similar to: Can't hear auto-attendant

Here is the output. These are Panasonic KX-TG2564's. Does something need to be set for the phones? I can call out fine, but all of the extensions seem to be busy. Starting simple switch on 'Zap/5-1' -- Executing Macro("Zap/5-1", "exten-vm|200@default|200") in new stack -- Executing SetVar("Zap/5-1", "FROMCONTEXT=exten-vm") in new stack

Hi all, as in last mail, i've installed Asterisk from CVS and AMP to manage it. I've made 4 extensions: moloch*CLI> sip show peers Name/username Host Dyn Nat ACL Mask Port Status 204/204 (Unspecified) D 255.255.255.255 0 UNKNOWN 203/203 192.167.125.9 D 255.255.255.255 5062 OK (3 ms) 202/202

Hi all, i've installed Asterisk with AMP. I've created 4 extensions (for 4 SIP phones) and this is what my "sip show users" return: moloch*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT 204 moira from-internal No No 203 michele from-internal No

Hello all, I've installed AAH 1.1 on my VIA C3 powered mini PC. I've made the necessary changes to the * makefile, so the compilation went well. The first thing I did was configuring two extensions from AMP, namely 200 and 201. Then I installed X-lite on two PC's and configured them with one of the extensions: System settings - SIP proxy - Default: Username: 200 Authorisation user:

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

Hi.. I noticed all recording activities suddenly stopped. It seems as if Asterisk is unable to manipulate files. Here is a sample of a session in which I dialed the Voice Mail system and tried to record my name: Any ideas? Thanks Executing VoiceMail("SIP/100-69a9", "b100@default") in new stack -- Playing 'vm-theperson' (language 'en') -- Playing

Hi I am trying to use festivall with asterisk , I am using RHEL4 , asterisk1.2.7.1 and festival-1.95-beta , I am able to hear the voice form the text file , when I dial to the extension, but when I have ?,? in my text file , it plays only the text upto ?,? and in the CLI , the ?,? is shown as ?|? I had cut and pasted CLI messages for reference -- Executing

I'm having trouble finding documentation for the auto attendant does anyone have an idea where there might be some???

All, I'm trying to hear the asterisk's auto attendant in its default configuration. According to VoIP Hacks in Chapter 4, I found the following excerpt after successfully configuring my SIP IP Phone (Cisco 7960G): In its default configuration, Asterisk has an auto-attendant that can route calls. To try it out, take the IP phone off the hook and dial 2. Then dial the BudgeTone's Send

Someone will probably tell you with more certainty, but (you don't say but I assume you are talking about FXO) the Caller ID normally comes in between the first and seconds rings, I think you can tell asterisk not to get the CID but if you don't, it waits for it. Also, I remember reading in a modem manual something about the number of rings you had to let your modem ring before doing an

Hi Folks I have had a look through the Features list, and I see that the system does support an auto attendant, however is it possible to have say 5 telephone numbers that a person would dial and have 5 different messages I.e Dial 555-1121 and you get a message for companyA call centre Dial 555-1131 and you get a message for companyB call centre Dial 555-1141 and you get a message for companyC

Are IVR and "Auto Attendant" interchangeable terms? They both do the "Press 1 for" thing. Sales is asking me how to word it and I've always used both terms interchangeably.

Hi all, been searching & not finding an answer to this, although I'm guessing it's absurdly simple... I just hooked up a T1 to our * box (1.2.0), which had been using POTS lines via a channel bank.. Now when I call the new T1 circuit, there are no rings, the Autoattendant just picks up right away.. Any clue on how to make it ring twice before getting picked up? I tried immedate=no and

Currently when I call in to my * box it answers after two rings. I'd like for it to answer without ringing. Is this an option somewhere in the dialplan that I'm missing? Thanks, Andrew

Hi: I am trying to setup a simple auto attendant with Asterisk using SIP extensions. I have an IP trunk to voicepulse and my outgoing calls go over that. I can also receive calls on that voicepulse trunk and want it to an auto attendant. Everything works except on the following: - one of the options is to allow the caller to press the extension that they would like to be connected to. I have

Hello all, I'm planning to create a simple Auto-Attendant (IVR Menu) for my home PBX yet all callers from incoming (trunk) calls must only press the extension numbers from the [analog-ext] else will play the "pbx-invalid". How do you do that using the GotoIf() (or probably using the other applications) but will check if the numbers entered belongs to a specific context? Also, how

Folks, I have a NEC 2400 pbx(non-voip) behind a Cisco 3725, connected via standard wic-t1 card. The NEC needs to call two different asterisk servers with 4 digits. I have two way calling working with the one * box, but the other is perplexing me. Here's the layout * <--> Cisco 2811(192.168.13.1) <--> 1.54 point to point <- Cisco 3725(192.168.8.1)<-> NEC 2400. The

does anyone know how to route a call coming in with ANI*DNIS* Oct 31 12:47:41 VERBOSE[30056] logger.c: -- Executing Set("Zap/49-1", "DID=1231234*4812*") in new stack I tried making a route for _.*4812* but that matched everything rather then just the dnis i wanted.. any ideas? I would preferably like pass the callerid along to my extensions, but for now the important

I never heard anything on the AMP list, so figured maybe someone here might be able to help me sort this one out.. I was making some updates to my attendant config, which is really very basic, and now incoming call processing stopped. Not sure exactly what the heck happened, but figured maybe someone could help me with a clue as to what broke. Now incoming calls are not being answered at

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to