similar to: Revert to dial tone?

[ Optimised G.729A 'Howlet' for Asterisk & FreSWITCH ] Howler Technologies are proud to announce today the launch of their fully indemnified and highly optimised G.729A solution for Asterisk, including a unique floating license model. This is the first in a series of products dubbed 'Howlets' that add highly performant transcoding and signal processing modules to open-source

hello I dont want to disgrace any company but i want to know from your(user)experience which one is good in case of g.729 (performace etc) is it Howler(http://www.howlertech.com/products/howlets) OR its Digium ( http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC&main_category_id=5 ) plz note i dont want to degrade any company... But to know what experience you

I have an interesting problem. I am attempting to install a payphone utilizing a Cisco ATA-188. The payphone actually works, but there are some timing issues. What happens is you pick up the payphone and the ATA grabs a line and goes offhook. While you monkey with putting money in and dialing the number, you are eating up the time before you get the offhook reorder tones (or howler tones

Is it possible to get an internal dial tone when I call to my asterisk and enter password? I would like to call my line enter extension - password - and get internal dial tone. once I'm in I would like to dial based on what context permits, mostly long distance calls VOIP. I can not preset the extension to certain number as I don't know what number I will be dialing. -- #Joseph

Hi, What i want to do - is to give ability for answered call to hear regular dial tone and be able to enter phone number - that i would dial later. I tried to look at WaitExten and PlayTones, but they seem to not work together - WaitExten doesn't interrupt going on PlayTones. Is there any way how i could do that - so that it looks really natural? It would be silly to create long-long-long

Hi I am using DISA to get my Polycom SoundPoint400 with H323 firmware to connect to * I have it working, but when I dial SIP end points there is no ringing tone on the phone. DISA gives dial tone but does not give ringing (if I understand correctly it is because it expects to transmit sound created by terminating side of the call) Is there a way to make DISA application to generate ringing

I have a SPA-3000 that is failing to hanging up pretty often; almost every day now. The weird thing is that an almost identically configured (same FW, different HW rev) second unit right next to it isn't having the same problem. Swap the lines and the problem stays with the unit. I've been going round and round with Sipura and their latest message told me to use a procedure detailed in

Hi All. I've been experimenting with SLA on Asterisk 1.4.13 (patched up to 1.4.14). I am using a SIP channel for my "trunk" line. On the whole things are good, but I have noticed that if I misdial an outgoing call, i.e. I get 404 "Not Found" in the SIP trace, then the trunk line just drops, rather than presenting an error tone or message to the user.

The way I have my dialplan configured, an internal extension is routed to a different context (with Goto()) on pretty much the first button press. 2 -> internal extensions 0 -> operator 5 -> VM 9 -> outside line etc. So a "201" will go to the internal extensions context, s,1, do some setup and then match on "01". The thing is that when the 9 is entered, I

Anyone know of a good dial plan example for call in / call out? I want to be able to call my Asterisk server, auth, and then call out any number. Michael

Anyone know how to let asterisk deal with the progress codes coming from the carrier? The problem I am having is when a customer calls an invalid number the carrier tells me the call is invalid via a progress code but doesn't route me to a recording (this number is invalid). Instead they hang up on me causing a fast busy or sometimes hold up the call with dead air for 15 to 30 seconds then a

Hi, We had a last-minute cancellation from Vivox for today's conference. It happens that someone suggested a guest idea, Howler Technologies CTO Jay Fenton, who agreed to join the call from the road. Anything you want to know about transcoding to and from g729 is out topic for the first hour. My pal David Duffet knows this technology well and has kindly signed in to help guide us through this

I haven't found this in any docs or faqs yet, so I'm wondering if I can achieve what I would like to do. On an Asterisk PBX with multiple PSTN lines, I'd like to call in from one PSTN line, probably via cellphone and access the PBX as if I were local to it. From here I'd like to get a dial tone and call back out. I know this isn't exactly call forwarding per se, but I'm

Does anyone know where to find an example or able to provide an example of how to do the following: When asterisk answers a call... Ask for number to dial...then dial that number? I am basically dialing into the asterisk box and then wanting it to take the digits I enter and dial them on an outbound zap trunk... I basically am just not sure how to have asterisk accept the digits and then use

DISA is currently disconnecting when I dial 8888 to access DISA. Below is my extensions.conf file from A@H and some lines which shows the disconnect. Should DISA be loaded as a module in modules.conf? When I do a 'show applications' i see that DISA is there. Help! -------------------------------------- ;Asterisk CLI as I placed a call from cell into the system. Playing

I'm having trouble getting an AGI exec command to spawn app_disa. The script executes properly, but does not spawn DISA. The CLI gives no helpful clues. Am I doing the exec incorrectly? I want to have a way to authenticate callers to the extension by Caller ID... if their caller ID is in my database and set to active, they can call out. [like a calling card but auth'd by CID instead

-------------------------------------------------------------------------------------------- Originally posted at http://forums.digium.com/viewtopic.php?t=18045 -------------------------------------------------------------------------------------------- Hi! I'm trying to configure a DISA setup (Asterisk 1.4.11). Only, executing DISA seems to prevent any DTMF detection capability when using

Hey, this is happening to anyone who I try this with. We get into the DISA, then hear the dial tone. Dial 1 then start dialing the number, and it hangs up. I thought adding a wait time after the DISA may help, I was wrong. Here is what I have thus far in the DISA extentions. [DISA] exten => 7,1,DISA(no-password||"Scheda" <565> 455-1337) exten => 7,2,Wait(45) exten =>

Hi Folks, I'm trying to test out Asterisk overall. I'm having some problems with DTMF. Currently I'm playing with DISA, but I'm worried this will happen when I get to implementing AAs etc. I have a free SIP trunk from IPKall that I'm trying to make work. I'm able to receive calls, and I've now setup and extension with DISA and a password. I connect ok from the

Hi, These days I run into something quite odd. I have an A@H that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a