similar to: trixbox - sip trunk with voipwise

Hello. There are so many sound files in /var/lib/asterisk/en. Is there an easy way to let me play back all of them one by one while I am watching CLI to see the current file name? Thanks for help. -- Jian Gao IT Technician SJ Geophysics Ltd. <http://www.sjgeophysics.com> jian.gao at sjgeophysics.com <mailto:jian.gao at sjgeophysics.com> Tel: (604)582-1100

Hello, all, I have a Linksys 3102 from a VoIP provider. It use SRV record to register to the provider's SIP server. When I configure this line into my Asterisk, the register doesn't work if I use their domain name. So it like this: If I use register => user:pwd at proxy.provider.com then I got: [2010-05-17 11:47:19] WARNING[2366] chan_sip.c: No such host: proxy.provider.com

I have a list of CLIDs prefixes that I want to use in a context. Basically, I want to do this but the list of prefix numbers is much longer. List of prefixes (556,557,557,989.....) [custom-inbound] exten => _556,1,answer exten => _556,n,playback(beep) exten => _557,1,answer exten => _557,n,playback(beep) exten => _558,1,answer exten => _558,n,playback(beep) exten =>

hi, I am quite new on using Samba and sorry maybe ask a silly question here. I set up simple Samba server on Fedora3 using the samba rpm package comes with fedora3( version 3.0.10-1.fc3). I use the SHARE security level to make things easier. Everything goes fine so far, except that for some windows user, some times, on browsing some directories, it takes extremely long time to display the

Hey all. My boss asked me to implement the following When DID 713xxxxxxx is dialed send an email to mmosier at xxx.com. with the time date and CID included in the email. I know how to code some but am looking for the best way to do this. Sorry I might have asked this a couple months back. I forgot. Mmosier Houston Respectfully Michael D Mosier Ftoc Certified -------------- next part

Hi list, playing around with HTB showed that it may introduces pretty much latency for our setup. Docum.org says the following: "The default qdisc added to a htb class is pfifo_fast qdisc. The size of the qdisc is the device queue length and this is 100 packets for an ethernet device. So if you want to have a shorter queue, you have to add a shorter qdisc to the htb qdisc." Can

We are planning to have about 100+ parking lots defined in features.conf , each with about 4 unique park positions. Asterisk will be handling all the parking and unparking (we don't exclusively use Park/ParkedCall in the dialplan): [parkinglot_a] parkpos => 1-4 context=parked [parkinglot_b] parkpos => 5-8 context=parked As far as I can tell, Asterisk adds/removes extensions to the

Today I upgraded my Asterisk to the new 1.4.39.1. One of sip trunk stop working after the upgrade. Here is the sip debug: --------------------------------------------------------------------------- <--- SIP read from 208.65.xxx.xxx:5060 ---> INVITE sip:1778xxxxxxx at 10.11.22.77:5060 SIP/2.0 Via: SIP/2.0/UDP 208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-d9175178645e9146-1---d8754z-;rport Via:

Hey. I found references to my problem 'scp: command not found' in the archives. But I could not find a solution to this problem. Could someone please help me out here? Info: OpenSSH 1.2.3, RedHat 6.1 Thanks! -- Steve

Hi, We heavily use meetme/SLA functionality in Asterisk, and continuously run into issues with dahdi timing. The two errors we get are: ERROR[6518] res_timing_dahdi.c: Failed to configure DAHDI timing fd for 0 sample timer ticks WARNING[22024] app_meetme.c: Unable to write frame to channel Right now, dahdi in our setup uses the software timer (with res_timing_dahdi.so which gives much better

Hi All, i want to make an extension from pbx1 able to tlak to another extension from pbx2 or use pbx2's trunk to dial outside calls. so i edited in both servers accordinally the iax.conf: register => pbx1:pass at 172.16.200.175 <pbx1%3Apass at 172.16.200.175> [pbx2] type=friend host=dynamic trunk=yes sercret=pass context=[default] ; i used the biggest context to avoid confusion as