similar to: Multiple SIP phones ringing for same extension

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

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 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

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 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 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

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

Possibly slightly off topic, has anyone ever had Cisco 79xx Series phones come up with "cannot complete conference" errors when trying to conference two calls together? This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may monitor email traffic data and the content of email for the purposes

I have a Windows server 2003 Domain controller with a share called DC02Data. I can connect to this share successfully from several RedHat boxes, running samba 2.2.7. However, on several other RedHat machines, running samba 3.0.9 I cannot connect properly. In fact, I actually seem able to establish a mount successfully, but cannot then view the files, as shown below: [root@ixapp01 mnt]#

Dear Sirs I would like to make you aware of the following training taking place in Chicago next month. Analysis of experiments using ASReml-R, Chicago, 24/25th April This workshop is aimed at scientist/practitioners that are interested in analyzing complex datasets by fitting linear mixed models, particularly users with experience in breeding programs and design and analysis of experiments.

> Apparently this is a known problem past v8 firmware: > http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update- > version-9/ I've done some more playing about and what I've noticed is that even when using TCP SIP on the 8.x Firmware conferencing doesn?t work - making it use UDP fixes this. So has anyone managed to get the 9.x firmware working with UDP?

We were using G722 - I thought similarly and tried a call with alaw. Same problem occurred, any other ideas? > I'm willing to bet you are forcing the use of G729. 7940 and 7960 phones can > only do a single G729 channel, and if you require G729 for the second leg of a > conference, it will fail. This message may be private and confidential. If you have received this message in

Hi folks, I'm trying to set up shared folders via symlinking and have come across a problem. I created a folder for one user, then symlinked it to another. I figured that one thing that is likely to happen at some point is that user 2 is going to decide they don't want to look at that folder any more, and will delete it, so I tried this. Much to my relief, it didn't delete the actual

Hello, I'm having a problem with a few Polycom SoundStation 6000s. Everything works fine, but they drop registration to asterisk after about maybe 30 minutes - the phone does not re-try to register and if you try to dial out on the phone it says "URI Dialing is Disabled" Has anyone else had this issue? I'm running asterisk 11.7.0. This message may be private and confidential.

I'm willing to bet you are forcing the use of G729. 7940 and 7960 phones can only do a single G729 channel, and if you require G729 for the second leg of a conference, it will fail. On Tue, Jan 20, 2015 at 10:03 AM, Jordan Cook - Gyron Networks < jordan.cook at gyron.net> wrote: > Possibly slightly off topic, has anyone ever had Cisco 79xx Series > phones come up with ?cannot