similar to: Ring back when free?

Hi. Has anyone had any success in implementing a callback or ringback function in Asterisk? I've had a look at the callback-voicemail example on voip-info.org http://www.voip-info.org/wiki/view/Asterisk+tips+callback However it won't quite work for me. I need it for local SIP users which most of them don't have voicemail. If one SIP user calls another SIP user and the second user is

I am using Asterisk (Debian unstable packages) with an OH323 connection to my provider. Everything is working except for the generation of ringback tones when I receive inbound calls from the PSTN. My provider tells me that we're sending call progress indications and that because of this they're expecting us to generate the ringback tone. Does anybody know how to configure this in

I have a line from BT (UK) connected to my asterisk system, on a TDM400P. I am able to see either distinctive ring cadences or caller ID but not both. If I try to enable both, all drings show up as 0,0,0. This is a pain because, if I make a call out over that line and the number I call is busy, I can elect to camp on it (ringback), which results in a different cadence of ring when the called

Hi, What is the best way to implement Automatic Redial on No Answer ? Looking at http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-12.txtI can see how Automatic Redial on Busy could (should) be done. How would you do it on No Answer ? Is there any event you should SUBSCRIBE to so that you're notified that you're callee is available ? What if you ask to be notified

I have a puzzling situation, and would be grateful for any insight. I have a dialplan that forwards an incoming call out to another number via the same SIP trunk as it came in on. e.g. [from-siptrunk] exten => 0123456789,1,NoOp exten => 0123456789,n,Dial(SIP/siptrunk/0987654321) Now, if I use a different SIP trunk for the outbound call, than the inbound call came on, the call is set up

In article <20151125133008.6369360.14455.17239 at gmail.com>, Israel Gottlieb <isrlgb at gmail.com> wrote: > Try putting progress instead of answer Yes, I tried Progress already, and it didn't help. But thanks for the suggestion! Tony > I have a puzzling situation, and would be grateful for any insight. > > I have a dialplan that forwards an incoming call out to

Hi, If there are 5 asterisk servers on the local net and each server runs meetme, eg. 3311,3321,3331,3341,3351 respectively. Can I connect these 5 meetme conferences to one meetme using IAX2? Regards, Zen

I'm not sure if this is an Asterisk thing, a handset thing or a telco thing, so please be gentle with me if this is not the right place to ask ..... When placing a call over a SIP channel to a mobile phone, if the phone is engaged, it does not return a BUSY status straightaway. Rather, I get a ringing-out tone for the timeout duration specified in the Dial() statement; *then* I get

Hello list members. I would like to know how to playback an audio file to the caller, and while it's played asterisk to continue executing the next priorities on extensions.conf That's not the case when using "playback" command, because the next priority is executed until the audio file ends playing. I want to evaluate some variables while caller hears the audio file. Any

Hello List, I have seen from the following link that, for SIP channels there is no audio communication possible in MeetMe with AGI_BACKGROUND. http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe Currently we are using asterisk-1.6.2 and the problem still persists. Is there any solution available to overcome this problem? According to our requirement, we have to run an AGI script in MeetMe.

Thanks a lot Tony and Dhaval for your much appreciable suggestions. Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com Date: Wed, 20 Apr 2011 13:55:25 +0530 From: DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>

Here is the call flow: [ivr-incoming] exten => s,1,LookupCIDName exten => s,2,DigitTimeout(2) exten => s,3,ResponseTimeout(10) exten => s,4,Wait(1) exten => s,5,Background(custom/ivr-incoming) exten => 1,1,Background(pls-wait-connect-call) exten => 1,2,Dial(${RINGPHONENUMBERS},20,r) exten => 1,3,Voicemail,u${VMBOX} exten => 1,4,Hangup Running * 1.0.5. The calling party

We come into this section of the dialplan: exten => 88670333333,1,Wait(1) exten => 88670333333,n,SayUnixTime exten => 88670333333,n,NoOp(If you know the extension ...) exten => 88670333333,n,Dial(${PHONE_6003}) The caller from the GK hears only ringing, not the time. The extension 6003 rings and I can pick up, but without any voice nor video. athome*CLI> -- Executing

I am trying to make an Inalp Smartnode 1200 (SIP-to-ISDN gateway) work with Asterisk. It works ... Partially. We are using the Inalp to connect ISDN phones, it basically acts like an ISDN ATA. First of all, when I make a SIP call to the unit with a simple Dial() command (no "r", so Asterisk shouldn't generate its ringback tone) I hear Asterisk's ringback tone anyway (I'm

Hello! An extension registered at asterisk 13.23 initiates an external call (pjsip). After the Invite, the callee (-> ISP) sends a 100 Trying 183 Session Progress (*without* SDP) Asterisk now sends to the extension: 183 Session Progress (*with* SDP) 183 Session Progress (*with* SDP) (really two times) The callee meanwhile sends 180 Ringing (*without* SDP) which is

Hi, I'm using OH323, mostly with success, to interface Asterisk to a provider's switch (World Telecom INX). I have noticed a particular effect, and I wonder whether anyone else has seen the same? The effect is audio flutter (almost like the flutter one gets on MF or HF radio sometimes) which only happens intermittently. Audio coming into Asterisk is unaffected, as proved by using the

Does anyone here know anything about GEN-GEN analogue circuits, also known as Manual Ring-Down (MRD)? Apparently they are widely used in Hoot'n'Holler systems for financial dealer-boards. I have been asked to try and interface to such circuits, and have been having great difficulty locating any specifications for the interface. Apparently, they are always-on 2-wire analogue circuits with

In article <E02FA554-9D6D-4E7D-8A78-5FBDE1DE939D at kicp.uchicago.edu>, Valeri Galtsev <galtsev at kicp.uchicago.edu> wrote: > > > > On Sep 26, 2020, at 8:05 AM, Jerry Geis <jerry.geis at gmail.com> wrote: > > > > I have a disk that is flagging errors, attempting to rescue the data. > > > > I tried dd first - if gets about 117G of 320G disk

It gets definitely better every day. List of bug fixes follows: Release 1.0.5.9 7/26/2004 If SIPRegister doesn't proceed due to conditions unmet, release channel resource Fix the LED flashing issue when connection to the SIP proxy is lost. Fix the issue where the device will not resume registration when it loses connection to the outbound proxy for some time. Fixed the

Does anyone know how a program, script or shell user can best determine whether the machine is running on bare metal or is a VMware guest? Cheers Tony -- Tony Mountifield Work: tony at softins.co.uk - http://www.softins.co.uk Play: tony at mountifield.org - http://tony.mountifield.org