[App_rpt-users] Autopatch operation on Allstarlink
ajp26 at buffalo.edu
Sat Nov 10 10:09:39 EST 2012
I have the autopatch function working, except that our club router is terrible breaks this. I use les.net for VoIP termination, because they're cheap, and very flexible in how they can be setup.
Just like with rpt/IAX, you must configure sip.conf with the correct parameters for your adapter. If you are using a machine with multiple adapters, you may have to specify an address but that may be more for inbound calls than outbound.
I have a register statement to register to the remote proxy, since I have a dynamic IP address. Under this, I have a stanza to define that connection. The title of the stanza is going to be whatever you'd use to reference that connection with the SIP channel driver when placing a call. So, if it looks like [sip_provider] , when you place a call, you'd dial SIP/sip_provider so it knows what to do. I use g7.11 ulaw. I don't know what the audio from the USB fob is considered to the system, I would imagine it's PCM or something that it is transcoding, so whatever you choose, make sure it works with your provider.
Extensions.conf needs to be edited as well if you're doing this with your own provider. You set the area code in the web portal, or, apparently it uses a global variable HOMENPA that's set at the top. If you are not in the US/Canada, you will probably not find this variable, nor most of the autopatch extensions, helpful. I can't really help you with the configuration there.
By default, my rpt.conf listed "radio" as the context under my node stanza. As I mentioned in another mail, that's going to send everything to the check_route stanza that's right under the radio one in the extensions.conf . If you're getting the "the number you have dialed is not in service" message, this may be your culprit. It will absorb things that it doesn't consider formatted right or permissible to call. Otherwise, it goes back up to the top to the pstn-out stanza.
That's going to look like this:
This is going to take the properly formatted NANP number and dial it over that defined SIP peer. Then hangup once the call terminates from the provider. If you are using IAX2 you'd simply change that first step there to reflect that.
That's all I did. You will need to make sure that 5060 UDP outbound works, as well as the ephemeral RTP range.
If you have a more specific question I'd be willing to give it a shot but I'm not a digium pro.
>Date: Wed, 07 Nov 2012 18:01:52 -0500
>From: Eric Meth - ve3ei <ve3ei at iescomm.ca>
>To: app_rpt-users at ohnosec.org
>Subject: [App_rpt-users] Autopatch operation on Allstarlink
>Message-ID: <509AE860.5040408 at iescomm.ca>
>Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>I have been fighting with activating the autopatch on one of my nodes.
>I have obtained a SIP account an IAX2 account from a provider. I have
>received their example scripts, and I have reviewed the ACID System
>Administrators Guide to no avail.
>I would like from someone who has successfully activated the autopatch
>The reverse patch will be a future project.
>Eric Meth - ve3ei
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