search for: televolve

We have licensed both products and sent a support request on 6/11, with zero reply or any activity on it at all so far. No replies to subsequent ticket updates or e-mails. -- Carlos Alvarez TelEvolve 602-889-3003 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130617/c0e347d9/attachment.htm>

Hello List, Since I'm looking for a new VoIP provider for US origination/termination, I will very appreciate if you can chare your experience with Flowroute, Vitelity and Voip.ms Thanks in advance! Elder D. Arohuanca dCAP 1497 Lima - Peru -------------- next part -------------- An HTML attachment was scrubbed... URL:

Has anyone been able to configure Asterisk to work over 3G? I bought Nokia Cell Phones just for that purpose and they register fine over WiFi and 3G but the quality is just not good enough and sometimes the call just disconnects. I have Allow as: ilbc gsm ulaw alaw -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667)

...c123@>' failed for '68.2.x.x' - No matching peer found Typically of course we'd expect to see: <sip:abc123 at server> We're running the latest available firmware, but it's from 2009. Any ideas on this before we just trash all the older phones? -- Carlos Alvarez TelEvolve 602-889-3003

Hello, For those customers with only analog lines, who ask for fax2email and email2fax, whats the most reliable solution available and tested with Asterisk? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120529/3e28b56e/attachment.htm>

Hi guys, I am trying to make a trunk between two asterisk system SIP Trunk on Asterisk 1.6 but I cannt make it work, can any body help me plz? Thank you

...fter=0) exten => s,n,Page(${PAGINGLIST}) exten => s,n, Hangup The SPA phones simply ring. I have verified that Auto Answer Page is set to yes (the default). We've tried a variety of firmware versions and phone ages, going back to an old 942 and new 504s. Any ideas? -- Carlos Alvarez TelEvolve 602-889-3003 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130321/67d75c30/attachment.htm>

I have a customer who would like to play a series of sound files during a phone call on demand. There would be several played in order during a call. Any simple ideas on doing that without developing a whole web app to do it via AMI? -- Carlos Alvarez TelEvolve 602-889-3003

Client - Not for Profit in the Middle of the Jungle/Rain Forrest Infrastructure - Datacenter is Non Climate Controlled, Prone to Flooding, and has Sketchy Power, LAN - NEW Cabling in main Office building, Hodge Podge of DYI wiring across remaining buildings. Phones - Total of about 50 extensions. Only about 25 - 30 phones will be IP phones, 20-30 more will have to be analog due to the distance.

Hello, I need to setup system of aroud 60 DECT phones with asterisk. So far I found http://www.grandstream.com/index.php/products/ip-voice-telephony/enterprise-ip-phones/dp715_710 However is there some cheap base station(similar to GSM cell) so I can handle all DECT phones centralized and plug it inside asterisk ? Thanks, Peter

Hi I came across couple of pointers on the Internet regarding solutions available for providing hosted PBX service. 1. Multiple PBXs: Using separate hardware to host each PBX. Pretty straightforward, but no hosting company wants to use it. 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way

Hi, I am looking into advising a client on the pro's and cons of using Installing asterisk on a server vs appliance(e.g digium mypbx). the appliance seems cheaper initially.

...r on the phone. The calls would come into one of a few dozen DID numbers, each one for a specific company. The agent needs to know which company the call is for and answer appropriately. I've done a lot of this in dialplans but haven't found a way to do it in a queue. -- Carlos Alvarez TelEvolve 602-889-3003 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121206/f67d5d74/attachment.htm>

...e group is busy. This is good for a user with multiple devices, but not useful for teams where any person could take a call, like a customer service group. Does anyone know if it's possible to have a hint with multiple devices which only shows busy if every device is busy? -- Carlos Alvarez TelEvolve 602-889-3003 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130809/4429f986/attachment.htm>

I'm trying to understand why vendors keep making 100Mbps integrated 1-port switches in their hardware SIP phones. Even the recently-announced D40 and D50 Digium phones are limited to 100Mbps. Only the more expensive models (like the D70) can run at 1000Mbps. However, you can't expect a firm with hundreds of extensions to buy the most expensive model... And gigabit speed is important when

I am running Asterisk 1.8.10.1. I am trying to set up some routing in my dial plans and having some issues (the issue being that I don't quite understand all of the syntax and patterns that can be used: Examples: DID1 = 6140000000 DID2 = 6140000001 CNAME1 = 6149999999 CNAME2 = 6149999998 CNAME3 = 6149999997 context1 context2 context3 I have looked at several examples (patterns) and I

We've deoplyed a number of pure VoIP wireless (wifi & proprietary) phones, but not dect. Is there a simple overview of integrating DECT phones with Asterisk somewhere? I assume the DECT basestation has a multi-account SIP VoIP interface, and the handsets are just plain old dect? Can you push configuration info to individual phones? (Are they individually addressible / configurable

Hello, ChanSpy can not be used on a Channel that is being recorded with MixMonitor. How can I verify if a channel which I want to spy on, is currently not being recorded ?! Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120202/7954fe9e/attachment.htm>

...riting the results of sip show peers to a text file if customers report issues, but it would be nice to have a tool that logs all the time and lets us do some better reporting. For example, graphs of latency in a time range, or a list of unreachable phones within a range, etc. -- Carlos Alvarez TelEvolve 602-889-3003 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130410/80114c73/attachment.htm>

hi, We have a phone number from third party provider which is used for inbound calls. How could I monitor if this *phone number* is reachable? the initial idea doesn't sound elegant: - on my SIP server I set couple seconds of ringing before Answer(). - the monitoring server calls to that phone number for few seconds, checks if it "hears" the ringing and hangs up the call. ** I use