similar to: Menu for call forwarding or voicemail

I'm trying to add a second extension to my setup. The second device is able to successfully connect to the Asterisk server. I am unable to contact extension 101 from 102 and vise-versa. Also are my context setup logically or is there a better fashion to organize them? My error is at the bottom. Here is the extension.conf [default] ; ; By default we include the demo. In a production system,

I have a Asterisk server setup and I am able to connect to the server using a soft client 'x-lite' and call and leave a message on my second extension 102. I have setup a Vitelity account and add what I believe to be the correct information to my sip.conf and extension.conf. I would like to setup incoming and outgoing calls with voicemail support. I've searched all over but many of the

I've searched around and found a few similar situations where the phone will call out when using a Asterisk server but not receive inbound calls. My issue is a little stranger. If I call out from the phone then the phone will receive the next inbound call. The phone will not receive another inbound call until a call out again from it first. Any ideas? I am using SIP and am using the latest

Is there a way to include: _NXXNXXXXXX _NXXXXXX _011. _911 into my current plan:

The host I am working with has two accounts from the same DID provider. Incoming calls work correctly and dial the appropriate extensions. This also allows incoming calls to be billed appropriately to the individual DID accounts. Outgoing calls from either extension default to the first DID, i.e. calls from either extension have the same callerID. How can an extension specify separate outgoing

My latency is kind of high and the voice delay is noticeable. The Asterisk server is on a dedicated host outside of the network. I am performing PAT/NAT using a Cisco router. ns1*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.116

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

I had this problem some last year and never got it figured out. Now it is bugging me. It seems that sometimes when a student writes his/her file to a directory, it will not keep the correct group. It puts his/her main group as the group owner and that fouls things up. Here is what I have. Unix Permissions /school 3777 admin.teacher /school/bhs 3777 admin.teacher /school/bhs/reese

Ernest, I hadn't thought of doing that, though having that added protection would be nice. However, what I'm trying to do it have an incoming call at my home number follow me to my cell phone for selected numbers -- Since I already have three way calling, I'd like get Asterisk to essentially three way my cell phone into the call (or my office number, etc.) I understand the

Hi, I've been trying to use the FaxGateway application to send T.38 out over Zaptel using asterisk but I don't seem to be having any luck. I'm executing it in the dialplan like: FaxGateway(Zap/g0/[number]) Has anyone had any luck using this thing and can enlighten me on how it's supposed to be used? Thanks.

I'm looking for a source for 50-pin amphenol cables, the ones used to connect Adtran's to punch down blocks. Preferably, one that's mail order and takes orders over the internet. Thanks.

Hi, all. I'm probably doing Something Dumb(tm), so please feel free to point out whatever I'm missing, no matter how stupid. Anyway, I've got IAX set up to Vitelity. When I try to call my DID, I get: Rejected connect attempt from 64.2.142.19, who was trying to reach '6031234567@' This leads me to my first question -- why doesn't it show a context? (My second is,

Greetings, Currently my asterisk box is using Voicepulse. It works fine with the exception that people need to enter the 1+area code for local calls. I'd like to get around this if possible. The following is what I have in my extensions.conf.. exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=6162997590) exten => _1NXXNXXXXXX,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN}) exten

On 2011-08-09 16:48, John Criswell wrote: > On 8/9/11 6:49 AM, Bjorn Reese wrote: >> I have an optimization pass (FunctionPass) where I need to add global >> constructors. >> >> For cleaness sake I decided to add these in my own module. My module >> is created in my FunctionPass constructor: > > This is not how I would do it. A FunctionPass has

Hi All, I'm currently having trouble with dropped phone calls. The following error message is always in the log. This is a Grandstream GXP-2000 Firmware 1.1.6.16 . The Asterisk box is currently 1.4.22-rc5. The problem has been occurring on other versions also. [Sep 19 15:48:02] WARNING[13657]: chan_sip.c:1958 retrans_pkt: Maximum retries exceeded on transmission 8acaea6dc4c6e9b5 at

Hi All, I'm stumped on this and I looking for some clues to fix this. This is a new install of Slackware 12.1 onto an IBM x330 Server. Asterisk 1.4.20.1 plays the wav files and the Cepstral_Allison Swift just fine, but when I play the gsm files the audio quite choppy. And, the files produced from the MixMonitor don't even record any audio other than noise. I have a hard drive from

On 8/9/11 9:15 AM, Bjorn Reese wrote: > On 2011-08-09 16:48, John Criswell wrote: >> On 8/9/11 6:49 AM, Bjorn Reese wrote: >>> I have an optimization pass (FunctionPass) where I need to add global >>> constructors. >>> >>> For cleaness sake I decided to add these in my own module. My module >>> is created in my FunctionPass constructor:

I want to do something like: exten => _NXXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _Nxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _1NXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _011.,1,Dial(Dial({TOLL}/${EXTEN}) exten => _9NXXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _9Nxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _91NXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten =>

I connect Asterisk to my analog PBX using X100P. In my analog PBX, I need to dial 0 (zero) to pick up the line. How can I use Dial command to dial (0 + telephone number) directly? I used exten => 10,1,Answer() exten => 10,2,Dial(Zap/1/0) exten => 10,3,Hangup It works, but I need to dial 10 and after the ring tone, the telephone number How can I do?

Hi, I have just signed up with Voxee.com and have attached my Asterisk server to dial them via IAX2. Below is the start of the log which dials the number and promply hangs up when the call is answered, with the logs saying that the channel is not compatiable. I have traced this down to the g.729 codec which I don't have installed. Any ideas on how to force that the codec not be used?