similar to: beginners question about extension context

Hey, how can I implement a dial plan like the following: incoming call: 1. ring phones 1,2,3 monday to friday between 9:00 and 20:00; if no answer after 15 sec also ring phones 4 and 5 2. ring phone 1 monday to friday between 0:00-9:00 and 20:00-24:00; if no answer after 20 sec also ring phones 2 and 3 3. ring phone 1 saturday and sunday all day I do not need a in detail answer for each of the

Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI> exit No such command 'exit' (type 'help' for help) *CLI> quit No such command 'quit' (type 'help' for help) *CLI> Any other ideas? I started asterisk with -cvvvvg option. Same problem if use asterisk -r to connect. Can not exit. Any

Hi, all I am doing "prrof of concept" system. I will have two IP phones connected to Asterisk box. Box itself will have 1 PSTN conenction and one analog phone conenction. A basic minimal configuration. At the moment I am planning to use an old PII-350 with 128M of RAM I have lying around. I can not test anything yet, as I am waiting for phones to arrive, so question is will that be

Hi, all I am building a new server. Have installed FC 6 and put in TDM400 card. Checked out latest asteriusk code, run make install in zaptel directory. So far all is fine. Now I am trying to install the drivers. # modprobe zaptel FATAL: Module zaptel not found. Fair enough, no zaptel driver is found on the system. Is there are any known problems with FC6? I did not have much trouble running

>From this list I found that I could use SetGroup and CheckGroup to do what I wanted. But I'm not quite sure how I do it. The case is that I have 3 user groups, and one main group. The main group is for all the incoming calls from external phones. The main group should be allowed to have 3 calls at the time. The 3 user groups are internal groups that I want to limit by ony having one

Hi, all I am in Australia and I have to setup Asterisk in few offices. There will be IP phones in each office and I must be able to call between offices. I need actual handsets. I need "standard" handsets to be used by people. Those must support features like CID, call forward, etc. --- your normal office feature set. Also I need some sort of more complex handset to be used by

Hi, all I am still battling to connect * and voipbuster. What protocol does it use? Ethereal capture shows UDP traffic, but no SIP or IAX traffic when using their client. VoipBuster client connects to connectionserver.voipbuster.com on port 11112 for authentication. Call itself is placed on different server. I have tried to connect using SIP and IAX and it seems that no authentication is

Hi, all I want voicemails to be delivered to recepients by e-mail. I have set up voicenail, and I can see asterisk is using sendmail to send messages out. Using Ethereal, I can see that messages are leaving my network, but receipeint mail server never replies back. As a result, mail delivery is timed out. I got a book on sendmail and it looks quite complex. It will take quite a bit of time

hello i am using asterisk-1.0.9. i have a NAT problem. without NAT registration is ok. and if user is bhind NAT it is registring on asterisk. but SJPhone is showing "not registered". i think asterisk is properly sending request to UA. any comments............this sip.conf setting was working previously -- Registered SIP '5000' at 0.0.0.0 port 5060 expires 120 -- Saved

> Still trying to get NAT working. Try adding a canreinvite=no. Nabeel

Hi, all I am trying to figure out how to forward incoming PSTN call on SPA3000 to VoIP extension(s). Basically, I have converted my home to VoIP. I have normal phone (connected to SPA3000) and couple of IP phones. All call coming from VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I need to do same thing for incoming PSTN calls. I have enabled gateway function in SPA3000 and

Hi, I am very new to Asterisk. I wanted to know how to retrive the Voicemails. I could see some voicemails assosiated with some extensions. Any ideas?? --------------------------------- How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a number of Polycom phones to setup with my * server. For my initial few phones I hand wrote configs. Does anyone here who uses Polycom phones have some form of management utility for automating their setup? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc.

G'Day All, Can anyone give me some direction in setting up the TFTP server on my RadHat ES3 box? I did quite a bit of reading, but I think I am more unsure now than before. I found the information nebulous. TFTP is already installed. I am trying to determine where the root directory for the tftp services is located so I can copy the CISCO 7960 firmware files onto it. Thanks.... Ferg

Does any one know what happened with voice conduits? I have been trying to reach them for nearly three weeks now. Their voice mail boxes are full and writing email to them does not get any returns. Thoughts or sightings are appreciated. -- R.J.

Hi, all I have installed TDM400 card. I can see it is there (lspci). But Asterisk does not find it. phonebox2*CLI> zap show status No Zaptel interface found. I assume that driver is not loaded, but I am sure I have installed it (I compiled zaptel and then re-build asterisk and did make install for both zaptel and asterisk). When asterisk is started I get: Jan 2 06:28:08 WARNING[3473]:

Hi, all I am setting up Asterisk for the first time and have some problems. Setup is very simple -- Astersik box and two Polycom SP300 phones. I will add bells and whistles as I go, at the moment things are very simple. No TFTP servers, so phones run with their default configuration. I set up IP addresses, netmask and gateway IPs manually on the phones. Now, I have read of problems with

Location = US asterisk/zaptel from CVS. Updated last week some time. Currently rebuilding with todays checkout. I have 2 fxo channels hooked up to outside standard Bell South phone lines. If I configure as so [channels] context=pstn group = 1 signalling = fxs_ks callprogress = yes channel => 4,3 Then any call routed from asterisk to the outside line will ring, and can be picked up, but *

Hi, all I am trying to set up voicemail. I've done it to the point where I can leave messages. How do I retrieve them? Actually I have few questions: 1. I want voice mail to be available at certain extension, say 100. How do I set it up so all users can call this number and get to their respective mailboxes. 2. How do I let users to create their own voicemail passwords from the phone? 3.

Hey guys, I'm looking for a programming or Development Team/Company to do some custom coding for Asterisk. What we need is not exactly simple. In fact, I'm not sure the extent of the coding as far as technical terms go at all. Currently we have a "call center" with 4 phones. There will be a total of 8 people using the phones. Obviously, no more than 4 people will use