similar to: How do you transfer a call to a busy extension ?

I know this isn't an asterisk thing, but since the recommendation to get one came from here I figure lots of people out there have one. I read the docs, and it says that in order to do a blind transfer I should hit "flash", then dial "*__" then the number. Now, how on a normal phone do I dial "asterisk underscore underscore"? Can someone tell me how doing a

I have a client that has an old Merlin system. They would like to move to an Asterisk based system, however, with their existing system each phone is capable of displaying who is on the phone within there office. This is done by lighting a red light for each line(extension) that is in use. Has anyone been able to neatly create this feature? Perhaps an XML application can be written for the Cisco

Finally got everything up and run with the help of Manny Wise last night. So I am setting up my digital assistant and getting down to the task I need this box to perform the most. I need to have a custom app that I can call that will take me pressing 2 at the menu and have it transfer the call to a offsite phone number utilizing my Zap Trunk. I'm sure someone has done this already. Anyone want

Ok, this is probably a stupid question, but I can't seem to find anywhere where it tells how to put someone on hold on a zap channel. Flash gives me a dialtone and # tells me to enter a new extension, how can i just put the caller on hold. Pressing # then hanging up drops the call. Is there a simple way of doing this without transfering the user to a parking lot? Thanks, Jon.

Hey Guys Im trying to forward a call with asterisk to a regular phone. Something like " I get a call on my regular phone, and he's trying to reach some buddy of mine.. then I tell him "wait a sec" and push "Flash" and get a other dialtone.. then I dial that other number then hangup the phone, so the one that called will be connected to where I dialed it to"...

Does anyone have any experience with iax.cc/sixtel? Are they a legitimate company? From their website it looks like you can get a private incoming 800 number for 30 cents/month plus 2 cents/minute. Somehow that pricing seems a little cheap for a DID number. I assume there has to be some minimum usage or something. Any info as far as actual costs and/or voice quality would be appreciated.

Is it possible to get an internal dial tone when I call to my asterisk and enter password? I would like to call my line enter extension - password - and get internal dial tone. once I'm in I would like to dial based on what context permits, mostly long distance calls VOIP. I can not preset the extension to certain number as I don't know what number I will be dialing. -- #Joseph

I'm looking into solutions for providing a live stream of an event in Belgium [1] - for example, as follows: * Event --> mobile phone --> software answering machine --> Internet server * Event --> mobile phone --> VOIP --> Internet server The live stream should be available in a format so that people can listen to it using XMMS or similar software. Comments? Would

I would like the ability for a sipura sip device to instantly connect to an asterisk server as soon as the sipura sip device goes offhook and before any digits are pressed. This way asterisk can provide the dialtone and the dialplan. This also allows me to play a greeting to the phone before giving them a dialtone. Is there any way to do this, like possibly having the sipura device dial a

i have got a music file with extension mp3 and it is not workign with background() is there any way to convert the mp3 to gsm or any other codec? Kindest Muhammad Muzzamil Luqman -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050222/2ba5a4f0/attachment.htm

I have added the following to a macro that is used for all extensions so a user can access voicemailmain by pressing * during the voicemail prompt ; check voicemail exten => a,1,voicemailmain(${macro_exten}) exten => a,2,hangup The behavior is a little weird, the * key is not recognized during the portion of the greeting where the extension number is being played back, after it is

I am using the following command and the excluded directory /usr/share/doc/ still appears with all it's contents in the backup directory when it's contents should be deleted. Is this a bug or am I doing something wrong? rsync --delete --delete-excluded -a -R --exclude="/proc/*" --exclude="/dev/*" --exclude="/usr/share/doc/*" --exclude="/backup*"

I have Asterisk setup in a SOHO environment. I have 2 X100P cards at Zap/1 and Zap/2. I have 1 TDM400P card with Zap/3 - Zap/5. I have subscribed to callwaiting, callerid and calleridcallwaiting from Qwest on the 2 PSTN lines - Zap/1 and Zap/2. My problem is when I'm in an active call to the outside thru Zap/1 or Zap/2, I can't pickup the incoming callwaiting call. I can see the

I have a problem. Incoming calls work without problem but I cant call out. Using AAH.Gets a busy tone Anyone who can see a mistake in Outgoing settings context=from-pstn host=ipkund1.rixtelecom.se insecure=very nat=yes secret=xxxxxxxxxxx type=peer username=0406082250 Regards Anders Svensson -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I'm trying to figure out if there is some way I can log agents in and out by just having them call an extension. Ideally I'd like to have it set up where each agent just dials an extension to log in and the same one or possibly another one to dial out. I got the logging in portion to work by not using a password in agents.conf , but the manual log out has me stumped. Any input

Am a little confused on how I can design a dial plan like this: extension 700 should prompt for a PIN and password lookup mysql to see if this is correct if correct, log the time / date in mysql announce "logged" and hangup any ideas? Thanks a bunch --------------------------------- Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! -------------- next part

I read the feature list of asterisk and I cannot see if it is possible to conference a call between extensions. Is it a supported feature of asterisk or is it done in the UA (ATA186 in my case) Here is what I try to do. phone-a -dial-> phone-b tap the cradle (flash on phone-a) phone-a -dial-> phone-c tap the cradle (flash on phone-a) Now I like all 3 phones in a conference call.

The subject says it all. A couple of my sons have very annoying friends that tend to call ALOT. I usually don't like to answer the phone but these kids keep calling back with in 2 minutes of calling. I'm sure someone else has this problem and maybe using * to do a callerID match and block? Even add logic that if they called so many times in an hour? Or in my case, make it a

The wiki says this about the MusicOnHold command: "Plays hold music specified by class. If omitted, the default music source for the channel will be used." http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold How do I set the default music on hold class for the SIP channel ? I tried adding musiconhold=test to my sip.conf. musiconhold.conf looks like this:

The persistentmembers=yes is suppose to keep agents in a queue over a restart. It might do this, but it doesn't do much good as even if they all remain in the queue, they are all logged out on a restart. Is there any way to keep the agents that are logged in, logged in across a restart? Thanks, Jon.