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We use Direct Routing (CUCM with MS Teams). Being able to assign a phone number directly on the Teams Admin Center web page has never worked for us in the past. The Assigned Phone Number field was always grayed out (but we COULD assign a Voice Routing Policy with no issues). BUT...Several...

I'll check that out. I was expecting it might be a couple lines. One with from-internal used to ID the ext's in question and another line with the set calleid stuff.

Yes, they allow it. Do you have an example of the proper code I'd use?

Hi, I believe this can be done with the from-internal statement and specifying what extensions I'm interested in...but here's what I want to do... If a certain group of extensions (maybe I note them individually in one line or create multiple entries, one for each different ext?) calls outside...

We have 3 asterisk servers but I'm having trouble configuring the 3rd one exactly like the others. The guy that setup the others is out due to an injury. In the meantime I noticed we might be able to do SIP right out of the Avaya. I can assign the trunk group and the associated signal group...

Stupid question. We're on CM 3.1.4 at the moment (S8710). I noticed we have the option to administer SIP trunks, BUT we don't have any capacity (no licenses) to use SIP trunks at the moment. Is it as simple as getting an order in with Avaya to enable some SIP trunks for us? If so, any idea about...

Hey it's working now! I've been playing around with the h323 part of the dial string. I'm sure I tried it both ways, OOH323 and simply H323 in the dial string for the trunk....but just now switching it back to OOH323 it works even though I'm using the h323.conf file and not the ooh323.conf...

Well I guess this is better than before. I get 'all circuits are busy now, please try your call again later'....a message from the asterisk server. I'm actually watching for a specific ext in the route so I'm sure it's matching that one entry. If I remove the entry in the route so it doesn't...

So the outbound dial rules would cover the ext I'd be dialing on the Avaya right?...so I don't need to create a new ext in the Asterisk for this? Just create the trunk in the GUI, create the route so that someone dialing the exact ext or the range of ext's would be forced to this route and the...

I need specific info :) I.E. EXACTLY what to type in (based on my above h323 config) on the custom trunk and/or custom extension fields on the FreePBX interface. I'm a novice as you might guess when it comes to Asterisk and even more so on the FreePBX/AsteriskNow/Trixbox solutions.

Hi, I've got an Avaya Communication Manager R3 PBX successfully making calls TO the Asterisk (Trixbox/FreePBX install) but I can't seem to get the settings right to have a SIP softphone registered to the Asterisk call the Avaya system. Everything is great if the Avaya calls the Asterisk, but...

Thanks

Our usual asterisk guy is out sick. I'm trying to decipher exactly what these entries below mean. I understand some of it because it's obvious, but other parts I'm not so sure. If you can, would you please break this down for me and tell me what each entry is doing? I edited the real phone...

I did install asterisknow on a spare machine to play around with it and get more familiar. Ah...so that's why the changes I made in freepbx on my test system didn't show up in the normal asterisk directories. Now that makes more sense. For the very basic config we're using it probably doesn't...

More info. We have a very basic Asterisk config and pretty much only use it to convert H.323 to SIP and vice versa. What I'm looking to do is leave everything the way it is but simply add a web interface for administration purposes...if possible.

Is there a built-in web interface that can be enabled for Asterisk? I'm sort of filling in for someone that's out sick and I don't know as much about the Asterisk...so if there's a web interface/gui I can enable that would make life much easier on me than using the command line. Any input is...

I see no mention of what operating system is used for Cisco Call Manager release 7 / latest version. Is it Linux now or still Windows or both available?

We’re currently at CM release 3.1.4 on our main S8710 servers. I’m wondering what the drop-dead date is on when Avaya will not allow any more sales of R3 or at what point a newer release won’t work with our existing configuration. For example, let’s say we want to add a new location that will...

Ok, we solved this using tenant partitioning. All is well now. Basically we put the IP trunks in tenant partition 3 and the PSTN trunks in tenant 2. Phones remain in tenant 1 with full access to all partitions. We told tenant 2 it can't call tenant 3 (the IP trunks). We told tenant 3 it...

In order for a speakerphone to not clip the conversation, it must be a true full-duplex system, meaning that audio must flow continuously from each end to the other. Conventional speakerphones do not do this, they operate by switching audio back and forth, and so cause clipping. Only...