similar to: Voicemail Emails

Hi all, I have a problem with voicemail. My boss has asked me to send via email, the message that a user leaves on the voicemail. This is very easy. :) After, he asked me to check before sending the email, if the receiver's mailbox is full. If the mailbox is full, Asterisk should call the receveir intern (example 2001) and using a Playback tell him that his mailbox is full. How can I do?

I am currently using AsteriskNow v2. What I am trying to accomplish is having all calls from an area code go directly to the person responsible for that area. While searching for a solution for this I did come across a post that had a few examples. So Josh at extension 1902 would receive all calls from the 808 area code. exten => s,1,GotoIf($${CALLERIDNUM:0:3}" = "808?1902|1)

I am trying to record calls on demand both inbound and outbound calls. I can record outbound calls just fine but not inbound calls or calls from an internally between extensions. I am using the latest asterisk 1.8.x certified version. On an outbound call I see: == Using SIP RTP CoS mark 5 -- Called SIP/ BVTrunk /7190000000 -- SIP/BVTrunk-00000163 is making progress passing it to

Does anyone know what version of Asterisk Switchvox uses, and if it is modified in any way? FWIW, I am dealing with a provider that claims compatibility with Switchvox but not Asterisk for their SIP trunking service.

I just inherited a client that is using a Switchvox system. I normally install a CentOS based system with freePBX and some custom endpoint management stuff for Polycom phones. This Switchvox is making me feel a bit stifled. I am having nightmares of another recent encounter with Trixbox Pro. Can I really not ssh into this box? If I could is there anything useful that I might change

Hello, I need to have an extension on a SwitchVox server dial out to one on an Asterisk (FreePBX actually) box which will host a voice directory. The Asterisk server will then need to dial one of the SwitchVox extensions if it gets a voice match. Anyone has done that, and could let me know how? So far it looks like IAX peering (what SW calls "SwitchVox peering") could work? Thanks in

I have a customer that wants to get switchvox, since I have never used it, I would like to hear some feedback from active users of switchvox. In specific: 1. Does it use realtime or conf files 2. Is it possible to change it manually? 3. Is SSH access to login to console/shell available? 4. Are you or your customers happy with the user interface? TIA

Does anybody know of a Similar list for SwitchVoX? And would like to post to proper list if one is available. I had posted on digium forum, but have not received any responses yet. http://forums.digium.com/viewtopic.php?f=38 <http://forums.digium.com/viewtopic.php?f=38&t=77031&sid=4adb81c464701e0039d e21a300aa273f> &t=77031&sid=4adb81c464701e0039de21a300aa273f

Hello metafor users, I'm using metafor to perform a single-effect summary estimate of the raw proportion of patients experiencing a post-operative complication, and I'm interested in seeing if this proportion differs between the three most commonly used surgical techniques. The software is working as expected, but I would like to double check on the interpretation of my mixed-effect model

As you may have heard, Digium announced this morning that it's acquired Switchvox, a well known provider of Asterisk-based phone systems. Since several people have already asked me about the deal, I figured I'd let you all know my feelings on the matter. First of all, let me say that I personally think this is a great thing for all the parties involved. Obviously this gives Digium a

Hi, I want to start managing my asterisk boxes with a centralized graphical based interface so I can (due to customers request) give control to customers to add/change extensions to their current PBX intallations such as (not complete list) Add/del/mod extensions sound recordings (ivr or voice attendants) email to fax/ fax to email voicemail to email SIP and ZAP, no IAX needed configure calls

I'm impressed with the call log reporting that Switchvox offers. There's a screenshot here: http://www.switchvox.com/sv?cmd=screenshots&pic=13 Can normal Asterisk do this? I installed AsteriskNow on an old computer, but I couldn't find anything like that in the GUI. I found the CSV file in /var/log/asterisk/cdr-csv, but that's not much help unless there's some other

What are the HA options for Switchvox systems? Is it possible to set up redundant systems with DRBD? I know on the digium website they talk about "Optional cold spare failover" What does this mean? Is this an active spare ready for some sort of automated failover? Thanks for you help, Bob

Hey all, First-time poster and been following the list for some time. Great postings, all... Currently we're implementing the voipconnection vs-1 for a customer. It's a decent device, and the included admin interface leaves a lot to be desired... The idea of the storing settings compact flash is nice, although sometimes they choose not to stick. All in all, we're thinking about

If you are having trouble reading this email, read the online version<http://now.eloqua.com/es.asp?s=491&e=78675&elq=55426a8b6c714f5bb6f2bf4b5d37bf55> . <http://app.en25.com/e/er.aspx?s=491&lid=215&elq=55426a8b6c714f5bb6f2bf4b5d37bf55> Dear Lito, *The information in this email is given to you in advance to make you aware of an impending product release

Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their software/GUI to run on my hardware. Does Asterisk BE come with a GUI management console for managing phones, queues, VM and the like? -Matt -- Matthew S. Crocker Vice President Crocker

Hey all, I thought I'd run this by the list before I attempt to go through official support channels (just because I never get anywhere). I've done 3 small business installations with asterisk and polycom sound point IP 500s. The first 2 installs run CVS-Head, the most recent install is just running stable. Now I have 28 polycom soundpoint IP 500's on their system, all using SIP no

Hello to all, I have a problem with an asterisk qsig. I have three machines: Nortel CS1000 --- Card Sangoma PRI ---> Asterisk QSIG ---SIP Trunk---> Asterisk I use Snom phones on Asterisk. If I call from Asterisk to Nortel, Nortel reminds me of the name of the person i'm calling and I visualize on the display of Snom phone, but if I call from Nortel to Asterisk, the QSIG does not send

I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. bye Ronald

Hey guys, Cal someone help me. I'm register voiceplus DID i try to config fllow example but not work. When i test call to number and debug iax2 in my asterisk not found packet. My iax.conf -------- register => in-xxx:yyy@gw5.voicepulse.com [voicepulse] context = voicepulse-incoming secret=yyy auth=md5 type=friend host=gw5.voicepulse.com ------ extention.conf ---- [voicepulse-incoming]