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I've been looking around for an example of a method of reading back a caller ID value, but I haven't found anything that doesn't use Festival. I'd rather not resort to the Mr. Roboto voice if I can avoid it. Playback of the numbers one at a time is perfectly fine, so I'd like to use the default female Asterisk voice (the sound files are in place on my server). Does anyone have

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:

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'.

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).

Hi, I recently installed a TE120P in my lab with a full voice PRI (23 channels + 1 D channel). Everything is working well except echo cancellation; for the most part this isn't an issue unless one of the users is in a conference. I'm getting the following error when a call is picked up (incoming or outgoing): [Feb 25 12:54:01] WARNING[8661]: chan_zap.c:1437 zt_enable_ec: Unable to

Hi, Is it possible to dial voicemail from a particular phone line and automatically enter the extension that is being dialed from, thereby only prompting for the password? I've been searching around to find if this is possible, but I haven't been able to find an example of this. I have a feeling it's more of a endpoint function, but I thought I'd ask if anyone has accomplished

Hi, I'm putting together a fax solution for my company that utilizes T.38. I wanted to throw out my plan and get some feedback if anyone is doing something similar or sees a blatant problem with it. We're currently rolling out SPA-942 phones for the standard desk phone with vanilla Asterisk 1.4.15 (just upgraded it today) on the back end. Most calls for satellite offices are handled by

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

Hi, I've ran into an issue with a PRI provider in a major metropolitan area that I haven't needed to deal with before and I was hoping someone might have some insight on how to handle this within the Asterisk dialplan. At this location users can't always tell if a number is long distance or not (there are a lot of area codes and prefixes in the vicinity). Additionally, users are

Hi, I've been playing with a TE120P on Asterisk 1.4.18 with zaptel 1.4.9 and I've ran into an issue. After a call placed any DTMF tone causes the server to lock up entirely. Calls placed work just fine (except for a problem with echo cancellation). The phone registered to the server is a Linksys SPA-942. I am seeing in zttool some IRQ misses, but it never seems to go above 74 (below).

Hi, I've recently come across LDAPget (version 2.0rc1) and I've been trying to get it functional in my test environment (Asterisk 1.4.15 and MS Active Directory 2003) but I can't seem to get it working. I put together a test extension to try to change the CALLERID(name) by way of a LDAP query to AD: extensions.conf exten => 100,1,Answer() exten =>

Hi, Is anyone still using backticks on 1.4? Or is there another way to pull a variable from a shell script into Asterisk 1.4? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080218/bd9a92f8/attachment.htm

Hi, I recently found someone was using one of my Asterisk servers to make international calls via some SIP method that allowed them to bypass authentication (running 1.4.21.1 so I'm not sure how they did this since the major vulnerability for this was patched in 1.4.18.1). At any rate I caught it the same day they started this, so I've blocked their IP range and put in some monitoring

Hi, I have been looking for an example of accomplishing this, but I've been unable to locate something similar to what I'm trying to do. Here's the scenario: Users caller ID is set to their internal extension (200-250). This is set in sip.conf for each user. Each user has a local DID as well (hosted through Vitelity, for example (555)111-2222). The problem is that this extension was