similar to: Does X100P clone provide Timer?

I attempted to install an X100P card but it was not correctly recognized by my Redhat 9 install. I had a test install running without any cards which was working great minus the outward dialing since no cards existed. Now that I have a card, I want to add it to the system. Do I have to scratch the whole current install in order to get the X100P running on my system or is there a way to get it

I''ve got a nice XML view of my data that works just fine when I call it from the browser. Now I''d like a different controller method to take that XML and write it to the file system. Since the XML view lives in an RXML file, is there a trick to telling the controller "Get your output into a string" instead of writing it back to the browser? After writing it out

Hey folks, Using FreeBSD 5.2.1 and I've got the current zaptel driver installed from ports (0.8_1) and current ports asterisk (1.0.1). I've set options HZ=1000 in my kernel config, recompiled and rebooted and as far as I can tell, I've done everything right but when I try to use the conference, the audio is very delayed, choppy and segmented -- totally unusable. At the

So I''ve inherited a legacy application and I''m trying to work around the edges as I put an admin tool interface on top of the existing code base. I install Devise for user authentication, since I''ve used it in the past. I change none of the default code. And yet, on successful sign in, I get an error: Render and/or redirect were called multiple times in this action.

The past two days, I've been having issues with my two VoIP service providers where calls just suddenly hang up. The following is from the log: Mar 14 13:50:55 WARNING[5887] chan_iax2.c: Max retries exceeded to host 64.34.45.100 on IAX2/voipjet-3 (type = 6, subclass = 11, ts=250000, seqno=80) Mar 14 13:50:55 DEBUG[10428] channel.c: Didn't get a frame from channel: IAX2/voipjet-3 Mar

Googling for this has not solved my problems. Is there a particular site that is recommened for its examples and tutorials on Rails'' AJAX capabilities? I know exactly what I want to do, just not how to do it, and am looking for some code to pore over. 1) I have a dropdown that represents the name of an image in a library. When the dropdown value is changed, I would like to refresh

Actually doing both sounds good to me. Can you explain further about ringing them all at once? Here is how I tried to make mine work and failed... {global} PHONES0=SIP/2000 PHONES1=SIP/2001 [local] exten => 6001,1,Dial(${PHONES0&PHONES1),20,trf) When I dial 6001 I see my debugger tell me that I am using the wrong syntax. Do you know the correct syntax for ringing them all at once? I

My Alert object has_many Links. I''ve got the whole system setup where, once I have the Alert object created, I can use a nice form_remote to show the user all the Links currently attached, and add more. But the form itself starts out with 0 links. What I''d like to do now is construct the form with a couple of default "Place link here" objects in the array, and

That could be it. What I want to do is set a group of callers and have the event cause the phone to ring them in order. I will tie it to my IVR portion and thus I can make sure peole in sales get calls based on our hierarchy in the office. So if I am reading your example right the syntax is.... Exten => 501,1,Dial(SIP/PHONE1&SIP/PHONE2&SIP/PHONE3), rtf) Is that a valid way to cause

Hi, I am running Asterisk on Fedora Core 3. I am trying to use DDD to debug Asterisk. However, when I attach the debugger to the Asterisk Process, the Asterisk CLI promt hangs. Does not give any output, and Asterisk stops processing calls... What could be wrong and what is the best way to debug Asterisk...? Appreciate pointers.. Thx a lot, J --------------------------------- Do you

Does anyone have an update patch file to get Spandsp installed? I'm running asterisk CVS-HEAD-11/19/04-21:53:37 on redhat 9.0 I installed spandsp-0.0.2 when runnig the patch I get patching file Makefile Hunk #1 FAILED at 41. Hunk #2 FAILED at 69. 2 out of 2 hunks FAILED -- saving rejects to file Makefile.rej -------------- next part -------------- An HTML attachment was scrubbed...

Anyone knows where can I find spandsp? Official site seems permanently down... TIA, Simone.

How to do it? 25.04.2018 14:33, Felix-Antoine Morin <antonixfelinski at gmail.com> >Its okay now, it is working... ;) > > > Au plaisir, > Félix-Antoine Morin > > > 769 Villeray > Montréal, Qc. > Canada > H2R 1J2 > > (514) 690.36.89 > > felixantoinemorin.com > Facebook <https://www.facebook.com/felixantoine.morin> > >

Hi there, (sorry for my bad english speaking) I was broadcasting my Iphones audio feed to the Icecast server : http://104.236.126.104:8000/ or http://104.236.126.104/ (port 8000) But now it seams like that server is dead... An other developper set it up for me, so I don't know if we just need to re-activate it by paying or else... It was working in December 2017. I use the application

hello I'm trying to figure out if anyone's accomplished putting someone on hold with a hardphone that doesn't have a hold button or multiple lines. I'm thinking transferring the caller to a specific extension or something...is this possible? Has it been done? thanks hank

Hello group I'm working on getting festival installed and working on my FC1. I ran into a problem and after searching Google I found this message talking about a patch for Speech Tools and Festival http://lists.digium.com/pipermail/asterisk-users/2004-May/045134.html The above site does not have the files. Does anyone in the group have this patch? Marc Sutter & Reed Wade do you still

Hello, First, sorry for my english. I'm a french student. I have a problem with asterisk. I use Budgetone SIP phones. When I dial 555 (VoicemailMain), I hear "You have 5 new messages, 1- Read your messages, 2- , etc ... ) But when I dial 1 or 2 or everything else, nothing happen. Are they some lines wich do that asterisk listen my phones ? Thanks for your help, have a nice day Thomas

Using CVS from 7/12/04 and trying to get the voicemail broadcast feature to work. Voicemail.conf has [mycontext] 3722 => 1234,BroadCast Test,,,cc=*@mycontext . then many other voicemail boxes. ----- whenever I leave voicemail at box 3722, only box 3722 gets the voicemail. It is not expanding it to other voicemail boxes in the [mycontext] context. Even if I replace the cc= line with

I ahve been searching to no avail for a referenc eon how to setup a part of my dial plan that will ring certain groups of number based upon the context. Essentually, I want to be able to designate 3 people as sales and have my IVR handoff and ring their extensions in order. Then maybe I will ahve a couple of people I group together and have them ring if someone selects 2 on the IVR for tech

HI ALL; Anybody can explain the difference between "call parking " vs "call transfer" Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040720/2975991b/attachment.htm