search for: skihills

Hello guys, I've one SIP trunk that support multiple DID. Only the trunk is documented in sip.conf (called DID is taken from the sip-header in real time). I would like to limit the number of simultaneous calls on each DID. Is there a way to achieve this ? My understanding is that the SIP configuration parameter "limitonpeers" will limit at the trunk level, right ? Thanks in advance

Hey hey... I'd like to create a new feature code in asterisk so when a user dials... say... *00, it would then call some other extensions and play a sound file to them. So far, this is what I have... [testing-custom] exten => *00,1,Wait(1) exten => *00,2,Playback(beep) exten => *00,3,Playback(beep) exten => *00,4,AGI(festival-script.pl|I will now attempt the call) exten =>

...dialplan... Because we prefer attended transfers, and the user experience seems more modern. So, does anyone know of a way to detect whether a call from a SIP phone is the first step of an attended transfer or an original call? Thanks! -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 197 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20091125/e382d0c0/attachme...

Hello I have configured TDM400P with asterisk . The problem is that when i make a call to server. and while going on it dont detects call hang up. ie i called the Asterisk server and it start playing files that i indicated to do so in extensions.conf i suddenly put down the phone. now the server must detect that phone is hangup but it dont. How can i make server to detect this -- Best Regards

Greetings, I have a fresh asterisk installation. When I install I get all of the config files. What is the best way to get a 'stripped' down system with just the bare config files I would need to do a sip connection? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091014/4e1042b1/attachment.htm

...ng of the member list. We have switched to the "rrmemory" strategy for now, but we've yet to notice what effect that has--and our ideal would be to use "leastrecent" along with the behaviour that Asterisk 1.2 exhibited. Thanks! -- cc -Wall The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 252 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20080317/4a4...

Hello, all. I've come across a nasty problem just as we are ready to put our first system into production. During our final testing, we were plagued with several "invalid extension" or "password incorrect" messages when we knew the information entered was correct. Upon investigation, we saw that DTMF signals were occasionally but not consistently duplicated. We might

Hi everyone: I've searching for a while and haven't found what i need. The thing is that i have a tdm422p with the two fxo ports connected to the pstn. I want my sip users to be able to call other numbers(any number) in the pstn through my zap fxo channels. I have a big number of sip users so as you can imagine there will be congestion when some of them(more than two!!) want to call

Hi all, I have the following in my extensions.conf: exten => s,4,GotoIf($["${CALLERID(number)}" = "8585979857" | "8585970327"]?15:5) The numbers listed above are known spammer numbers. However, when I call from any other CALLERID, it still directs me to s,15 which is the Hangup() application. Here are logs from the asterisk CLI: -- Executing

In "The future of Telephony", it says "... We should also note for security's sake you should always make sure that your [incoming] context never allows outbound dialing. (If by chance it did, people could dial into your system and make outbound toll calls that would be charged to you!) The book was demonstrating using a PSTN environment and the zapata.conf was something like:

Hello Everybody, I have a genuine problem in Asterisk setup. I have three inbound trunks in my asterisk box, everything is working fine but the only problem is when any user make an out- going call through his/her extension it goes with same number labeled on this. Can we set each of these lines to have fixed outgoing numbers like if extn: 201 make an outgoing call the recipient should

Evening, My latest asterisk box is having a difficult problem. It is configured with one TE210P and TDM400P with four FXO modules. I'm running FC6. The TE210P only has a single PRI. When the system boots, it is completely random what order the zaptel modules will get loaded in. Sometimes zttool shows the FXO as the last span, sometimes as the first. When it does load as the first, which

Hi all! I am looking for some software that would work as a proxy between a SIP device (SIP phones and ATA boxes) and the Asterisk system running IAX. The reason is that I can only get IAX trow the firewalls, and can't change the settings. One solution I am using are getting several Asterisk system communicate trow IAX bout in this case would I rater have every persons computer act as a proxy

My company for various reasons has asked that I come up with a way to have previously parked calls be re-parked in the same parking slot. I have looked at setting up asterisk so that the receptionist chooses which slot to place a call, but I think there is an easier way. That is when I came up with the idea of "Sticky Park". Here is how it would work. A call would come in and