similar to: Newbie help/pointers required - configure xlite with asterisk

Unfortunately I seem to have another problem! I am using sipgate for the incoming line - and it appears that you cannot get DTMF to work in that configuration. Unless anyone knows anything different of course!! > > I just want one of my incoming numbers to go to an IVR service that > > will allow me to select what I want. > > > > For example > > > >

Where can I get some asterisk books.. or tutorials..? I?ve been searching in google.. but I find just some tutorials explaining how to fast set up an asterisk server. I want to learn how to use it and how to make my own configurations. So, the thing is that I want to know what is the best book or tutorial that you know? recomendations? Thanks to everyone... Danko Miocevic

Hi, all: I am a newbie to the asterisk and its architecture. :( After reading some help in the tarball of Asterisk, I am still in the mess. So I want to know where I can find a detailed explanation of the Asterisk which including the Architecture, Install, Configure, usage example document. Maybe what I want is too much, after all it is a open project, not commercial product. If I want to get

Hi All, I've installed digium TDM02B. The PSTN line connected to this card. At the IP network side, I have SIP phones registered sucessfully to asterisk server. How do I configure asterisk, so once there is incoming call to the TDM channel from PSTN, the caller will hear another dial tone from asterisk then to key in the extension of intended destination (SIP phones number) that already

Hello All, Please forgive the lack of understanding as of yet but I have been trying to follow the mailing list messages over the last few days and would like to know if someone could wither point me into the right direction or possibly give me a brief overview of the complete process. Basically, I see that the Asterisk PBX systems can run on linux and seems to offer the engine base that is

Please help me I am brand new to this. I have been following this tutorial from OnLamp http://www.onlamp.com/pub/a/onlamp/2005/01/20/rails.html which tells you how to install ruby, rails and mySQL and get going. Doing this tutorial convinced me RoR was the way to go for me. So I got hold of this e-book: Agile Web Development with Rails - The Pragmatic Programmers The book tells me to install

Hi, I'd like to make small 20 users setup with BTs. I'd like each of them to have its own context (for recording prompts, conference, ...). For them to have same extensions I should put them in separate contexts and let BT call them offhook. But these contexts are pretty similar (for instance dial to conference on 5 goes to different conf. number for each user, ...) How could I describe

Hi Friends, I want to setup multiple SIP accounts. How can I do this? I have installed Asterisk, created Asterisk SIP extensions and registered in www.sipgate.co.uk. Now, what I have to do? 1) Am I need to install SER or OpenSER in my server along with Asterisk? 2) If yes, can you please recommond SER or OpenSER? 3) I searched in Internet. But, I didn't find good tutorial for this. Can you

I'm using sipgate.de as my sip provider. When I'm using xlite as client on sipgate.de, everything works fine: I call number, hear ringing (real progress tone form called party, not one generated in xlite) and then talking with called person. But, when I'm using Asterisk as sip client on sipgate.de, I don't hear progress tones: I hear only one (locally generated) ring tone, and

Hi, I have installed Asterisk, Zaptel, Libpri, Addons, Sounds in my Linux system. I got registered with sipgate.co.uk and got the UK phone number i.e., 0207100xxxx. I configured my Asterisk server with 0207100xxxx. When I made a call to this number from outside phone, my XLite extension is not ringing. Its directly going to Voicemail or telling that "person is unavailable". When I

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I need to find some basic configuration files. Is there a place I can check out how to set up an office using sip telephone and Digium FXO and FXS ports? Don Moskaluk don@moskaluk.com www.moskaluk.com 416 737-8230 Cell 416 614-8230 Home --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.732 / Virus Database: 486 - Release Date:

hi all, I don't know how to setup asterix to work as PBX. If I want just basic configuration with 2 SIP phones (snom, ata), what all I have to write in the configuration files, or respectively in the configuration of ata and snom ? If there is any good documention available, send me URL too. All (ata, snom) are behind firewall (nat) and astrix is on the public IP, but I can move for

Hi all! I am using ohphone (well, I am trying to) to make calls. I will make an H.323 - SIP Gateway but I don't understand the architecture of all this. What is the exact role of asterisk? It can be used as gateway, that I know, but what else can he do? Is it necessary to have ohphone to make calls or asterisk can also do that? So when the gateway it is going to be implemented how is it

I am trying to get an initial setup up and going which I assume is a very common question here. My basic questions are the following: Can I get Asterisk up and going without voice cards using it with SoftPhones internally as a proof of concept. (just calling extensions and leaving voice mail) Is there a jump start config that would accomplish this? What is the recommended SoftPhone that is

Hi, We have set up our Asterisk server, our extension.conf and sip.conf according to http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=4 It's quite basic, and extension.conf is set up properly. The difficulty we are now encountering is in sip.conf, in trying to get any softphone to register at our own Asterisk server. We have searched the mailing list, and find bits and

Hi, I've got an Asterisk box with grandstream and xlite clients on it. No here's the thing: - I grey out all the codecs on the Xlite except for GSM - I call the Grandstream from the Xlite, the Xlite uses the GSM codec and the Grandstream uses ulaw, with Asterisk doing the conversion, everything fine - I call the Xlite from the Grandstrea, the Xlite ends up using the ulaw codec as

Hi all. Just as a quote note, can I thank everyone on this list. I find my self finding pretty much every answer I am looking for on here. And a big thanks to all thoughs helping us out. Mass Respect :) Ok, I'm using a SIP provider (SipGate UK) to do my international dialing etc, working great from extension 8 on phones. However some more friends/contacts have started using SipGate also, and

Hi, I am sorry to be asking this but the wiki is down and has been for a couple of days and I need to get this working before Monday to get my live system setup. Trying to get the Snom 190's and soon to arrive 3com 3102's to use the function keys and for the life of me I can't work it out from the conversations on the archive what I am going exactly wrong here? The snom 190 with