similar to: Read back of caller ID

I would like to setup an option in my digital receptionist that callers can select to hear a read back of their Caller ID. It would be something like, "the number you are calling from is...". I think I can reuse the festival script that is built in, but ideally this could be accomplished without using festival because Allison's voice is so much more pleasant. I'm just a few

Hi all, I'm unable to put app_festival to work. I successfully patched, installed and tested festival (interactive logon and telnet to server port) which seems to work without problems. But when I test it in asterisk I got the following trace in console: -- Executing Answer("SIP/bsenicar-850b", "") in new stack -- Executing

Hello, I am having a Mac 10.6.4 (Snow Leopard). I have compiled and built Asterisk 1.6.2.9 and Festival 2.0.95:beta on my machine. Asterisk is working fine with SIP channels without Festival. I have written following context in extension.conf: [connect-to-me] exten => s,1,Answer Exten => s,n,SayDigits(?1?) exten => s,n,Festival(hello john) exten => s,n,Hangup I use call files to

Hello, Suppose a dialplan such as: exten => 6004,1,Answer exten => 6004,n,Wait(1) exten => 6004,n,SayDigits(1) exten => 6004,n,Festival(This is a test of Festival) exten => 6004,n,Hangup When watching in the CLI, I see this: == Using SIP RTP CoS mark 5 -- Executing [6004 at internal:1] Answer("SIP/505-00000004", "") in new stack -- Executing [6004 at

I am trying to use festival (latest version 1.4.3) I have downloaded all the files needed and patched it with the provided diff. festival does work and does tts fine. but when I call Festival either from an extention or an AGI script, I get this in my asterisk messages log, but no sound on the channels (H323 or SIP) - they (the clients) just say "trying" and then hangup... Sep 24

I would like to setup a test number, that speaks back my phone number. How can I set this up? bye Ronald

>From the queues.com file. ; An announcement may be specified which is played for the member as ; soon as they answer a call, typically to indicate to them which queue ; this call should be answered as, so that agents or members who are ; listening to more than one queue can differentiated how they should ; engage the customer ; ;announce = queue-markq This allows you to have one

I've been asked to look at deploying Asterisk in a high availability environment and I've been looking so I've been searching for methods to decouple the voice PRI circuits from the Asterisk server so failover to another server could take place. I've been looking at the RedFone foneBRIDGE2 2e1 product here:

Hi, I'm having difficulties with using DUNDi between two servers. If it were three I think I could control looping by limiting TTL, but with two I'm not sure how to prevent a loop causing bad things to happen. I've tried ttl=1 but things still blow up. The DUNDi configurations are pretty simple and work just fine in both directions as long as only one of them is using the switch

I need to dial an extension that tells me what extension I'm dialing from. I'm running a bunch of analog phones off a channel bank to * over a T1. I have the following in extensions.conf. exten => 98,1,SayDigits(${EXTEN}) This says the digits the caller enters on the keypad, not the extension they are calling from. Thanks Guys!!!!!!!! Paul Paul Mahler pmahler@signate.com

We have an Asterisk server that speaks IAX2 to Magrathea to get to the PSTN. Our local phones are a mix of Cisco 7940s and Grandstream BT100s all configured for SIP with silence-suppression disabled. Everything is configured to use a-law encoding. The version is: sip*CLI> show version Asterisk CVS-05/06/04-18:45:57 built by root@sip on a i686 running Linux Incoming callers are complaining of

I am running the most recent asterisk 1.2.13 on a Fedora 3.0. When I go into asterisk (asterisk -r), defaults to verbose 3 and I get a stream of messages: Remote Unix connection Remote Unix connection disconnected ... ... (keeps on repeating). I went to google and searched on "asterisk Remote Unix connection disconnected" but cannot find anything I can recognize. I checked my iax.conf

Recently with Asterisk 1.4.17 I've been running into some stability issues. I started looking through my logs, and I found this: [Jan 29 09:41:45] WARNING[13132]: loader.c:620 inspect_module: Module 'codec_g729a.so' was not compiled against a recent version of Asterisk and may cause instability. I'm using the newest version of codec_g729a.so from the Digium website (v33).

I've been digging around and I haven't found a way to do this, but I have a feeling I'll feel like an idiot because it's something I'm over looking. Normally if I need to specify an additional option (such as different language sound files) or I'm building an Asterisk server with a lean configuration and need to remove some modules I do so with 'make menuconfig'.

Got no responses to this, but the list seemed to be down for a while, so here it is again. Sorry for the extra bandwidth! John Hi, I've been messing with getting SIP working for days now, with limited success. I've got Asterisk set up on a remote server with the echo test. Please try it out to verify I've got the server working right: sip:robot at nixon.butchwax.com

hello, I have a test setup with 2 asterisk servers, each having a one snom 100 via sip using it. I`m experimenting on how trunking between them would work. I have them setup for RSA authentication which I plan to use in the future. So I`ve setup the keys and servers seem authenticate to each other. One is named phila and other hurricane. Here is what I see on phila: -- Registered

Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks

My compliments on the release of syslinux-5.0 and happy holidays to the entire Syslinux team.? I have a longer message in me.? Lots of curiousity about the direction of the project, but that will have to wait until I finish my first 5.0 project (*not* my first Syslinux project), which just happens to touch past problems: drive enumeration. This particular effort is on a USB flash drive, with two

I saw a couple of posts about this in the archive, but none seemed specifically to address the problem I am having. If I missed something please let me know. Right now I would classify myself as "novice," and there is probably really nothing so trivial that I couldn't possibly have screwed it up. :-) I'm trying to use the AMD command to detect answering machines, and have

25.12.2014 15:46, Anthony Messina ?????: > On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: >> I want to change call files, which has caller id in them, to call >> originate from dial plan. >> But I don't see such parameter here >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate >> >> How can I pass callerid