similar to: AGI pdf book

Hi i have problem with AddQueueMember logic. I need login Agent(Member) in asterisk. use this option: for example: AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13) and now i want to call to this Agent: exten => _1XX,1,Dial(Agent/${EXTEN:1}) call to 113 and asterisk should call to Agent => 13 on interface SIP/ekiga. This doesn't work, How can i do this on Asterisk 1.4(not

Hi Sorry to use this public place, but IRC and emails to webmaster at digium have not helped in the past. I have several issues with using the files server downloads.digium.com, which has replaced the simple ftp/http file server ftp.digium.com. In downloads.d.c the directory listing is served through a seperate per-directory script with an obscure name. Let's look at

Hello Fellow Asterisk Mailing ListMembers, When I tried to download the latest version of Asterisk this is what I get: http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz Opening fileinfo database failed http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz Opening fileinfo database failed Where are all the latest Asterisk 1.4.x source files? Thanks in advance, -E

I figured it out. The ftp site was not named well and corrected. The other problem I have it after the extraction and make; it was suppose to go under /etc but that did not happen. I am trying to figure out why. On 12/28/07, broadband Voice <broadbandvoice at gmail.com> wrote: > > I successfully downloaded the Asterisk package from Digium but not able > tar zxvf zaptel-*.tar.gz.

Hi, all I want to add more than 200 sip accounts into sip.conf, username from 6000 to 6199, password is the same, i remember there is a better way to do this case, however, i have not searched the method yet. Anybody can tell me this method, TIA. BR Mike Li -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have an account with FWD and I have configured my SIP.conf with [fwd] type=friend secret=password username=901835 host=fwd.pulver.com But when I am trying to dial out my own DID , I dont see any call landing in asterisk. In extension.conf (vicidial) file I have exten => 2062036895 ,1,Ringing() exten => 2062036895 ,2,Wait(1) exten => 2062036895 ,3,Answer() exten => 2062036895

> On Feb 4, 2015, at 5:43 PM, Warren Young <wyml at etr-usa.com> wrote: > > SSH as shipped on CentOS doesn?t allow 1,000 guesses per second, as this calculator assumes Hmm, just thought of a counterattack: If CentOS?s SSH currently allows 10 guesses per minute *per IP*, all you need to do to get 1,000 guesses per second is to rent time on a 6,000 machine botnet.

What is the syntax to progress with a dial plan after hangup please? Michael

Dear All, I have the following CPU info on my asterisk server: Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 EST 2008 i686 i686 i386 GNU/Linux I need to install G729 on the asterisk server just to pass through and not for encoding...Which G729 package do you advice me to install? I tried several packages with no luck Regards -------------- next part -------------- An

Hi all, I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way

Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me

Hi there! I am new to the list, and to OpenSolaris, as well as ZPS. I am creating a zpool/zfs to use on my NAS server, and basically I want some redundancy for my files/media. What I am looking to do, is get a bunch of 2TB drives, and mount them mirrored, and in a zpool so that I don''t have to worry about running out of room. (I know, pretty typical I guess). My problem is, is that

Dear All, I would like to ask please if someone has a AGI script that select a value from a database and dial this value as a destination number Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090220/e2aa530c/attachment.htm

Hi all, I took my decision to use Asterisk server for handling my VOIP calls...My next step is to choose the best hardware that I should use i order to have the best performance...Here I faced 2 choices for my hardware (CPU)... 1- Using Intel CPU or AMD 2- Use 32 or 64 bits Can you help me please to choose between the above choices and what is the advantage and disadvantage of each of choices

Here is the current dial plan section: [custom-michael] exten => _900,1,Playback(custom/extn-xfer) exten => _900,2,SayDigits(${EXTEN}) exten => _900,3,MixMonitor........... exten => _900,4,Dial(SIP/${EXTEN}|${DEFRT}) exten => _900,5,Playback(custom/extn-xfer2) exten => _900,6,Goto(custom-michael,901,4) exten => _901,1,Playback(custom/extn-xfer) exten =>

Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Regards -------------- next part

*Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 > Asterisk_IP.5060: UDP, length 345 What i need to know please what TTL means specifically and what is the best value og TTL and what is the lengh vale mean

Hi all, I'm planning to build a VOIP solution for handling SIP calls coming from endpoints registered on a specific SIP proxy...I made some research regarding network architecture and found out that the best solution is to use OpenSips as SIP proxy for registration and local calls between registered endpoints and use asterisk server with a2billing for PSTN calls, rating, routing and all other

Anyone has an idea what happend to voip-info? it stopped working about 24 hours ago. Nir S -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070314/23cdc0f6/attachment.htm

Hello, On Mon, 2 Apr 2018 10:01:56 -0400 m.roth at 5-cent.us wrote: > Turritopsis Dohrnii Teo En Ming wrote: > > Good evening from Singapore! > > > > The foremost question which I want to ask is, what is the universal > > (world wide) understanding behind degaussing hard drives? > > > > I work for No Secrets Agency (NSA) Pte Ltd (fictitious company name