similar to: Telephony Migration Hardware

Hi, I am looking for hardware sip phone with very good sound quality. Can anyone recommend ? I use to have Grandstream Budge-Tone 100 but I feel that the sound is not very satisfactory and volume too soft Regards ASLAY

Hi, I am running Asterisk v1.2.7.1 with a Digium TE110P. My dialplan is very simple, when a call comes in on my analogue BT PSTN line, it rings the other ZAP interface (my house phone). Slightly pointless (having a 1x1 switch) I know, but I am planning on doing more with internal SIP extensions, and outgoing SIP services etc.... At the moment it's just simple though whilst I try to fix this

Hi I have previously had good success on smaller installations with TDM400P cards. I now have a UK customer looking for 8-10+ lines and it seems like a PRI would be most economical + reliable? Has anyone here used PRI interfaces in the UK and can confirm that it works well (using Trixbox for preference?). I have had some niggling issues with the TDM400P cards which puts me off adding lots

During the course of a conversation with an member of the IT group who handles the E911 center for our county, I learned that all of the county's E911 is voip based. This got me to wondering why we could not just configure up a SIP or some such trunk directly to the E911 center to handle our emergency traffic. The county seems interested in exploring the possibility. So I'm wondering if

Hi All, I have a GSM box, which needs to connect to a analogue phone line. I've plugged the GSM box to a Grandstream ATA (386). This ATA has extension number 600. Now what I want to accomplish is the following: - If a mobile-number is chosen by a user, asterisk needs to call the ATA (600), wait for a few seconds, and then send the mobile-phonenumber. Or, if it's possible, define the

Hi List; Is there an Digium cards support GSM SIM cards so we can fix an SIM card to be used for calls within mobiles as it is less rate? Or I have to use an FXS to SIM adaptor? If yes, then anyone advise a models and prices? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo!

Hi all, As we know we can configure in astersik like before 5:00pm calls go to reception and after 5:00 pm calls go to some mobile no. One of my client requested that he wants to manually shift the dial plan like above as he has flexiable timing sometime he finishes at 3:00pm some time 8pm. I can not give him freepbx access. Any idea or solution. Regards Farooq --

Evening all. I'm looking at building an Asterisk system for one of the projects of the Charity where I'm the SysAdmin. The project has two analogue phone lines - BT Featureline Compact, we're in the UK - that I'd like Asterisk to handle. My current quandary is which FXO interface to use. I've been looking at the Digium TDM400P and Sangoma A200, which are similar prices for

Hi, I'm trying to configure 2 parallel sip trunks between 2 boxes. However I seem to have the problem that when making a call from Box 2 to Box 1, it sometimes says authentication failed because it is using the username of the other trunk. Here's my configuration: Box 1: [dp-dp2] type=peer username=dp-dp2 secret=mysecret qualify=yes host=box.2.ip.address context=from-internal [e911-dp2]

Does anyone have any experience with any SIP or IAX providers that support E911? I'd love to convert entirely to Asterisk at my house, but the lack of emergency dialing has been a major hold-up for me. Thanks in advance for any suggestions! -- Kyle Sexton

Can Asterisks properly handle outbound Enhanced 911? Blake

Hi i'm planning to migrate a callcenter to asterisk and VOIP, the call center can have up to 25 cuncurrents agents logged in. I'll have some simplty IVR business logic and the some queues. Can a normal server with 1 GB ram 100 GB HDD Pentium 4 3.6 Ghz CPU Ethernet 10/100/1000 Support this? Would you suggest me a particular products? The server and the agents will be in the same LAN,

How exactly does Asterisk provide E911 service??

Our asterisk system gains access to the PSTN through a voip provider. We have no digium or other telephony hardware in our system. Do the zttest results still matter to us? Our results were as follows: --- Results after 1007 passes --- Best: 100.000000 -- Worst: 99.780273 -- Average: 99.975763

hello friends friend i had just finished my chapters of asterisk. ill be configuring asterisk in for home for r/d purpose. i am having p4 machine with 1 GB RAM, ill be configuring asterisk on centos 5.3, the only doubt which i am having is which hardware ill have to buy to configure asterisk. i think analog card ? plz clear my doubt. n be with me from beginning till end, of the journey of

One of my providers has a different SIP account for each number. I have all of my users in one outbound context (caller ID passes fine). How do I ensure that the callers get routed down their correct SIP account with my provider for e911 purposes without each having their own context? ----- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part

Hey everyone, I have a more of an opinion question then a technical question. The asterisk server I am setting up is going to host 3 different businesses. Each business is in the same building, and on the same network. My question is regarding calls coming in and going out. We are a small ISP and have a lot of numbers that are forwarded to our phone system. The other companies have about 3

I have asterisk successfully dialing out using a X100P over a normal analogue PSTN line. But when I try to dial out over an analogue line that goes via a hardware PBX the call asterisk does not dial. Is there a configuration change I should make ? I am thinking of something like not wating for dial tone.

I believe the 911 is a serious issue if one does an asterisk installation in an office. How do you test 911? Won't they arrest you or something for dialing 911 for no reason and talking to one of their agents who could have taken a more important call? On the other hand what an emergency comes up (like someone got seriously injured) and on top of that asterisk crashed all of a sudden

I have a client that would like to use asterisk to link their multiples locations together. However, if a person in the remote office dials 911, How can the 911 operator determine WHERE the emergency is?? Since all calss would be going out of the PRI in the main location, the police/fire trucks will show up at our COLO!! I know that there are some that are doing this multi site setup, how did