similar to: Debugging issues with setup

I have Sipura 841's talking to a CVS-HEAD of August 4. If I disconnect the power to the Sipura, Asterisk does not hang up the channel. My sip.conf for this phone looks like: ; [super1] ; Sipura 841 disallow = all allow = ulaw callerid = "super1"

Hello, Reading through the Wiki: "Asterisk supports the ability to write dialplan instructions in the Lua programming language. This method can be used as an alternative to or in combination with extensions.conf and/or AEL. PBX lua allows users to use the full power of lua to develop telephony applications using Asterisk" My question is, what is the benefit of using Lua? I recently

Hi All, I have installed astriesk 6 and am able to make calls using sip x-lite. Its working as I expected. Now I want to make call from sipx-lite to PSTN using asterisk. can any please suggest me which Hardware card that I can buy? and use ( pl give me all ths list of cards.... which are good.). 2) what is that I need to do after buying the card to make it talk to the real world PSTN network?

I have a TDM400p with 3 fxs and 1 fxo daughter cards. It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5" hard drive. I'm getting a bunch of clicks and pops on all ports. Has anybody had a similar experience? Did you find a solution? -- Thanks in advance,

Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3234 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090226/a46e68fa/attachment.bin

I'm trying to determine the capacity of my host running Asterisk 11.8.1 on CentOS 6.5. The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet. The primary application will be bridging groups of users using meetme(). I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1 behaving a bit more like a production box -- bridging calls (box2). The call

I have a client that wants to put a phone on their web page for customers to call them via their Asterisk server. ) A keypad is needed to enter credit card details. ) "Speed dial" buttons like "Tech Support," "Sales," etc. are a requirement. Actually, passing the SIP address in the HTTP link would work with a bit of arm twisting. ) Free is preferred, but not a

Hi All, I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service

Please don't top post. > Am 29. Mai 2015 09:42:55 MESZ, schrieb Luca Bertoncello > <lucabert at lucabert.de>: >> Zitat von jg <webaccounts173 at jgoettgens.de>: >>> Yes, it is called "core set verbose 42", the other options is "core >>> set debug 42". Enjoy the show! I know you can specify a level to the verbose application,

After i rebuilt my server i did default install of Asterisk using the steps off freepbx site. i used these steps before without any issues. this time i have to start Asterisk manually every time the server reboots. if i start it by using ./start_asterisk script in the freepbx directory i get this from grep root 3840 0.0 0.0 4480 544 pts/1 S 12:13 0:00 /bin/sh

I want to be able to allow a caller to dial a ddi, system to verify identity etc (this is all done) I then want them to sit listening to music, until an event happens. When this (external) event happens, I want to play a certain file to the caller, using playback (so that they have ff / rw etc), and when finished, go back to the music. 1) I thought of redirecting to an extension that played the

Hello folks, I want to write an AGI script doing this: 1-user call a number. 2-asterisk call the agi script 3-the script dial the peer 4-if the call is answered, let the call up for 1min 5-then the script hangs up the channel. I tried either in php or in java but no success. In java i did this: ////////////// exec("Dial", "IAX2/400"); boolean t=true; while(t){

On Fri, 26 Jun 2015, Dale Noll wrote: > I added a filter to the /etc/rsyslog.conf file > > :syslogtag, contains, "asterisk" stop > > Syslog is still receiving the messages, but is discarding them. Nice to learn a new (to me) feature of rsyslog. What does 'logger show channels' show? -- Thanks in advance,

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 guys, I'm using Asterisk 1.4 and i need a way to hangup all channels. I want to use the teleyapper system for broadcasting call for security reason but i need that all channels are free when a security call is ready to start! I already search in the old post without success. Can anyone help me? Thanks and sorry for my newbie english -------------- next part -------------- An HTML

One of the above is frequently used to front-end Asterisk. I used OpenSER to front-end a farm of Asterisk servers and was very happy with it. The ability to take a box out of service or to route a specific DNIS to a box for testing rocks. Since OpenSER has died (I don't care about the politics/personalities/trademarks), Kamailio and OpenSIPS have risen from the ashes. What are you using?

Created this function on one of my machines today, thought others might find it useful: freesip() { comm -2 <(seq $2 $3) <(cat $1 | grep ^\\[ | sort | uniq | tr -d \[ | tr -d \]) | grep ^[[:digit:]] } On RedHat/CentOS based systems you can create the following file to have the function available on login: /etc/profile.d/freesip.sh # Free SIP extensions freesip() { comm -2 <(seq $2

Does anyone know how to send a text message from Asterisk?

Hay friends, I want to know how many simultaneous call can i do throughout a voip gateway from the internet call to the normal telephony network, because i want to see what implementation do i have to do multiple call from internet to differents telephones. -------------- next part -------------- An HTML attachment was scrubbed... URL:

How many simultaneous conference meetme setups can be supported in the same time on Asterisk, and what are the corresponding server's specs for this. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100414/ba26a927/attachment.htm