similar to: Sound file quality

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

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

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

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

I have no analog line interfaces on my asterisk system, but I do have two UK 0870 numbers routed to two separate VoIP accounts (one with FWD, one with gossiptel). Asterisk is configured to register with these accounts. I get voice calls through just fine this way. I thought I could get one of these 0870 numbers to route through to rxfax, thus allowing folks to fax me directly. I've set up

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 curious as to folks experiences in selling asterisk-based solutions. In sales-speak, what are the common "compelling reasons to buy"? I can think of the following potential ones, but I'm keen to find out what seems to work in practise: - Customer wants to cut cost of calls, implements * and signs up to a VoIP/PSTN gateway - Customer wants a new PBX but doesn't want to

I'm using the SJphone softphone, and I've got a nice little SIP-only setup, using (amongst others) stanaphone, VOIPtalk and FWD. But I'd like to be able to use my SJphones to dial directly to folks who provide a SIP URI, eg: 100@calluk.com, without either having to switch profiles in SJphone (to direct SIP) or having to define calluk.com (in this example) as a destination in

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

I found some postings from Google (notably from Mark Spencer) about successful integration of a legacy Toshiba Strata system and Asterisk. I am also facing that current dilemma. The general legacy solutions that I can come up with is very easy -- either making Asterisk a "proxy" (or frontdoor) to the Toshiba system, or have it operate behind the Toshiba via regular extensions. I'm

Hello, I am running error rate analysis. It is my results below. When I compare aov1 and aov2, X square = 4.05, p = 0.044, which indicates that adding the factor "Congruity" improved the fitting of model. However, the following Z value is less than 1 and p value for Z is 1, which means that "Congruity" is not significant at all. Therefore, these two parts are not consistent,

On 9/7/20 4:48 PM, Nicolai Hähnle wrote: > Hi Johannes, > > > On Mon, Sep 7, 2020 at 11:17 PM Johannes Doerfert > <johannesdoerfert at gmail.com> wrote: >> >> > As a separate comment, I don't find the reference to the C++ spec in >> >> > https://reviews.llvm.org/D86233 to be informative enough. Whenever >> >> > that

> The cost of down-sampling, if done efficiently, is probably less then > the cost difference between 32 kHz and 44.1 kHz so it's probably worth > it. If you don't care about standard sampling rate, you could even to a > 2/3 conversion which would get you 29.4 kHz... I'm curious why not just sample at a lower rate if it's just VoIP anyway? My opinion is that 44kHz

Dear All, Is there anyone who is having FastAGI support for C & C++? We do have FastAGI working for the JAVA and rest of the language / script. But I am unable to find FastAGI for C/C++. Please let us know how to write FastAGI using C/C++. Thanks in Advance, Kashyap -------------- next part -------------- An HTML attachment was scrubbed... URL:

https://bugzilla.mindrot.org/show_bug.cgi?id=2695 Bug ID: 2695 Summary: inconsistent outout of "ssh.add -l" between ed25519 and rsa keys Product: Portable OpenSSH Version: 7.3p1 Hardware: Other OS: Linux Status: NEW Severity: minor Priority: P5 Component:

On Fri, Apr 24, 2015 at 9:59 AM, G?k?en Eraslan <gokcen.eraslan at gmail.com> wrote: [...] > > But "Watch" only notifies when there are new pull requests and issues, > which doesn't make sense for the r-source repository. Following Github Atom > feed[1] sounds better, however the feed only provides commit messages not > the diffs. > Right, sorry, I

Hey, I've compiled DarkIce 0.6 dynamically linked to LAME 3.89. I'm running Slackware 8 and using gcc 2.95.3. Running DarkIce yields the following output: DarkIce 0.5 live audio streamer, http://darkice.sourceforge.net Copyright (c) 2000-2001, Tyrell Hungary, http://tyrell.hu Using config file: darkice.cfg Using POSIX real-time scheduling, priority 98 LAME version 3.89 (beta