similar to: How to set asterisk NOT to answer incoming lines?

How do you set the expiration date in ASTCC? DO you have to customize the CGI script? A maintenance fee field would be nice as well. Anybody? -- Kelly D Griffin Network Engineer Tantella Wireless http://tantella.com 479.273.9992 Voice 479.464.8998 Fax

I have just setup music on hold by downloading and installing mpg123 r Now I have music on hold but it sounds terrible - clipping, buzzing, digital distortion, and its too loud (which probably isn't helping) and I'm just running it thru the 'default' line in music onhold.conf line default => quietmp3:/var/lib/asterisk/mohmp3, with the default mp3s. This is a standard 1.0.3

Hello This might not be a 'pure' * question, but it is relevant to general VOIP technology. I tried x-lite on my notebook with wireless connection(802.11). The software has been tested with the fixed line connection. It worked fine to call through *. When using wireless connection, it is clear on my side using notebook; however, there is loud noise on the other side of the call which uses

Does anyone have any experience with iax.cc/sixtel? Are they a legitimate company? From their website it looks like you can get a private incoming 800 number for 30 cents/month plus 2 cents/minute. Somehow that pricing seems a little cheap for a DID number. I assume there has to be some minimum usage or something. Any info as far as actual costs and/or voice quality would be appreciated.

Hello, I have some problem using ASTCC application. I've installed the application and everything works well. I've created card numbers, routes trunk and others. When I dial the desired number (77) in my case, I'm prompted to enter my card number. All goes well till I'm prompted to enter the destination number. When I enter a destination number, the system says it's not a

Im experiencing a problem with a current setup and I've run out of ways to debug it and come to a resolution. I have two E100P's in a machine which is routing traffic over the internet to a machine that has 1 E400P connected to the PSTN. Clients are able to make calls successfully but when the call is connected experience one-way audio. They cannot hear anything said by the person

Has anyone had any experience with wireless LANs and Asterisk? We have and here are my impressions. We configured an Asterisk in the office as a precaution to see how it would work for our own retail customers. Our office is open space, about 800 sq ft. (20x40 area). We use Snom200 and Grandstream SIP phones. Using the latest Linksys wireless access point (WAP54g) and 3 wireless bridges

I have an asterisk box and SIP / IAX2 phones. To call out, users have to add 0 (zero) before a real telephone number. That means, that if they want to call someone that has a number 123456, they have to call 0-123456. Simple, right? This has a serious drawback though - when someone calls us from the number 123456, we see the callerID 123456, and we're unable to use the callback/redial

Hi, I am quite new to asterisk so I am not sure what is needed to figure out this problem. If more information is needed and not provided I will gladly provide it. I have a very basic asterisk setup. 1 x100p card and a grandstream handytone 286. I can make calls fine to most phone numbers from the handytone device the trouble seems to come when I dial this number 591-1079. It puts me through to

Hey everyone, I haven't seen an issue quite like mine, so I am hoping anyone who used the Polycom 601's may have an idea. We are going to be switching our office over to Asterisk. All the phones are going to be 601's, I am going to set up a boot server, but for now I am just going to test everything on one phone. My question is I have the phone registered in Asterisk (phone icon

Hi Group, I am sending my question again why I don't have answer yet: I am developing a application, this use "Manager API" to connect with Asterisk. But when I call to an external number (over a zap channel), I don't receive any event when the target answer, Who can help me?, Which event notify me that the phone call was answered? Thank you. Ezequiel -------------- next

Is there anyone with any experience of using the AMD app and the settings that worked for them in the UK ? Any help would be appreciated. Julian

On an Asterisk 1.2.12.1 system using a TDM400P with two FXO and two FXS modules, servicing two POTS lines: When dialing a number, such as a bank, or pharmacy, where it is required to enter a long series of numbers via the phone's keypad, an unexpected hangup occurs. The hangup does *not* occur when entering the numbers, as one might initially expect, but after the called end begins to

Hi Guys, i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way: 1) use a phone in PBX1 2) call extension in PBX2 3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a cellphone) my questions now is : am i gonna be able to dial from an IPphone registered within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2? anybody know

How will this fare? I am planning on putting an asterisk box for my brother in the Philippines but they only have dialup internet. I want them to be able to use a telephone set on a phonejack or linejack card and call me and vice versa via VOIP. My setup in the US is working already with a broadband cable connection. I am thinking that dialup may not work because of the bandwidth required

Hi there, I've got a small problem with the zaphfc channel. No MSN of an any incoming call which comes trough the ISDN card (Acer ISDN, with HFC chipset and zaphfc driver) which will be forwarded to the SIP-Phone will be displayed. Always it will be shown "asterisk" an the Display. --- snip (zapata.conf) --- [channels] language=de switchtype = euroisdn signalling =

Hello, I'm trying to receive faxes with asterisk. My configuration is like this: PSTN fax -> ISDN -> Cisco router with VoIP module -> Asterisk When I try to send a fax from PSTN fax I got the standard fax signal, Asterisk starts rxfax application and then call ends and there is no tif anywhere. On the fax display there is still one message: Calling... Part of my extensions.conf:

Dear All, I'm using Asterisk 1.4.16.2 with Zaptel 1.4.7 on Debian with kernel 2.6.18. I have two analog lines Zap/1 and Zap/2 as group 1 in zapata.conf. I'm using below context for dialing out. [outbound-local] exten => _9XXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten => _9XXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten => _9ZXXXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten =>

When i made internal call into my LAN using x-lite sip phone client I retrive in askterisk CLI : ----------- ERROR ---------- Verbosity is at least 6 -- Remote UNIX connection -- Executing Dial("SIP/201-979d", "SIP/201|60|t") in new stack -- Called 201 May 29 18:09:28 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call

Todd How do you have the context in the phones sip configs set? Bryant From: "Todd Reese" treese65 at gmail.com Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk