tag=ZS9p1ZN164S4Q
To: *<sip:(null)@192.168.1.30;*transport=udp>
Call-ID: a78e7d46-c2c1-122f-d9a3-0013a9866b3c
CSeq: 23455584 OPTIONS
Contact: <sip:gw+192.168.1.50 at 59.92.192.53:5080
;transport=udp;gw=192.168.1.50>
User-Agent: FreeSwitch
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Length: 0
On Thu, Jan 26, 2012 at 4:33 PM, <
freeswitch-users-request at lists.freeswitch.org> wrote:
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> Today's Topics:
>
> 1. Gateway Configuration with No Authentication (Vishal Kakkar)
> 2. Re: Gateway Configuration with No Authentication
> (georg at riseup.net)
> 3. Re: Gateway Configuration with No Authentication (Avi Marcus)
> 4. Re: Registration Failed with status DNS Error (Antonio)
> 5. mod_callcenter - ring tones when phone is ringing (Weigel, Stefan)
>
>
> ---------- Forwarded message ----------
> From: Vishal Kakkar <vishal.kakkar at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Cc:
> Date: Thu, 26 Jan 2012 14:04:40 +0530
> Subject: [Freeswitch-users] Gateway Configuration with No Authentication
> Hi all,
>
> 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.
>
> 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).
>
> 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.
>
> 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.
>
> Please help to configure FS to send option something like this-
>
>
> *OPTIONS sip:10.129.43.154:5060;ttl=0 SIP/2.0*
>
> Via: SIP/2.0/UDP 10.129.43.23:5060;
> rport;branch=z9hG4bK-
>
> 6df4-1156281802-19999-423
>
> Call-ID:
> 5aa9-1e61-7222006212322-chiloe-0 at 10.129.43.23
>
> CSeq: 1 OPTIONS
>
> Max-Forwards: 70
>
> *To: <sip:10.129.43.154;ttl=0>*
>
> *From: <sip:10.129.43.23>*
> ;tag=95ffcd055e0f78f7d5d397020e8
>
> 9288db5f2
>
> User-Agent: Dialogic-SIP/10.3.2.57 chiloe 0
>
> Contact: <sip:10.129.43.23:5060>
>
> Accept: application/ sdp
>
> Content-Length: 0
>
> i.e. They want OPTION with only IP address not user at IP. So that on
> recieving any call they can transfer it to our FS.
>
> Please Help.
> Thanks
> -Manav
>
>
> ---------- Forwarded message ----------
> From: georg at riseup.net
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Thu, 26 Jan 2012 10:08:12 +0100
> Subject: Re: [Freeswitch-users] Gateway Configuration with No
> Authentication
> He Vishal,
>
> > 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.
>
> I don't think you have to use a username. I've got a similar setup, my
> external.xml config is looking like:
>
> <include>
> <gateway name="xxx">
> <param name="proxy" value="THE.I.R.IP"/>
> <param name="register" value="false"/>
> <param name="extension" value="extension_that
> _should_be_called_in_your_dialplan"/>
> <param name="rtp-autofix-timing" value="false"/>
> <param name="caller-id-in-from" value="true"/>
> </gateway>
> </include>
>
> HTH,
> Georg
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Avi Marcus <avi at avimarcus.net>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Thu, 26 Jan 2012 11:58:56 +0200
> Subject: Re: [Freeswitch-users] Gateway Configuration with No
> Authentication
> I usually have my origination providers send to $number@$myserver:5080 so
> it goes to external and hits the public context.
> Or, if they won't, I create a user with their ACL and set the context on
> the user to public.
>
> Hmm, perhaps external gateway that doesn't register would work and be
> cleaner..
> -Avi
>
>
> On Thu, Jan 26, 2012 at 11:08 AM, <georg at riseup.net> wrote:
>
>> He Vishal,
>>
>> > 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.
>>
>> I don't think you have to use a username. I've got a similar setup, my
>> external.xml config is looking like:
>>
>> <include>
>> <gateway name="xxx">
>> <param name="proxy" value="THE.I.R.IP"/>
>> <param name="register" value="false"/>
>> <param name="extension" value="extension_that
>> _should_be_called_in_your_dialplan"/>
>> <param name="rtp-autofix-timing" value="false"/>
>> <param name="caller-id-in-from" value="true"/>
>> </gateway>
>> </include>
>>
>> HTH,
>> Georg
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
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>
>
>
> ---------- Forwarded message ----------
> From: Antonio <asilva at wirelessmundi.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Thu, 26 Jan 2012 11:05:06 +0100
> Subject: Re: [Freeswitch-users] Registration Failed with status DNS Error
> **
> Mike, thanks for the answer.
>
> 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.
>
> Shouldn't freeswitch try to resolve first to the SRV entry and then if
> none is found, try to resolve the A entry?
>
> you can find my logs in http://pastebin.freeswitch.org/18231
>
>
> 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.
> Just to be sure i also try to put in the profile all the possibilities to
> disable-srv and disable-srv503, without luck.
>
>
>
> Thanks,
> António
>
>
>
> On Wed, 2012-01-25 at 13:53 -0500, Michael Jerris wrote:
>
> 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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>
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>
>
> ---------- Forwarded message ----------
> From: "Weigel, Stefan" <Stefan.Weigel at allianz-warranty.com>
> To: "'freeswitch-users at lists.freeswitch.org'" <
> freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Thu, 26 Jan 2012 12:03:16 +0100
> Subject: [Freeswitch-users] mod_callcenter - ring tones when phone is
> ringing
>
> Hi all,****
>
> ** **
>
> 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 ?****
>
> ** **
>
> ** **
>
> Thanks and best regards****
>
> ** **
>
> Stefan****
>
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