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@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@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@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@gmail.com> To: freeswitch-users@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@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@IP. So that on recieving any call they can transfer it to our FS.

Please Help. Thanks -Manav

---------- Forwarded message ---------- From: georg@riseup.net To: "FreeSWITCH Users Help" <freeswitch-users@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@avimarcus.net> To: FreeSWITCH Users Help <freeswitch-users@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@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

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---------- Forwarded message ---------- From: Antonio <asilva@wirelessmundi.com> To: FreeSWITCH Users Help <freeswitch-users@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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---------- Forwarded message ---------- From: "Weigel, Stefan" <Stefan.Weigel@allianz-warranty.com> To: "'freeswitch-users@lists.freeswitch.org'" < freeswitch-users@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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