similar to: Call file misbehaviour

Just installed V 0.4 of asterisk@home Programmed up 3 sip budgetone extensions, they call call each other fine. Tried to dial '9' for an outside line through an X100P to a packet8 ATA but got 'all circuits are busy now'. Here is the console output. == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/30-8d25' -- Executing

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

I'm having trouble getting outbound calls going with aah 1.3 and a tdm400p w/ 4 FXO. Incoming calls work fine, outbound I get this: -- Executing SetVar("SIP/231-af2b", "OUTNUM=6643955") in new stack -- Executing Cut("SIP/231-af2b", "custom=OUT_1|:|1") in new stack -- Executing GotoIf("SIP/231-af2b", "0?19") in new stack

Hi, I have some problem to get this setup working. I have a CAC Channel Banl I, with FXO and an Asterisk box ( I am using Asterisk@Home 1.2) and I have a TE110p installed in this box. What I want to do is, just to be able to dial one of those lines that already are connected to the channel bank, and transfer that call through TE110p and Asterisk to a user agent somewhere through Internet.

Hi All, I am a new bee to *. I just installed Asterisk@home on FC3. I hv a FXO card. I hv configured two extensions one x-lite and other iaxComm. I configured * using AMP. The following setup works - x-lite (x 200) to iaxComm (x 201) - PSTN to x-lite - PSTN to iaxComm Voice mail, weather etc work fine. When i try to make an external call i am getting message "All routes are busy". In

I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway

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'm a newbie on * trying to setup an HFC card. I'm locked for many days getting the all-circuits-busy. And no idea what else to look for/how to diagnose. I'm in Spain, I've tried changing many parameters on zapata/zaptelcong with no luck, also NT & TE modes (honsetly, I've no idea what is). Any clue will be very much appreciated! I've installed *@home on my RH9,

Hi, I configured a pair of asterisk box, one of them (the first) ISDN connected to the PSTN. I configured an "integrated" dial plan, and I SIP connected the two pbx. No problems in dialing extensions defined in one of the two pbx from any pbx. My problem is this: How can I make the users logged in to the second pbx to dial an external number using the ISDN connection of the first ? I

Dear List members- I am trying to configure ASTCC (Asterisk calling card application) but having a hard time to configure it properly. My project deadline is approaching and couldn't figure out how to make ASTCC functional. Here are some details what I have done so far. 1) I have installed ASTCC successfully. 2) I can access astcc-admin.cgi script without any problem. 3) I have created

Hello, I have *@Home 1.5 installed and all is working fine for incoming calls and sometimes outgoing calls. Installed in the box is a digium TDM04B (4xFXO Ports) setup as ZAP1 to ZAP4. I have incoming calls coming in on lines 1-4 in that order and outgoing calls prefering ZAP4 then ZAP3 then ZAP2. When i try to dial out to the PSTN from a SIP phone it sometimes works (normally after a reboot)

Hi all, I've recentrly starting to play around with *, when all I wanted is to configure an fritz ISDN card with A@H. Currently I'm stuck at the phase of what do I do with capi after everything is installed. I'm trying to understand how to setup incoming and outgoing calls at A@H since I'm getting a bit lost with the default dial plan. It seems that * answers but disconnect

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm

Hello all, I have been posting some questions about this problems that I cannot yet solve, but I think I have a better diagostic, so maybe someone can give me a clue why it is happenning. I have Asterisk + AMP configured as a PBX with a Customer Center Queue with 4 agents that login/logout dinamically. If there are no agents, queue timesout and gets derived to another queue that somebody

Hello all, I have been searching thru all docs that I can find on wiki and such but can not get an answer. I am trying to collect a date from user input in the form of digits dialed from the phone to use in an agi script to do a database look up. I have tried to use "Get Data filename, timeout, maxdigits " in the agi script. In * console I get message saying playing filename but it

The SIP channels have a "call-limit" parameter (which is badly documented and I haven't tested yet) How can I have the same behaviour for IAX channels? I can't see anything related to it. Ah, I'm using Asterisk 1.2.13... maybe there is something in the 1.4 versions... but I can't change to 1.4 right now because of MFC/R2 BarZ

Is it possible using asterisk to allow someone to dial in and remotely change where their call is forwarded to? For example, I'm working from home so I want my calls to go to 555 1234, now I need to go out for a bit so I'd like to phone the office and using DTMF tell the asterisk PBX to now forward my calls to my cell phone 555 3456 Has anyone implimented anything like this? R.

Hi all, The CentOS community is pretty limited in what we can do to the core distribution. Since our mantra is "aiming to be 100% compatible with Red Hat Enterprise Linux" we cannot fix bugs or improve the CentOS core without waiting for Red Hat to make those modifications first. We have limited leverage and a 6-month release cycle against us. But that is not the complete truth,

Hi: I have a dataset and need to interpolate for missing days. In Excel I either average from sampled days from above and below the missing days or use the TREND function to make up for the missing values. I have been reading about na.approx, is this function similar to the TREND function? Which is the best recommendable way to make up for missing data? Here's my dataset: weeks 17,18,26 and 46

On 2016-12-16 12:52, Ilia Mirkin wrote: > On Fri, Dec 16, 2016 at 12:10 PM, Joseph D. Wagner > <joe at josephdwagner.info> wrote: > >> When using the nouveau driver, my screen freezes (and stays frozen) >> when it >> starts to more than a minimum amount of work. To fix this, I've been >> using >> the nvidia driver with the Preferred Mode set to