similar to: Selling asterisk-based solutions

Has anyone out there used the PipeMedia/PipeCall PSTN gateway? Anything good/bad to say about it? I'm considering using them for a new customer. They seem to have good rates, good provisioning tools and (better still) give commission on usage to dealers. -- David Gurr Congruity Ltd. Fax: 0871 661 1756 Hemel Hempstead UK

I'm looking for recommendations for UK-based VoIP-PSTN gateways. They should ideally offer: - IAX connection - Multiple simultaneous calls on a single account - Lower call rates than BT Business - Auto-top up or invoicing in arrears I can find several that offer one of these facilities, but none that offer all. Thanks! -- David Gurr Congruity Ltd. Hemel Hempstead, UK

I'm building a phone-in demo system to use for introducing Asterisk to prospective clients. One of the things I'm wary of is their likely preconceptions that VoIP systems will have poor audio quality. As a result, I'd like to ensure that the voice prompts I'm using have the best possible audio quality. Is it possible to use sound files at higher than 8kHz sampling? My callers

What FXO interface methods are folks using successfully in the UK? I'm looking for good, known-to-work solutions for commercial use for two PSTN trunks on an Asterisk box. Here's the options I have, as I see it: i) Two Digium X100Ps. Pro - cheap (c. ?120), CE approved. Con - UK line impedance mismatch, with resulting echo problems, plus needs two PCI slots. ii) Digium TDM400P with two

Now this is really frustrating. Everything was working fine, and now it isn't ... I don't think I've changed anything that would affect this, but I guess you never can be too sure. My setup is as follows: SIP softphone (SJphone) connected to Asterisk running my Linux NAT firewall box. This is all on the internal network. Asterisk then dialing out through various means - SIP to

Can someone confirm what I should expect the correct behaviour to be on incoming calls if I have multiple SIP phones configured for the same username? I'd expect all the phones registered under the username that that extension is associated with to ring, and the first one that answers gets it. What I get, is just the first phone that registered gets a ring. The second one doesn't ring at

I'm trying to get inbound calls from a TNT working but get 407 errors from the TNT. This is what I have in sip.conf: [maxtnt] type=friend host=x.x.x.x dtmfmode=rfc2833 callerid="MaxTNT" <maxtnt> context=demo qualify=yes disallow=all allow=g729 allow=ulaw insecure=very This is what the TNT is spitting out: Jul 24 14:55:12 tnt1 1/17: Releasing

I am using IAX to send a call to my cell phone. I want to be able to hit # and transfer it back into the office. I have added tTr to the dial command and hitting # prompts me for the transfer, but after I start dialing 103, it stops at 1 and tries to transfer it within nufone instead of my dialplan. This is the debug output: -- Called me@NuFone/1515480XXXX -- Call accepted by

I finally got the power supply for my 7960 and am having problems getting it working. What should be in sip.conf and the SIP(macaddr).cnf file? This is what I have in SIP0002FD3BA8F7.cnf # SIP Configuration Generic File # Line 1 appearance line1_name: Asterisk Test # Line 1 Registration Authentication line1_authname: "phone1" # Line 1 Registration Password line1_password:

Am I the only one that has noticed there is no way to wallmount a Grandstream phone? There are screw notches on the back, but no hook to hold the handset in. -- Dave Weis "I believe there are more instances of the abridgment djweis@sjdjweis.com of the freedom of the people by gradual and silent encroachments of those in power than by violent

I'm having problems in dialing numbers over SIP that include characters from the UK international phone number conventions. I have my contacts in Outlook, with the numbers represented as: +<countrycode> (<area code>) <numberpart> <numberpart> eg: +44 (20) 7834 1234 or: +1 (801) 555 1234 I'm using the SJphone softphone, doing my testing through the Stanaphone

I have the need for a slightly odd * configuration for testing purposes. I have a working * setup with SIP softphones, VoIP trunks and a single X100P clone for PSTN access. The PSTN line I'm using for testing is also in use by other folks. For incoming calls, I'd like to set is up so that * functions as a voicemail backstop on this line. This much is working fine. For outgoing, I'd

I'm interested to hear how folks are handling NAT SIP routing issues "in the wild" for commercial use. Are you using a commerical SIP-aware NAT router solution? If so, what? Are you using a software SIP-proxy like SER or siproxd? If so, which? Do you set everything to "canreinvite=no" in sip.conf? Any comments about real-world implementations would be welcome. Thanks

Looking for folks experiences with ISDN BRI cards in the UK ... what's good and what's bad and any gotchas. Thx -- David Gurr Congruity Ltd. Hemel Hempstead UK

Hello, I would like to know if voicetronix card (specially openswitch6 and 12) can be used with asterisk. Is there any driver for this card? Best regards, Daniel -- Daniel ANDRE (mailto:dandre@iris-tech.fr) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com

Hello. I suspect the answer is no, but I'll ask anyway. Commercial phone systems have phones with multiple extension buttons e.g. 20, that can be programmed so that when you press one, it will call the extension. Is there any 'open' phone that can do this with asterisk, does an 'open' phone even exist. Further to that, commercial phone systems have a operators

Hi, I am new to Asterisk and wanted to ask a question concerning PSTN trunking. Is there a way to have DID's sent over IP to a switch? I know if One switch has traditional PSTN like a PRI this can be done, but is there a service provider offering this so I dont have to buy any tradtional PSTN trunking? Thanks, Jim -------------- next part -------------- An HTML attachment was

Does anyone out there use Asterisk with voip(sip or iax) long distance provider? Care to share about your experiances doing this? Michael

Hey all, I'm having problems reliably dialling out my FXO card. About 30% of the time I'll get a "your call cannot be completed as dialed". I'm thinking it might be the dialling speed, but I can't find any configs that change that setting. Any suggestions for troubleshooting? Thanks, Brad Waite

I am new to Asterisk and Digium card implementation issues. My VAR is strongly recommending using Apple hardware and Yellow Dog Linux for my telephony project, because of his familiarity with this OS. Is the PowerPC an appropriate and stable hardware platform for Digium/Asterisk development? Charles Hatchette chatchette@generalcare.com -------------- next part -------------- An HTML attachment