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The one that says "Sangoma" on it.

Why do you have UDP 5060 allowed from the outside world? Stop that and the problem will go away. Look at your call records - if you don't have strong passwords on your SIP endpoints, it's quite possible you've been owned.

Firewall issues cause this behavior. In that case (may or may not be your case), the SIP responses are not returning. Asterisk will destroy a SIP conversation if it doesn't get an ACK from the other end. The default setting is 3600ms IIRC. Look for any kind of SIP settings in your FW...

You could do a simple PRI interface between the two. That would give you 23 channels of voice per PRI. I have worked with folks in the past on connecting Call Manager to Asterisk, but the projects never went much past a pipe dream once they found out they would have to license some extra...

Heh. Shows you what I use, eh?

You do NOT want to do that. First of all, these should be UDP, not TCP. Opening TCP/10000 exposes Webmin. NOT good. Second, you only need 5060 UDP for SIP, and then (maybe) the 10K->20K for RTP.

Sometimes the easiest solution is the simplest... like just recording the conversation via an external device: http://www.1topstore.com/product_info.php?language=en&currency=USD&products_id=11905 ~$5, problem solved.

question 1: Yes, you can do this. It helps if your doctors' homes have static IP's. If not, perhaps the ISP has a range you can stick to... a small one, preferably. You can restrict extensions from registering except from specific IP's, which is a big + if you have to open them up. Failing...

You can only use rfc2833 on sip-based trunks/lines. All DTMF on analog devices, is "inband" due to the nature of the signaling (there is no sideband to send DTMF - it comes in the same channel as the audio for the call does). Other DTMF options are only for digital-type connections, like SIP.

There's a memory overflow bug in the sidecars. If you have more than a certain number of extensions (55, I think?), it will do that. We ran into that problem and eventually just pulled all the sidecars and went to a PC-based display (HUD/HUD2/iSymphony) to get around the issue. From what I...

One other comment I just noticed: I would NEVER suggest having different brand cards in the same system. I know it technically can be done, but I've never seen anyone have a good experience with that kind of setup. If you have brand A in box A, stick with it, or change everything to brand B...

I don't use the Rhino card for anything more than a timing source. It's one of their 8-port FXS/FXO card with no modules in it. That being said, I have used a number of Rhino cards in the past - I have had audio problems, but have always been able to clear them up. All of these installs...

It will work, but IIRC you might need the "beta" drivers from Sangoma for support. Anyway, Purple has Dahdi 2.4, and I think the current listed drivers only support up to 2.2. In reading their site, you need at least 3.5.16+ for 2.4 support.

Searching seems to help. http://pbxinaflash.com/community/threads/trunk-ip-addresses.10536/#post-66874

We were recently beating our heads against the proverbial wall :banghead: First off, this will ONLY apply if you have a Rhino Ceros unit - meaning *their* server hardware. This does NOT apply if you just have one of their PRI/POTS cards in a whitebox. When we'd go to run an update-source...

Look - these gateway devices are extremely complicated things. They have to be because there is no driver telling them what to do. If you're not willing to get your hands dirty on the raw configs, I'm not so sure it's the right answer for you. If you want something that's better...

Did you even bother searching before you asked? http://pbxinaflash.com/community/threads/audiocodes-mp-118-fxo-gateway.534/#post-2931

Sangoma, specifically, has a lot more experience than most other vendors when it comes to telephony devices. They also do more than "just" Asterisk stuff. But when you boil it down, it's like asking why would you buy a Land Cruiser if you can buy a cheap Kia that holds the same amount of...

Same thing goes for extensions if you want to allow from multiple sources as well. Just separate IP's and/or ip/subnet with "&"

I never use trunk CID's. I always set the CID on the extension and let it be handled that way - sometimes my routing will send a call as a failover event, and I want the extension to give the CID, not the trunk. Try clearing the CID setting (on all trunks) and, with the CID set on the...