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Has anyone been able to do the following: 1. Set the phone to automatically record all calls to the USB stick, now you have to press three keys. 2. Put Record on the main screen when a call is active. This would eliminate having to press the 'more' softkey. Thanks, Matt

Hi list, What possibilities are there in asterisk to send an *alphanumeric DTMF*from/to asterisk !? Regards, Sammy -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120228/e62e7890/attachment.htm>

Aloha, Are there any hard phones that support ILBC? I see lots of soft phones but not Cisco, Polycom etc. Aloha, Matt

Aloha, Anyone know how to disable call forward on a Polycom Phone. Calls being accidentilly being forwarded somewhere is the #1 trouble that we have to respond to. The real issue is the 'end call' button becomes 'forward' when the call ends....therefore the user thinks they are pressing 'end call' but the call ends just before they press the button so they end up

Anyone know what happened to netxusa? Seemed like they dropped off the web overnight. -Matt

Most manufacturers charge in excess of $80 to upgrade from a 10/100 switch to a 10/100/1000 switch built into the phone. The cost might have been in the chipset 5 years ago but I can get a 5 port gigabit switch for $30. What are most folks using for people that need gigabit to the desktop and don't want to run another cable? -Matt

Is is possible with a Polycom phone to update the LCD with the callee's name after dialing them? When you dial ext 103 now, it says 'To:103'...would be nice if could have 'To:Dan Marino' This is the case even when you have a contact for ext 103. None of the phones I have ever tested do this, Polycom, Linksys, Cisco, Grandstream, Yealink, etc. -Matt

We have a queue that agents log into through the dial plan. Extension Sip/101 logs in as Agent/101 We have 'ringinuse = no' in the queues.conf file. The issue is that when Ext 101 is on a 'non queue' call (they placed a call, someone called their DID, etc) they still receive queue calls. Is there a way to stop this from happening? -Matt

Have not been able to confirm this, anyone heard this from another source? http://bit.ly/jCj5uj "We expect that users of Skype for Asterisk will be able to continue using their Asterisk systems on the Skype network until at least July 26, 2013. Skype may extend this at their discretion." -Matt

Is this command the best way to access a MySQL database - MYSQL(Connect connid dhhost dbuser dbpass dbname) ? I thought I heard that using ODBC was a bit more stable. Anyone have any experience? Thanks, Matt

In the past we have used Adtran Atlas 550's to break out FXS ports for devices like modems. The great thing about the 550 is that internally it is all TDM so there is absolutely zero latency. We are able to use ATA's for faxes and analog phones but devices that use modems, they fail 99.99% of the time when using an ATA. We tried to migrate to TA908 devices; they have FXS ports built

...tusnet.com.au> wrote: > Digium PSTN cards seem to work. > > > > PaulH OpenVox works well. Best Regards, -- SplatNIX IT Services :: Innovation through collaboration ------------------------------ Message: 21 Date: Wed, 20 May 2009 22:04:21 -1000 From: Matt Darnell <mattdarnell at gmail.com> Subject: [asterisk-users] Polycom Productivity Suite To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <52b8ace90905210104u4239b2e2t3a539b34d75599a3 at mail.gmail.com> Content-Type: text/plain; charset=ISO-885...

I heard mention of inexpensice FXS gateways. I only know of Adtran TA 750's. They are way beyond $500. Anyone have any recomendations? Aloha, Matt

Aloha, Did anyone ever get the formating manual for the XML brwoser on the Polycom IP600? Does anyone have a sample? Aloha, Matt

Has anyone been succesful pushing a VLAN setting to a Polycom phone via DHCP? I can push the boot server via option 66 but that is about it. I have set it for 'fixed' and tried many different option numbers with a couple differnet DHCP servers. SIP firmware 1.3.4 or 1.4.1 doesn't make a difference. Aloha, Matt

>From the Wiki: 'There are aready a few bugs in 1.5.2 but more fixes some good new features' Anyone know where the bugs are being listed? I am working through a few issues: 1. When rebooting, the phone will pause for exactly 180 seconds with the screen reading 'updating initial configuration'. I know it is 180 seconds because when I tail the FTP server log the entries are

Anyone seen anything on the IP670 & the Color Expansion? -Matt

A client of mine asked about a Polycom IP321..anyone else heard about it? -Matt

Is there a way to make a virtual extension busy programmatically? I want to be able to turn lights on and off on a Polycom phone from a script. -Matt

We have a customer that does not care for the default MoH. We have downloaded some royalty free music but it sounds 'fuzzy' when we test it with the system. We down sample it to 16bit, 8KHz, Mono. We have tried with Audacity, CoolEdit Pro, & VLC. Does someone have a file they can send me that we can test with, or has any tips? Much appreciated, Matt -------------- next part