We want to make outbound call to our partner company, but the only address they’ve provided is sip URI address ([email protected] style). I have fully configured FreePBX 12.0.76.3 with Asterisk version - 11.21.2, but sip URI addresses is new for me and I had no idea about this technology until today. I’ve done a lot of searches about how to configure Asterisk, but there is a little information about this. So I want to ask you help me, if someone has any experience about this.
Here’s what I’ve done already: I’ve enabled SRV Lookups on Asterisk, then I’ve created dial-plan pattern which matches any digit and set this rule as a default outbound route for this particular extension. But anyways, when I’m trying to make call from the device (Grandstream VC3200 Video Caller), it says, that it couldn’t find dial plan that will match this string and it make sense as the pattern any character - . is actually any numeric character and not alphabet, so what should I do? How can I teach asterisk (FreePBX) how to treat SIP URIs?
As I said above, my goal is only to make outside call, not for incoming. I already tried to call them to ask normal phone number, but they’ve told me that they have special policy and the only way to connect to their Cisco VoIP server from the outside of company, is to use URI address.
Thanks for your reply. Yes, I’ve read that thread, but as I’m using FreePBX distro, everything I’ve configured via WebGUI, in the past I had a try to changes directly in config files, but than system was broken and stopped working, as they have custom configs. So, yes I’ve read that post, but how to add alphabetic patterns in dial-plan of outbound route, I have no idea. As in WebGUI there’re only special characters for numeric wildcarding (lke X, Z and etc), but I will give a try to there somthing like [a-z] and this [A-Z], maybe it’ll work.
No, I just tried to put [a-z] wildcard into “Match Pattern” field, but system does not recognize it and after applying settings, it has just disappeared.
As I said before, I had a try to change something in dial-plan configuration manually, but I had trouble and we were unable to make any outbound calls until I didn’t deleted what I changed. Though I’m Linux system administrator and have a big experience with working in Linux terminal I’m afraid to change something in FreePBX’s config files, I’d like to stay with WebGUI for system configuration, if it’s possible.
I don’t not how, but I was able to make outbound call via URI address. I was trying to test on some online test servers and maybe the problem was in it, maybe they were not working. Yesterday I made call directly to the partner company’s Cisco virtual meeting room and it completed successfully. We were able to listen them and receive Video also.
Just in case if someone will have the same problem, I’ll explain what exactly I changed. So, as I said above, we have Grandstream GVC3200 Video Conference device which registered to our FreePBX server as a normal CHAN SIP extension. Then I just enabled in Asterisk SIP settings SRV Lookups. After this, I’ve created outbound route with dial-plan any-any (I just put a point in pattern - ‘.’) and then I’ve put this route as a default route for this extension. Then I’ve login into web interface of the device (as it’s impossible to dial alphabetic symbols directly from the device) and from web I’ve pasted URI address of other party and made video call successfully.
I’ve never done it, but I think setting up a “custom extension” and using the SIP URI as the dial string would be a good start. That way, you can set up extensions on your system that people dial like a normal phone and connect to the remote via the SIP URI.
Hello everyone - i too am seeking a way to use the sip uri that i have in my address book, it seems quite cazy that we set up internet telephony - enable incoming sip and then have to create bespoke extensions for sip URI outgoing. I need to just be able to call them directly from my address book - i actually have an increasing number of sip uri that i can use - into microsoft 0365 for instance (and probably google too).
As my softphone (Bria) has its own dial plan - and it in itself is perfectly able to make direct calls with a SIP URI - my intention is to modify the BRIA dial plan to route SIP URI out directly and anything that looks like a phone number out via the FreePBX - if anyone else has done something like this i would certainly like to hear.
i WOULD of course prefer to just be able to route all calls via FreePBX as it just make everything more controlled.
I tried something like that a while ago and I had a weird error about having an INVITE problem…
The SIP URI I was trying to call is actually one of my sub accounts with my main VoIP provider and that specific provider has free calls between different accounts and sub accounts… They probably allow calls from everywhere on those URIs because I have no way to lock them down as far as I can see…
I will try to dig up the specific error message I had when I tried that… Might be useful to know what is involved in getting this working as the question will surely come up again in the future…
