search for: skirgaila

Hi, I wonder if I may freely use the default soundfiles that came with asterisk (fpm-world-mix, fpm-calm-river and fpm-sunshine) on production server? Are there any official sources of royalty free music? -- Mvh, Aurimas Skirgaila -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101022/37d90a2a/attachment.htm

...heck if the pickup time is equal to the pickup time when the same phone number was previously detected as AM - if the pickup time is different from the last time, - it's HUMAN, else proceed standard AMD(). has anyone done this before,so I wouldn't be reinventing bicycle? -- Mvh, Aurimas Skirgaila -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110516/8017a252/attachment-0001.htm>

hi, We have a phone number from third party provider which is used for inbound calls. How could I monitor if this *phone number* is reachable? the initial idea doesn't sound elegant: - on my SIP server I set couple seconds of ringing before Answer(). - the monitoring server calls to that phone number for few seconds, checks if it "hears" the ringing and hangs up the call. ** I use

...n 0/1 does not have any effect, bchan=1-15,17-31 dchan=16 [zapata.conf] group=1 pridialplan = unknown switchtype=euroisdn context = trunk-1 signalling = pri_net channel => 1-15,17-31 Hardware - Dell PowerEdge R200. Now moved onto barebone test server, but same errors persist. -- Mvh, Aurimas Skirgaila -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100329/ef0f5c5d/attachment.htm

Hi, I have IVR and when I press 1, asterisk calls my mobile phone. If my mobile phone is offline, asterisk transfers to asterisk voicemail. I'd like asterisk detects my mobile voicemail and if my mobile voicemail answers, asterisk transfers to asterisk voicemail. For that, I used AMD. So I have problems ! Asterisk detects answering machine everytime! How do I do please ? extensions.conf

Hi, I wonder what is the difference between Transfer and Dial applications? Could somebody give me an example of Transfer usage? (documentation and voip-info looks poor a bit). I'm using Asterisk 1.2.5 if it matters. -- Mvh, Aurimas Skirgaila -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090508/fa2972e2/attachment.htm

Hi All, Is it possible to make a IAX2 connection between asterisk 1.6.1.0 , and asterisk 1.2.14 ? i tried to use a IAX2 connection between version 1.2.14 and 1.6.1.0 but it gave an error - 1.2.14 End - Error Msg WARNING[8313]: chan_iax2.c:7103 socket_read: Call rejected by 147.120.203.71: No authority found 1.2 END , IAX.conf [trunk14] type=friend host=147.120.203.71 secret=test123

Hello All, I do have asterisk installed for a call centre with aheeva application and i would like to know how to configure the sound for the inbound calls and if there is any possibility for agent to receive a file with the phone number and name of clients: For your information there is no problem related to the outbound call An help would be appreciated Kind Regards Salah. --------------

Hello , I would like to tweak my Answeing Machine Detection (AMD) in Asterisk. My current values are AMD(2500|1500|300|5000|120|50|5|256) and we were able to identify approx 25-30 % of all answering machines. Anybody have any suggestion to improve the accuracy of AMD. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, I need a hand to find the best dialplan failover solution when using two SIP Trunks. My reasons to do failover are: a) one of the two providers could be unreachable b) both providers may be UP but one of them could return a SIP error message (maybe caused by DOWN E1s) Googling I found a few possible solutions: 1.

We need to monitorate the E1 with nagios, somebody did this? any ideia? Thanks in advance! -- Atenciosamente, ------------------------------------------------------- Dario Quiroz (71) 9275-9080 gtalk: darioquiroz at gmail.com ------------------------------------------------------- -------------- next part -------------- An HTML attachment was

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