[Freeswitch-users] Setting Up an Sip Trunk without Authentication
Dave
dave at clancysystems.com
Fri Jun 17 00:27:32 MSD 2011
Yes, I am only testing inbound. When I call from our office phone (not connected to FreeSWITCH, it's on a trunk from broadvox) I Get a busy signal and at that moment FreeSWITCH shows...
"SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP.
If I call from my cell phone I get the Operator Message "your call did not go through" and at that moment the same thing shows in FreeSWITCH
"SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP.
----- Original Message -----
From: Michael Collins
To: FreeSWITCH Users Help
Sent: Thursday, June 16, 2011 2:11 PM
Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication
On Thu, Jun 16, 2011 at 1:02 PM, Dave <dave at clancysystems.com> wrote:
Under "domains" in acl.conf.xml I added
<node type="allow" value="xxx.xxx.xxx.xxx"/>
The correct syntax is:
<node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
x's are the IP address it's just a single address not a CIDR.
And I created the file velocity_did.xml with the following code..
<include>
<extension name="velocity_did">
<condition field="destination_number" expression="^(2064001950)$">
<action application="set" data="domain_name=$${domain}"/>
<action application="answer"/>
<action application="lua" data="cme_ivr.lua channel_name"/>
</condition>
</extension>
</include>
I still just get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP, but nothing more.
Wait - can you explain *exactly* what you're doing for testing? Are you trying to dial out via Velocity and back in to your DID? If so then you have 2 completely different things you need to set up. The instructions I gave were only for inbound DID, so use a cell phone or something to test that.
For outbound it seems like they are sending you an auth challenge, which means they need to give you a username and password as well as a host/ip. You need to create a gateway, preferably in your external profile. Just be sure to set the "register" param to false since they are not expecting you to register with them.
-MC
Do I need to change the line ..
<action application="set" data="domain_name=$${domain}"/>
to the IP address? If so what syntax?
They have no Domain Name.
Thanks again for your help.
Dave Goodwin
----- Original Message -----
From: Michael Collins
To: FreeSWITCH Users Help
Sent: Thursday, June 16, 2011 12:15 PM
Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication
Dave,
If you are simply handling incoming calls then you will need to do two things:
Add Velocity's IP addr to the domains section of acl.conf.xml
Add an extension to the public context (conf/dialplan/public.xml)
Allowing a call into FS via an ACL will send it into the "public" context; from there you need to transfer it to the default context or just send it straight to your Lua script.
Also, I don't believe you need to use a "bridge" app based on the description you gave. Bridge is used to create a new outbound call leg (B leg) and connect it to the inbound call leg (A leg). If you just are handling a call with an IVR then there is no B leg needed, thus no need to bridge.
Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the "bridge" book, ironically) as it discusses some of these basic concepts that will make your FS experience a whole lot more pleasant.
-MC
On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave at clancysystems.com> wrote:
Hi,
I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls through a Velocity Networks SIP trunk. They say I should not authenticate with Username and Password. Rather, I connect directly to the their IP address.
I put
<extension name="velocity">
<condition field="destination_number" expression="^2064001950$">
<action application="bridge" data="sofia/external/$ {destination_number}@their_ip_address"/>
<action application="lua" data="cme_ivr.lua channel_name"/>
</condition>
</extension>
in the default.xml under Dialplan
When I Dial the DID from a Phone (not one connected to the FreeSWITCH server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP, but nothing more.
What I hope to do is take the incoming call from 2064001950 and route it to the Lua IVR script above.
I Appreciate any help you may offer.
Dave Goodwin
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110616/9307e607/attachment.html
More information about the FreeSWITCH-users
mailing list