similar to: What do I need to build up DID services?

I have added in my zone file; _sip._udp.elmit.com. IN SRV 20 0 5060 vpbx.elmit.com. As I understand it should mean that any sip connection to <anyname>@elmit.com should go to the udp port 5060 at the host vpb.elmit.com. In Asterisk's extensions.conf I have in the context [default] exten => ronald,1,Dial(${PHONE_615},60,tr) exten => ronald,2,Voicemail,u615@office exten =>

Hi, Inspired by a recent rant about one particular provider, I am getting very curious about something I've never mastered. I'd like someone to explain this here or at least post a link or two that can educate me and probably countless others who have no knowledge in this area. I'm sure there are several of you reading this that know all about the subject. What are the various

Hello, this is Jesse with Webformix. We are managing an Asterisk installation for residential VOIP service, and we are having a problem where all inbound calls to DIDs which are assigned to us by our wholesaler but not yet assigned to a downstream customer get caught in a routing loop. ** For example: we get assigned DID block 1230-1239 and only 1230-1233 are given to our customers, then our

before I accuse somebody to "overbill" I would like you to calculate with me: Rate: 0.0189 for calling Taiwan via NuFone Duration: 930 seconds Lets vote for the answers: 0.7269 or 0.2929 ??? bye Ronald Wiplinger

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

I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger

I tried to use message waiting indicator, by "Subscribe for MWI" in the web menu of the phone. However, it does not light up / flash, even if a voice mail is waiting. Where is the switch to turn it to? bye Ronald

I try to setup SJphone on my PDA, but it does not register with Asterisk. I have setup a sip account on asterisk, ... Can anybody give me a hint? bye Ronald

I am looking for a way of not to install a softphone, preferable as a link on a web site to a webphone on MY SITE !!! Has anybody an idea for that? AJAX? bye Ronald Wiplinger

Skype is very succesfsfull and get more and more users, ... we can ignore them, accept them or do something,... My suggestion is that we try to do something, ... If we would peer to each other, than we get soon also a great amount of users together, and than our service becomes more valuable, ... Let's discuss advantages and disadvantages! bye Ronald -- Ronald Wiplinger (CEO of

I tried to add the databases from the localhost to the database server and changed the every /etc/asterisk/*.conf from host=localhost to host=192.168.10.10 (my dababase server) When I restart asterisk, I do not get any errors, but after a phone call I see: Jun 7 18:11:56 ERROR[7877]: cdr_addon_mysql.c:400 my_load_module: Failed to connect to mysql database cdr on 192.168.10.10 Or if I try

I bought three plus two Grandstream BudgeTone 101 phones. The shipping cost more than the phone itself from Pulver store. The first shipping had one phone defect. Nothing on the display. (Can happen!) The second shipment had one phone with a defect display, but it still worked. The second phone's handset was defect too (microphone did not work). Changing the handset from this one to the

I went on a service call yesterday to find an asterisk system with a T100P card on a Qwest PRI and a TDM40B card connected to fax machines. The TDM40B LEDs were not lit, and the system did not respond to keyboard input. However, calls were being processed for the PRI and 7960 phones. I replaced the TDM40B card with a new one, and the system now seems to be ok. But I'm wondering, why

Hello Everyone, Unlike going through DIDx, DIDLogic etc.., we have an option of getting DIDs directly from local telco Bell Canada. Currently our SIP Trunk provider assigned a DID to us, and as you know, they just redirect requests it to our PBX. However, when dealing directly with a telco, what equipment will we need? Basically giving us the same capability as a DID provider. If someone can

I would like to setup a test number, that speaks back my phone number. How can I set this up? bye Ronald

I would like to know which extension number is engaged in a call. show channels shows me: *CLI> show channels Channel Location State Application(Data) SIP/asterisk.elmit.com-0 690@default:2 Up Echo() SIP/8807-066 690@newcontext Up Echo() 2 active channels 2 active calls but it is not

OK, got a question on 911. Looking into setting up a couple asterisk servers at a country club, with VOIP phones in each of 100 short-term residential rental units. Approx 100 extensions, approx 24 outside lines. Since everything is geographically at one location, reaching 911 correctly shouldn't present a problem. However, the club wishes to ensure that 911 authorities are able to identify

Is there a way to test if the motherboard is ready for a Digium card (PCI 2.2) ? I would like to know from a remote computer, where I have (root) access, if this computer is ready for a TDM22B. bye Ronald

Hi all, a noob here, I am trying to get outbound calls through asterisk working with Broadvoice. I have consulted the following two online tutorials: http://www.broadvoice.com/support_install_asterisk.html http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice in an effort to make outbound calls. My current settings are as follows: sip.conf register =>

Is features.conf included in the cvs as of 8-1-04? I have updated, but am not seeing it?