similar to: Zaptel Config

Is there any method in Asterisk to enable the updating process into another SQL database through entering IVR options during the call. Thanks a lot. _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy!

I found "7777" and "*65" are not working. Please how can i re-enable them again. Thanks _________________________________________________________________ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you.

How can i activate "ChanSpy" to spy on a dedicated extension? I see the following in "/etc/asterisk/extensions_additional.conf" [chanspy] include => chanspy-custom exten => 501**,1,Chanspy(801) exten => 501**,n,Hangup exten => 502**,1,Chanspy(802) exten => 502**,n,Hangup But when i try to call "501**", it doesn't give any response. Thanks.

Suddenly i found an error while booting, it says: Fuck: can't open /dev/kmem for read/write (2) So this is why, the Asterisk and Zaptel can not start. Any Suggestions Please Thanks a lot Torintino _________________________________________________________________ Windows Live: Make it easier for your friends to see what you?re up to on Facebook.

I am using an IBM Server, after while in the MBR it said that Event logs are full, so after clearing it, the asterisk can't run. i think it deleted a file, so which file i have to create again. and what's its chmod. Thanks _________________________________________________________________ With Windows Live, you can organize, edit, and share your photos.

I have setup my asterisk box using freepbx. I can call extension and make outbound calls. the outbound calls drop between 10-30sec. we are using bandwidth.com and they have logged our call. below is your bad followed by what they say is a good call. I can't figure out where the problem is on your end. I know we are missing some stuff at the bottom but I don't know where to start.

Hi people, I'm executing some parallel Originate async, is there a way to know the result of each originate?... I was looking at the OriginateResponse event, but I don't know how to get it from my web service. Also, if I have 3 originate in parallel, how can I identify the OriginateResponse of each one? Thanks in advance... Anahi Ludue?a

Hi people, I need to update the voicemail.conf from the UpdateConfig Action (AMI). The problem is that I executed: Action: UpdateConfig srcFileName: voicemail.conf dstFileName: voicemail.conf Action-000000:append Cat-000000:test Var-000000:exten Value-000000:>999,test But I don't see the changes in the file. Can anybody tell me if there is something wrong in that code? Thanks,

Hi Guys, I am getting a complain that call on analogue lines (Sangoam A400D) drops all of a sudden. Here is what I see in logs: [Jul 28 15:49:08] DEBUG[21438] dsp.c: ast_dsp_busydetect detected busy, avgtone: 75, avgsilence 135 [Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Executing [h at macro-dialout-trunk:1] Macro("SIP/2111-b6a400b0", "hangupcall|") in new stack [Jul

Hello, I have a problem in Asterisk 1.4 each day I need to restart *asterisk service asterisk* restart in order to unblock the calls My question how can I do in order to check the issue, and if there is any tool or log? Thanks and regards. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I have Asterisk 1.6.2.13, I need to setup a queue of (6) agents, Ring All strategy, I need to set the maximum total time for the caller (Ringing/OR Waiting) on the queue is (2) minutes before going to a fail-over which is a Ring Group of external numbers. How the total max time is being calculated in terms of the number of agents, Ring Strategy, Agent Timeout, Retry, etc.. Can you

Hi everybody, What I need to do is to run a context where I'll pass some phones (for example: 3 numbers). I need to make something like a followme, if the first phone is not answered, I'll call the second one, and so on. That dial plan is not the problem, my problem is when I execute the AMI, I'm using the Originate. It needs a channel as an argument, so the context can be executed;

Hello I need to write a script that will dial a list of customers and play a message. I couldn't find a way to tell Asterisk/Zaptel to wait until the callee has actually picked up the phone before proceeding with Playback(): ============ ;call made through Dial(): Doesn't proceed after off-hook/hangup [internal] exten => 8888,1,Dial(Zap/1/${IPPI}) exten => 8888,n,NoOp(We never

How can i upgrade from PHP 5.1.6 to PHP 5.2.9 on CentOS 4. Thanks _________________________________________________________________ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear All, I am using Centos 4.7, i have an issue when i run yum update, the URL times-out even when i browse it on firefox #yum update php Setting up Update Process Setting up repositories http://mirror.centos.org/centos/4/updates/i386/repodata/repomd.xml: [Errno 12] Timeout: <urlopen error timed out> Trying other mirror. Cannot open/read repomd.xml file for repository: update

I was Using freepbx-2.1.1, Asterisk 1.2.29 and Asterisk Addons 1.2.9 i tried to upgrade to Asterisk 1.4.22.1 and Addons 1.4.7 all of the IAX trunks got not working at all. I tried to downgrade by "make clean; make; make install" in Atserisk 1.2.29 directory.but "make" gives errors in the end. How can i downgrade asterisk again and undo all changes i made?. (in steps

Hello, I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it in wav-audio at the Asterisk server. I found the loudness level of the recorded audio was too high comparing with the orginal audio. How can I ajust it, so that there will be no amplifier used for recording. Thanks a lot. best regards Felix -------------- next part -------------- An HTML attachment was

Dear All, I am newbie to Linux Clustering, i have 2 standalone CentOS servers, i want to setup a cluster on those servers, to synchronize between each other, and to make a one as standby to the other, if a one fails the other will switchover. I will mostly use Apache, Mysql, and PHP. I have read the "Cluster Administration" document, i found that there are multiple methods to setup

Hi gang, We are moving our 1.4 asterisk with DAHDI over to 10.0 with SIP. Everything is going nicely except that I can't get NV_FAXDETECT to compile properly into 10.0. Because of this, I will have to have my receptionist manually transfer incoming faxes. Any suggestions? Thanks in Advance Danny Nicholas -------------- next part -------------- An HTML attachment

HI, R-Users, I have one problem: I have written the the programs which process all file in one directory: for example: setwd("C:/Documents and Settings/lma/My Documents/Vappu-saved/Log") as the start. .......... But I have many folders like "Vappu-saved" and there are a lot of files in each directory. What I want is using the same program what I write for the above