similar to: Stripping characters from SIP dial strings

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

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

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 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'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 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.

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

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

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.

> 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?

Are the phones exposed to the internet (even using NAT)? If so there is a good chance these calls are not coming through your PBX but are coming in direct from someone, usually scammers. Polycom has a config option to disable accepting calls from unknown devices. No idea if Cisco has something similar. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at

Hi, In my dialplan I have the following line. same => n,Set(CALLERID(name)=Support) I am expecting this to always set the caller id name to 'Support' - however, we are getting calls come in as "Anonymous" with the number as something like "unknown at unknown" We're using Cisco 7945 phones - I possibly wonder if they are displaying this rather than asterisk