similar to: how to collect user entered digits

I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It installed pretty easily and has worked great so I went to order some more of these units today. When I logged into Tech Data this morning, the PAP2-NA was now marked as discontinued and no longer available and only the PAP2 version was available which is the Vonage branded version. :( I saw someone on the list say that

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.

Hello everybody: This could be a stupid question, or may be not; I'm not sure 'cause I have not a very wide experience working with Asterisk, actually I just started last week. I need to make an IVR system work and I choose working with AGIs, written in Perl. The available documentation I've found show it as a very simple proccess, but it doesn't work for me... and I

I'd like some feedback on potentially submitting a request (and probably a patch too) to change the way the AGI command GET DATA works. Right now, # terminates the entry, which is then returned with the # stripped off the end. What I'd like is to allow user configurable termination digits, which are not stripped off the end. Reasoning: Some entries you'd like to terminate with #.

I am trying to setup one time caller id block on my system(activated when an incoming call matches *811XXXXXXXXXX), and I have had little to no luck. Could you take a look at my context/macro definition and help me figure out what I am missing? Here is my context for my dialplan: include=default plancomment=user-default

Hello all, I have seen some chatter again about DTMF. I see most of the talk about DTMF around not being able to get an external IVR to recognize digits, not a big issue for me at this time but sill interesting. My issue though, is with talk off on a zap channel. It seems to be getting worse or maybe my patience is thinning. All my calls go out and come in pstn through an FXO as I do not

Hello, I am just curious how many people are hooked up with BroadVoice and have recently been experiencing a lot of dificulty. Joel

Hi all, I have two internet connections, both connected to eth0 and eth1, one is 1mbit, the other one is 3mbit. Now I want to use _both_ at one time via round robin or such. Anyone have an idea/docu how to set that up? Thanks in advance. -- ciao, Marc _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc HOWTO:

Hello all, OS OpenSuSE 10.3 * ver 1.4.26.2 zaptel ver. 1.12 Digium TE122 I have a request for remote users to be able to dial through the system so that the sales managers can barge/chanspy on the sales force. I have the DISA part working with authentication(rather straight forward) but what I can not figure out is how to enable the supervisors to be able to barge on these calls. Is there a

Hi, For my IVR, I use Read command. It works fine when ending bu # but I can't get anything without ending by # The wiki tell me is it possible with maxdigit option but it don't work for me. my command : exten => 3,1,Read(ILE,as/iles,1) Anybody can tell me howto do thanks Another question about read command: Howto sup file option and keep maxdigits options ? exten =>

Hi, In our office, we're slowly migrating from a cisco call manager set up to asterisk. Problem is management doesn't want to buy any other hardware ?as they had already invested a lot in cisco. The main cause of this is asterisk's added features like unique FAX number for everyone in the company (which will be the same as phone DID), Voice mail, Auto Answer etc yet we need thousands

Hello all, Okay when you are done laughing at the simplicity of this question could someone show me please what I have wrong in the following statement? GoToIf($[${numdial} != [1-9] ]?15:3); What this is supposed to do is if numdial is not a single digit from 1 to 9 inclusive goto 15, if it is a singledigit from 1 to 9 inclusive goto 3. Should be pretty simple but not working for me, always

Hello all, I am looking into building a calendar server (due to business requierments I can not use public hosted calender like Google), and am looking for suggestions based on experience with different calendar applications/servers available for Linux that you have integrated with Asterisk. If you can give a quick, simple list of what worked and what didn't I would be very grateful.

Hi Folks, Consider the following example (artificial, but it illustates the point): > r2<-sqrt(2) > x<-2-r2*r2 > print(c(pi,sqrt(pi)),digits=5) [1] 3.1416 1.7725 > print(c(pi,sqrt(pi),x),digits=5) [1] 3.1416e+00 1.7725e+00 -4.4409e-16 whereas I would prefer [1] 3.1416 1.7725 0.0000 which could be achieved by > print(c(pi,sqrt(pi),round(x)),digits=5)

Hello All, I am looking at doing some video conferencing with SIP. I was hoping to get some early pointers from any one that is currently doing this. I have been all over goggle and voip-info and there is a ton of anecdotal information but, I was hoping for more specifics of what people are actually using that works and even some of what hasn't worked so that I can stay away. What I am

Can anyone point me to a sample or information on using MeetMe like this? Conference room is set up with 2 PINs, one for the moderator and one for the participants. Participants get music until the moderator joins (to avoid wild, un-moderated tangents). Call is ended and all participants are kicked out when the moderator leaves (or the moderator can kick everyone out via phone keypad).

I have allowed ALL of the local users to ping the internet but they currently get the following error and cannot access the internet ! I know it is something I have done wrong (I think it is a routing problem but just cannot find out what) The error is:- Reply from 212.219.13.74: destination host unreachable. My eth1 is 10.0.0.1 and the users can ping that OK My eth0 is 212.219.13.74 (connected

I have asterisk 1.8.2.3 + A102D Sangoma card 2 port T1. when i am starting asterisk i am getting this error on console. func_callerid.so => (Party ID related dialplan functions (Caller-ID, Connected-line, Redirecting)) == Registered application 'PrivacyManager' app_privacy.so => (Require phone number to be entered, if no CallerID sent) == Registered custom function

As I did two years ago, "I'm posting a new thread with the "Top Posting" subject" rather than hijacking the "Paging for Praying" thread. Two questions: 1. Steve K: What do you mean by "/coat"? 2. How do we change rule #5? --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax -------------- next part

Ok we have found a better solution. Put everthing back the way it was and make sure that you have this line in your general section of you sip.conf file: srvlookup=yes We have added a SRV entry in the correct place now. So everyrthing should go the correct servers. -james --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 /