Re: [asterisk-biz] SIP to PSTN Hardware

When we talk about VoIP (I mean pure VoIP) there is no need for Digium cards. All the major and minor players support SIP. I have never worked with any card. So cant say much about them. -Jai www.bingotelecom.com From: Arya <[EMAIL PROTECTED] > Subject: Re: [asterisk

[asterisk-biz] Polycom, Digium, APC, Dell - Discounted For Sale!

DISCOUNTED FOR SALE: Polycom, Digium, APC, Dell We have the following items for sale, heavily discounted to move quickly. Almost everything is in new or almost new condition. Feel free to submit offers. Need to move quickly. Prices are listed each. Please contact me off list at the email or num

Re: [asterisk-biz] Checking Carrier Reliability?

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Alex Balashov wrote: > On Fri, 3 Aug 2007, Douglas Garstang wrote: > >> Ok, so I haven't spoken to them yet, but it looks like their product >> doesn't allow you to store QoS information realtime with an API to act >> upon it. Pretty graphs etc aren

Re: [asterisk-biz] SIP to PSTN Hardware

Andres Paglayan wrote: > There's some study case about Houston university using Asterisk, > and some success histories about using both, Asterisk (for the services) > and openser for the sip registrations, Anyone running Enswitch for a start. If they're running a 6 machine cluster or larger, they

Re: [asterisk-biz] SIP to PSTN Hardware

Thank you for that great explanation Jal So I would have to use Digital PSTN lines for large installations right? I have to see if Digital is available in the area. If I would have to use 2 Asterisk boxes would I need 2 digital PSTN lines? On 8/5/07, Jai Rangi <[EMAIL PROTECTED]> wrote: > > Arya

Re: [asterisk-biz] SIP to PSTN Hardware

Arya, This can be done very well with asterisk and ser/openser. Ser is very good in handling/distributing the load to asterisk servers. The whole system can be scalable. SEE developers claims that it can handle of thousand's of call at the same time. I have never tested 1000 calls. But I believ

Re: [asterisk-biz] Calls Hang up after 1 min 14 sec : SIPINTERCONNECT

The call limits are set at 90 minutes and there is more than enough funds for the account. >From: "Jaswinder Singh" <[EMAIL PROTECTED]> >Reply-To: Commercial and Business-Oriented Asterisk >Discussion >To: "Commercial and Business-Oriented Asterisk >Discussion" >Subject: Re: [asterisk-biz] Cal

Re: [asterisk-biz] Professional IVR recording.

My sister-in-law is from Peru (her whole family depending if you want a male or female). I am sure any one of them would do it for a low price. Email me off-list if you want to hear a demo or two. Thanks, Steve Totaro Carlos Rojas wrote: > Hello > I'm from Lima peru, i'm looking for sounds wi

Re: [asterisk-biz] Professional IVR recording.

Hello I'm from Lima peru, i'm looking for sounds with latin american accent On 8/4/07, Mark Phillips <[EMAIL PROTECTED]> wrote: > > What language or accent do you want? > > Alison Smith (the default voice) is available for American/Canadian > accent as are others for other accents. > > > On Sat,

Re: [asterisk-biz] Canadian E-911 Providers

We had great luck with a company called Northern 911. http://www.northern911.com/911.htm Clay S Perreault http://www.phaseglobal.com Ivan Kovacevic wrote: > > Hi Everyone, > > > > Can anyone recommend a reliable E-911 provider in Canada? > > > > Thanks, > > > > Ivan > > > > -

Re: [asterisk-biz] Professional IVR recording.

I always thought James Earl Jones would be a much better voice for Asterisk. Rehan Allah Wala wrote: > http://www.theivrvoice.com/ > > > Date sent: Sat, 4 Aug 2007 19:44:09 -0400 (EDT) > From: Alex Balashov <[EMAIL PROTECTED]> > To:

Re: [asterisk-biz] SIP to PSTN Hardware

Alex Balashov wrote: > On Sun, 5 Aug 2007, Arya wrote: > > >> lets say if you want to start with 5 concurrent calls than grow to as >> 200 or more concurrent calls >> > >A few Asterisk boxes supported by a proxy should be able to handle > that in terms of transcoding and call volume al

Re: [asterisk-biz] Calls Hang up after 1 min 14 sec : SIP INTERCONNECT

On 05/08/07, Jaswinder Singh <[EMAIL PROTECTED]> wrote: > A2billing limits calltime as per available balance . This is definitely you > a2billing configuration problem . Post some cli output it will show > parameters of dial command . ... and do it on the -users list for maximum assistance. --

Re: [asterisk-biz] Calls Hang up after 1 min 14 sec : SIP INTERCONNECT

A2billing limits calltime as per available balance . This is definitely you a2billing configuration problem . Post some cli output it will show parameters of dial command . On 05/08/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > Hi Members, >I am setting up termination fr

Re: [asterisk-biz] SIP to PSTN Hardware

I know cases when some companies uses Asterisk with 16 PRIs on Sangoma AFT-108D ( 8 PRIs per card ). Due to the fact that Sangoma cards doing lots of heavy calculations on board and uses only one IRQ to poll all of the ports and you have 4 slots on your PC so you can handle 32 PRIs at once or ev

Re: [asterisk-biz] SIP to PSTN Hardware

Maybee the RedFone gear might be appropriate. http://www.red-fone.com/ /Danny From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Arya Sent: den 4 augusti 2007 02:53 To: Commercial and Business-Oriented Asterisk Discussion Subject: [asterisk-biz] SIP to PSTN Hardware Hello