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
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
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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participants (3)
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Avi Marcus
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georg＠riseup.net
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Vishal Kakkar