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Good day all What is the amount of calls that asterisk can handle,SIP and from/to PSTN -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301

Can i register a cisco 7902G with asterisk????? what i have to do??? thanks in advance. R -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041130/12bf3c06/attachment.htm

Good day all I got my Music on hold to work but can I/how do i get music on transfer? Please help Thanks

Hi, I wonder if you guys can recomend a good SIP phone. A phone thats works great with * has a lot of features, and is cheap. Actually all kind pf VoIP hardware is of interesst. Is there a really good site for VoIP harware ? /Mike

...ged and confidential information intended only for the use of the addressee named above. If you are not the intended recipient of this message, you are hereby notified that you must not disseminate, copy or take any action in reliance on it. If you have received this message in error, please notify Stormcorp Network Solutions, its subsidiaries or associates, immediately. Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the view of Stormcorp, its subsidiaries or associates.associates.

Hello, it is possible to get MWI working with Grandstream and Asterisk? Thanks. -David

PH> I am setting up a snom 360, and the lights come on OK when the mapped PH> user makes an outgoing call, but when the user takes an incoming call PH> the light does not come on. PH> I do not want to install the bristuff patch if possible. PH> (although I can see that with the devstate command I can make the lights PH> do whatever I want) First, ensure that the 360 has

Hi I am having a problem with voice quality getting bad after about 10-15 minutes on a call. The call starts out fine but gets very chopy after 10-15 minutes. It seems like a buffer problem, I am using Asterisk with a Voicetronix Openlin4 card. Could someone shed some light on this?

I noticed I get echo on line when my internet connection is congested (during file downloads, for instance). Can you verify this isn't your problem? > -----Original Message----- > From: Doug Reid - Stormcorp [mailto:doug@stormcorp.co.za] > Sent: Tuesday, December 14, 2004 3:06 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Should echo cancellation be a "science" > oran"art"? > > Use ISDN lines! > > -----Orig...

Is H323 built into the current CVS? If so, could someone give me an idea of a simple config? thanks, darran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030429/fffec1d1/attachment.htm

Hi All I have a Junghanns BRI 4 port installed where only the first channel of each line is working i.e. channels 1 and 4 work but 2 and 5 don't. Our config is the same on this box as 15 other similar installations where all works well. the only error I see is in /var/log/messages: Jun 28 15:49:31 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jun 28 15:51:27

Are there any IAX speaking "hardphones" out there? If so, can anyone offer comment on their quality? Thanks! -Dorn

...nes for Asterisk - is there a list? > (Tomasz Chmielewski) > 10. Re: Advantage of IAX2 to SIP? (Rich Adamson) > 11. Asterisk Call Monitor and soxmix error (Craig Waddington) > 12. RE: Avoided deadlock (Brian West) > 13. RE: CallerID on X100P in South Africa (Doug Reid - Stormcorp) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 1 Dec 2004 08:23:49 -0600 > From: "Roger Hanson" <roger@makarios.us> > Subject: Re: [Asterisk-Users] software phones for Asterisk - is there > a list?...

Dear All, I've got a working asterisk installation which I need minor help from. Currently, I'm running a Sales Queue, which is answered by a selected group of people. Here are my queues.conf [sales-hotline] strategy = roundrobin timeout = 10 member = SIP/602 member = SIP/603 member = SIP/701 member = SIP/604 After calls come in, it works fine, however, I notice that even when SIP/602

Good day all So I've installed asterisk with my openline4 card and I've setup sip and I can do sip on the local network,we are using soft clients,x-lite. But... When a call comes in from the outside(PSTN) and the you dial the extension it forwards the call the the client and you see incoming call on x-lite,you accept he call....BUT there is no sound.It shows there is a call and you are

Good day all I have figured out most/basics of asterisk.I went with sip and made my own sip.conf and extensions.conf No I have 2 servers running sip in different towns.Both is connected with static ip so thats fine,but now. Lets say someone want to call someone else in the other town.How do I get asterisk to know,for instance sip extension 101 is on another sip server on a different ip. And I

Good day all IS there a way to personalise the voicemail message when you leave a message? Thanks Altus

Good day all Is there anyone who has experience with ISDN BRI&DDI? I want to know if asterisk can distinguish between the different numbers? I want each number to play a different intro/answering message? Please Help Thanks Altus

Goo day all I'm going to put a asterisk server running sip in at a client.The server is going to have a public ip so that it can talk to another server. My question is how do I secure asterisk/sip. I got a firewall only allowing tcp/udp 5060? I got sip to work with md5 What more? Please Advice Thanks a million

Hi All We have Grandstream 102's running ver X.18. When hanging up after a call has been made the grandstream seems not to disconnect the call and when you put the handset down the phone rings only to pick it up and be on the same call. This is happening quite often and gets very irritating. Can anyone help with this? Regards Doug