similar to: transfer application

I wish to have my Grandstream GXP-2000 phones make a different distinctive ring for internal calls ( Internal ) or if the incoming call has no caller id 'NOCID'. The Grandstream phones calls allow 3 distinctive rings depending on the caller id. I have one set up and working for 'Internal' calls but unfortunately the same tone will ring if caller id is absent on a call. My

Hello Instead of using PrivacyManager, I'd rather use my own dialplan to prompt the user for a ten-digit number if they called while blocking CID. This code does prompt the user, but 1) hangs up if the user didn't type the ten digits before the timeout 2) if the user did type the right number of digits, it still hangs up instead of Returning and then jumping forth to the "cid"

Hi, why would Asterisk core dump with the following test dialplan extension ? exten => 8100,1,Answer() exten => 8100,n,Set(CALLERID(all)="") exten => 8100,n,PrivacyManager() exten => 8100,n,GotoIf(${[${PRIVACYMGRSTATUS} = FAILED]}?:nocid) exten => 8100,n,NoOp(Number is ${CALLERID(num)}) exten => 8100,n,Hangup() exten => 8100,n(nocid),Playback(vm-goodbye) exten

I've been trying to get an AGI "transfer" to work for several weeks now. It isn't error-ing out, but it isn't working either. I can't use "dial" in this case due to what I'm trying to accomplish. Does an AGI Transfer actually work? -= Info about application 'Transfer' =- [Synopsis] Transfer caller to remote extension [Description]

Hello List! We're having trouble making call deflection on ISDN PRI. We would like to transfer a call to an external extension but keeping the callerid of the caller so it can be presented to the receiver of the transferred call. At the time we're using Zaptel 1.4.5.1, Asterisk 1.4.11 and Digium hardware TE420B. We've ordered the service (CD) from the phone company. The

Hello, I 'm looking for a way to pass the '302 moved temporarily' received from the SIP device back to the SIP provider. Here is the setup: Some SIP phones are connected to an Asterisk System version 1.8. External connection to the public network is also done via SIP to a VoIP provider. Phone A has a CFW all calls to a phone number in public network (Mobile Phone) incoming call to

Trying to set up Junction Networks for outgoing on 1.6.1: extensions.conf: exten => _99X.,n,Dial(IAX2/jnctn_out/${called-num}) iax.conf [jnctn_out] type=peer host=iax.jnctn.net username= secret= qualify=yes I'm not using realtime. But CLI: -- Executing [99xxxyyyy at internal:3] Dial("DAHDI/1-1", "IAX2/jnctn_out/1wwwxxxzzzz") in new stack [2009-06-27

I am using asterisk 1.6 at home and would like to send incoming calls without caller id immediately to voicemail (i don't want to use the privacy manager where people have to enter a number). The config examples i found are all for the pretty obsolete 1.0 and 1.2 versions of asterisk. Would anyone be willing to share a config example? Thanks!

I'm using a Snom 320 in a CAP position and the receptionist wants to do blind transfers. OK, no problem so far. Now she has asked me how to UN-transfer a call, as in, she transfers a call and wants to hook the call back before it connects (she wanted to tell the caller additional information for example) I don't think that this is possible as once my dialplan starts using Dial()

Hello Doug, Friday, December 13, 2019, 11:03:37 AM, you wrote: >> This is exactly what I do - “press 1 for a human” >> Works great > I do this as well, but I also do a database lookup to see if the number > is on our speeddial list and if so, pass the call directly on without > the IVR prompts. I do something similar for calls without caller ID, but I was still getting

Hi! I use the following configuration to register my asterisk server to my SIP provider: register => 12345:passwd@sip.provider.com/12345 sip.conf: [sipout-test] type=peer username=12345 fromuser=12345 fromdomain=provider.com secret=passwd insecure=very host=sip.provider.com qualify=yes context=test-incoming extensions.conf: exten => 12345,1,Dial(SIP/10) exten =>

Hi, i've some trouble with an * installation when the following scenario happen. 1) Inbound call to SIP/xxxxxxxxxxxx ; 2) Call is redirected to ring group 6xx 3) SIP extension 1xx answer. 4) caller want to speak with john doe on his mobile 5) assistant put caller on hold 6) assistant start a call to john doe mobile using a php script (AMI - Originate with custom context to force outbound

Hello, I am using Asterisk 11.0.1 and do not notice any changes regarding the Transfer on newer Asterisk 11.x versions. I am using AMI and controlling a channel via AsyncAGI. I send a Transfer cmd (such as the following) Action: AGI ActionID: C8 Channel: SIP/1004-00000002 CommandID: C8 Command: EXEC Transfer SIP/1003 Destination phone starts ringing. If it answers the

I all, I'm trying to transfer a iax2 channel trought dialplan app transfer to another extensions (IAX). The variable TRANSFERSTATUS report SUCCESS but the call isn't trasfered. I haven't other information, in console I see only hangup of a channel. My scenario is 3 asterisk box connected with iax trunk, I talk from BOX1 to BOX2 and I want to transfer user on BOX2 to another user on

Hello friends. I have a softswitch in which I cannot create a list of blocked source numbers; So, I have thought to use Asterisk and return a 302 message when the number can make the call, my dialplan is as follows: [from-external]   exten => _AX.,1,Verbose(=======> ${CALLERID(num)} to ${EXTEN})    same =>      n,Set(MYDESTINY=${REPLACE(${EXTEN},A,)})    same =>     

The Asterisk Development Team would like to announce the release of Asterisk 16.6.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.6.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Hello all, What my app does is accepts a call in on a Dial-In Number (DID) via IAX, and then prompts the caller for the top secret password (123) and then authenticates the user and prompts them to dial in the number they'd like to call. Once they press pound after dialing in the number it will read it back to them, if they press pound it will attempt to connect via the second IAX provider,

Hi, Firstly, what I'm trying to do is: * Get asterisk to pick up a SIP call via a DID * Prompt the user * When the user puts in a number, go to IAX.conf and route it according to what I've specified there, i.e Least Cost Routing, etc. I've set-up something similar to what I've found online, but it doesn't work! Asterisk doesn't pick up the call at all..... :( The files

I receive calls over a T1 with callerid and then *ani*dnis*. I am able to strip out the ani and the dnis in the dialplan but when I try to set the caller ID to be the ani, it looks ok but then if I do a NoOp callerid on the next line, I get unknown. Here is the section of my dialplan: exten => _*NXXNXXXXXX*NXXNXXXXXX*,1,Set(ANI=${EXTEN}) exten =>

Hello All, How can I SET EXTENSION from context? This is my context: - [docall-usa] exten => _NXXNXXXXXX,1,Answer exten => _NXXNXXXXXX,n,Set() ; <<What do I need to set here>> exten => _NXXNXXXXXX,n,DeadAGI(dousacall.php|1) exten => _NXXNXXXXXX,n,Hangup I need to add 1 in front of ${EXTEN} and then send the call to dousa.php. Set(CALLERID(number)=1${EXTEN}) will set