similar to: Any phone capable of displaying real time queue statistics?

Hi List, Does anyone know of a good dual wan router that can handle SIP well and can failover between connections if there is a SIP issue on one of the lines (meaning there still is a connection however there isnt enough bandwith or sip packets arent going thru etc.) ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

I testing gluster (the downloadable vmdk) under VMware Vsphere (ESX 4.0). The screen size is stuck at 800x600. I've tried increasing the size in the guest settings for vmware but that doesn't help. Some of the command buttons are off the screen and are inaccessible.

The bar is getting raised yet again http://www.voipmonitor.net/2007/06/05/New+Features+And+Services+For+Pack et8+Virtual+Office.aspx I personally use Snapanumber $30 or there abouts (after trialing a few other TAPI solutions and finding them sub-par) and think it's a great product but interesting to see how more people are expecting desktop/phone integration applications. Does anyone

On Mon, Apr 13, 2020 at 10:45 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Mon, Apr 13, 2020 at 11:38 AM Dovid Bender <dovid at telecurve.com> wrote: > >> Josh, >> >> What should Asterisk do if one of the real time methods fail? I have in >> extconfig.conf >> musiconhold => curl,http://localhost/moh.php,1 >> musiconhold =>

Josh, What should Asterisk do if one of the real time methods fail? I have in extconfig.conf musiconhold => curl,http://localhost/moh.php,1 musiconhold => mysql,db-east,asterisk_moh,2 If the first server sends back a 404 it does not go to the second connection. Shouldn't a 404 be considered a failure and it should then move over to the next rt engine? On Thu, Jan 2, 2020 at 7:06 AM

Hi, Is there any way in Asterisk to have multiple forms of real time for the same object? For instance my main source for real time is MySQL. I want a fail over that if a mailbox is say not in the MySQL database for Asterisk to try via curl. TIA. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061207/92c7f946/attachment.htm

Good morning, I have a Polycom 601 with two side cars. I created a list of contacts in XML and it shows up on the side cars exaclty how I set it up in the XXXXXXXXXXXX-directory.xml file (in the order that I wanted it etc.). However when I hit the directories button and then contact directory I see the list in alphabetical order based on the last name. I want it to show up in this list as well in

I am having major problems with the first run of Snom 360s that rolled out last month. I am working with the US vendor and they in turn are working with Snom but I wanted to see of anyone else was using these or having issues with them. Issues: Speakerphone/Hands Free volume spikes up and down during a call. You have to manually set the volume during every call. This makes it totally unusable.

Does anyone know if there is a way to get a DB or any other means to see if I can see if a call is coming from a cell phone or not. If I am able to see if it is cellular or not is there any way to see aprox. what area the phone is in (I know this wont be simple but would it work if I have an agreeement with the cell phone companies) ? This is for the US. Thanks. Dovid -------------- next part

Hi List, I was thinking the other day that even with T.38 there are still some issues with faxing. I was thinking of a protocol that instead of just sending down the fax tones an ATA or "VOIP fax machine" would get the entire fax convert it into some sort of image and pass it down the line to the receiving end. I got the idea from RFC2833. Yes I know that fax machines send bit by bit and

Hi List, I am using asterisk 1.2.14 with real time and I am trying to send the email to more than one email address. In that field I put in user1@domain.com;user2@domain.com. When the call goes to VM I see in the CLI: uniqueid => 17 customer_id => 0 context => techmast mailbox => 14 password => 1234 fullname => Sales and Service email => user1@domain.com email =>

Hi List, I have a client that has an interesting request. He wants to have people call in to a conference room and all be able to talk however they should not hear each other. There should be admin access to for one user to call in and be able to listen in to everyone that is talking (they may want this admin to be able to talk to). Any ideas ? Thanks. Dovid -------------- next part

Does anyone know if it is possible to change the packetization time in Asterisk ? I was told by a client of mine that adjusting this with using G729 can greatly lower the amount of bandwidth used. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070819/b0cc470f/attachment.htm

Hi List, I wanted to know if anyone knew of a way with asterisk to "switch the voice" of a caller from male to female or vice versa. Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070426/2d483875/attachment-0001.htm

Hi everyone, sorry if this is already written somewhere: Is it at all possible to connect a Windows XP (professional ed.) client to a Samba server running 2.2.3? I got this far: After logging into the client locally I can mount the server shares by hand, but the domain logon fails, saying 'The domain controller is not 'availabe' (or similar, this is translated from the german error

Hi, I am currently playing with DD-WRT and I like it. I am looking for something that is more "SIP Aware". Anyone know one those that are out there ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071126/eb28ce44/attachment.htm

Is anyone out there using Druid? After the switchbox announcement today I've been looking into some other gui's and as I'll probably do a trial install this weekend of the free switchvox iso but I thought I'd ask is there any other guis I should be burning trial ISO's of as well? Regards, Dean Collins Cognation Pty Ltd dean at cognation.net <mailto:dean at

Hello All, While looking for solution to solve my Callback DTMF problem, I came across LumenVox Speech Recognition software. Has anyone tried out? Need some feedback before I purchase it... Please help... Cheers, Nitesh

I know this has been asked before and I went over the wiki but I have not been able to come to a clear answer. 1) If I have SIP Provider ----> Asterisk -----> ATA and vice versa (ATA -----> Asterisk ----> SIP Provider) from what I understand if NO NAT is being used then asterisk just starts and stops the session however the RTP media stream will be passed directly from the SIP