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2007 Aug 24
3
which OS would be fine for asterisk
Dear which Linux version would be fine for asterisk CentOS 5.0 or Debian 4.0 or RHEL 4.0 Regards Satish patel --------------------------------- Choose the right car based on your needs. Check out Yahoo! Autos new Car Finder tool. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2008 Jan 29
2
Dialogic card
Hi list, Anyone knows where I can get information about configuring a Dialogic card to run with Asterisk?? The model I have is D/120JCT-LS. Somebody told me that I had to buy the driver, but I don't know if this is true and if so, who, how and how much...
2008 Sep 08
0
Streaming live music into a conference room
...default, where can i find this module ? and how do I install it, also is there an easier way to set up streaming live audio into a conference room ? that this http://nooss.org/wiki/Audio_Streaming_from_an_Asterisk_Meetme_Room :-( Thanks guys On Sun, Sep 7, 2008 at 3:34 PM, Edgar Guadamuz <eguadamuz at gmail.com> wrote: > Hello, > > I have been testing a trunk IAX and another SIP, using sipp to > generate SIP calls to a Asterisk box. > > > The testing dialplan just connects to another Asterisk box, who > answers the call and playback some files. > > I noticed...
2008 Feb 10
3
usability Testing Costa Rica, SanJose asterisk PBX / dsl/cable service
Can anyone that can & is willing to do some IAX usability testing from Costa Rica, San Jose to my Asterisk PBX please pmail your contact info thx tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080210/b0b0d6e3/attachment.htm
2007 Aug 23
0
asterisk-users Digest, Vol 37, Issue 88
...-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070822/347dae 77/attachment-0001.htm ------------------------------ Message: 9 Date: Wed, 22 Aug 2007 01:22:13 -0600 From: "Edgar Guadamuz" <eguadamuz at gmail.com> Subject: [asterisk-users] How to re-read values from database in Trixbox To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <ffbf31780708220022p61011a07lf5ae62135774f2e5 at mail.gmail.com> Content-...
2007 Apr 12
3
Sharing trunks between asterisk machines
Hello eveybody, I've been looking for a way to share trunks between two asterisk servers. I guest I have to use Dundi, but I've not found the exact method yet. I need a way to allow users registered in one server to use the another server's trunks in the case the first server's trunks were busy and vice versa. Is this possible? Thank you so much, any comment will be useful.
2007 Aug 22
2
How to re-read values from database in Trixbox
Hello guys, I'm using Trixbox and I have a PHP application that updates a value in the MySQL asterisk database as an interface to have a dynamic customizable IVR. After execute the UPDATE SQL query, the php application is supossed to reload asterisk or restart amportal in order to get the change working, but nor asterisk -rx reload nor amportal restart got the change working. So, the
2008 Oct 11
1
Asterisk 1.6.1 + openais
Hello, I followed the steps by Russell* http://www.venturevoip.com/news.php?rssid=1980* and I got it working for publish_event only. As soon as I add subscribe_event, Asterisk doesn't start and I just get the following message: *Oct 11 6:38:04.340485 [CLM ] nodeget: trying to find node ffffffff* I have no idea what's wrong. There is not very much information about this issue. Where
2008 Nov 03
1
asterisk and bigmem kernel
Hi all, I installed asterisk 1.4.22 on a Dell poweredge 2950, with 4GB RAM. I used debian, but the default kernel doesn't recognize the 4GB, just 3, so I installled the linux-image-2.6.18-6-686-bigmem kernel, that do recognize the whole 4GB. Asterisk seems to be installed correctly, but I had two issues: (1) I had an error with zaptel. Asterisk didn't start with zaptel modules loaded. I
2007 Jul 08
2
asterisk is not sip proxy
Hello Asteriskers, I'm confused about why Asterisk is not a SIP proxy and why exactly this can affect the performance of a large Asterisk system. I know that Asterisk acts as a useragent endpoint, but my doubt is why exactly Asterisk could overload the call flow if the RTP voice stream goes from the caller to the called party. Does someone know how many calls or pencentaje that could handle