similar to: dialing several phone numbers in one call session.

Hello! Asterisk box receiving calls. Is there some way to get information about current calls from external or AGI application? I'm interested in: - duration, how long calls already in the system (billing and actual time); - source/destination phone numbers; - etc. In other words can I receive information which we are usually getting in CDRs during the time when the call is still active?

Thanks a lot for all the suggestions! Unfortunately, it still gives problems. Most common error message is "ast_realaudio_callback Failed to write frame" after "paying the beep". Then it says "User disconnected". Also, it doesn't react to any extension entered and doesn't do any forwarding (as it should in "exten =>

I have a server running Asterisk 1.4.14, Zaptel 1.4.6, Libpri-1.4.2 on a CentOS 5 server. The server has a single TE110 card connected to a provider called Alestra in Monterrey, Mexico. Since we installed everything we have been having problems dialing certain numbers, those numbers always fail when dialed from Asterisk but if you dial from your cell phone they always go through. I once has a

I have a number of local users who can dial out on a pri channel using the fantastic new overlap dialing feature. I would like to add a speed dialing feature, such as 1. User picks up and dials out (dial startet with option 'H') 2. User hangs up call with '*' 3. Dialed number is stored in a variable 4. User dials a two-digit extension followed by the # sign to save the stored

Hi, Does anyone know how to dial toll-free (800) numbers through FWD or Siphone? Using the configuration below, I can dial out to SIPphone.com users by simply dialing their number (1747XXXXXXX) and can dial out to FWD users by dialing 1383<FWD#> However, when I dial 18005551212 through SIPphone, or through FWD (depending upon which line is selected in "; 800 Toll Free Numbers"

On Monday 26 Jan 2015, Antonio G?mez Soto wrote: > Hi, > > does anyone have a recommendation for a SIP phone, which > allows dialing from a phonebook, and hiding the dialed number > from the end users? Also from the call history of course. > > It seems Mitel can do this, and I have a use case where this is > a requirement. If I have this right, you want to make sure the

Hello, I'm trying to figure out how to get Asterisk to dial an extension when a call comes from the outside and contains the extension already. (Somebody wants to call a user of Asterisk with extension "111" from the outside) For example: I've hooked Asterisk to sipgate.de and received a landline phone number (say 0781205237). Now if you dial 0781205237 and and an extension

Can anyone tell me how me to get asterisk to dial out a phone number when a bluetooth device is not detected? I've tried putting the following under the clients section in /etc/asterisk/btp.conf: client =>user,00:12:34:56:78:90,Zap/4/1234567890 and in extensions.conf: exten => 222,1,Playback(pls-hold-while-try) exten => 222,2,Dial(BTP/user,60,m) exten => 222,3,Hangup but

I sent the following message a few days ago, but never received a reply, so I thought I'd ask again.. Can anyone tell me how me to get asterisk to dial out a phone number using BTP when a bluetooth device is not detected? I can get BTP to dial to a SIP phone, but I can't get it to dial through a POTS phone line using the Zap interface.. I've tried putting the following under the

Hello, I need to detect availability of SIP phone before dialing. I need to know if phone is BUSY, CHANUNAVAIL before dialing. If phone is "free", then I will dial it. I need for automatic callback (.call files), but I need to know if it is available both SIP phones before calling.

H, Just a quick question that has been bugging me for a while..... Most of my address book phone numbers are stored in the format: +<country code><area code minus the 0><local number> i.e. +XXXXXXXXXXXX In my asterisk (Trixbox) server I would like to be able to dial numbers from my address book using HUD or the SIP client on my 3G phone using numbers in this format. On

hi there. i've got a question about outbound dialing. here's my scenario: 1. i build a list of phone numbers from a database 2. when a call comes in, i begin dialing from the list 3. when an outbound call is answered, i connect the caller to that line. so far, i'm able to do this with an agi script to dynamically build a dialplan. i make repeated use of this perl call:

Hi, I have not been successful in working out how to dial a # within a phone number. EG: exten => _12345,1,Dial(Zap/1/0868563823#,5,t) or exten => _08XXXXXXXX,1,Dial(Zap/1/${EXTEN}#) I'm trying to append a # character so that I can use a cellsocket (mobile phone to pots adapter) connected to an x100p. I think that asterisk is simply ignoring the # character. The docs on

Hi everybody We are having periodically troubles with the outbounds calls, seem like the PBX cannot end to dial the entire string of numbers This is output from the PBX., few minutes after all work fine again ... :(, and few minutes after the same problem appears. Thanks in advanced ... Regards This is the PBX aout and my zaptel, zapata confs. Runing asterisk 1.0.9 libmfcr2-0.0.2

I've discovered that when dialing out using API's Originate action, a no answer is considered a failed attempt, while a busy is considered a successful attempt. The problem I'm having is that when I dial an invalid number, say a disconnected number that gives a fast busy, my CDRs are identical to those generated by a no answer attempt. Is there a way to distinguish between a no

I would like to prevent * from dialing 900 and 976 numbers. I setup the following settings in extensions.conf. But this does not seem to work! I don't know what I am doing wrong please help! exten => 1900XXXXXXX,1,Congestion exten => XXX976XXXX,1,Congestion exten => XXX976XXXX,1,Congestion exten => 1XXX976XXXX,1,Congestion exten => 91900XXXXXXX,1,Congestion exten =>

The following strange conditions is happening while I try to dial a SIP user from another SIp user. SIP to Zap dialing is fine, as all 4 users can call PSTN. I'm using Asterisk SVN-branch-1.2-r51359M Example: extension 3210 calls extension 3213. They are all registered properly: chrom01*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 3213/3213

Hallo! I have problem with incoming calls on 2 phone numbers registered on same SIP provider account. I've tried averything and nothing seems to work. No matter what I do asterisk system refuses differ betwen them and both got connected to the same extensions. I've tride with: registration => num1:pass@provider/ext1 registration => num2:pass:num1@provider/extt in sip.conf and

Hello all, i recently finished setting up my asterisk with sipura 3102 using PSTN. this is my dial plan relevant to wht i want: exten =>_01,1,Dial(SIP/$(EXTEN)@200) right now as u see i made my dial plan on a 2 stage dialing mode. tht means i dial 01, i get the pstn dial tone, and then i call whichever number i want through it. i want to have the option for my call to directly go through

Hello, I'm preparing a setup before installing it within the next few days. In this setup, I'm using a SPA112 as an ATA for an analog phone. The target phone is a Gigaset A400 DECT handset. In my lab, I've got another A400 handset and an old Matracom 46 handset. When I connect my Matracom 46 handset to my SPA112, I can send and receive calls. When I connect my A400 handset to the