similar to: sipura 841 mass provisioning

I have installed the sipTAPI from http://sourceforge.net/projects/siptapi/ when I use user names like joash.herbrink in Asterisk, it is not working when I change the sip username to my internal extension, like 1006, it works fine. Anybody any idea as to why this is? met vriendelijke groet, Joash Herbrink Technical Consultant "Control the flow" De Kahuna groep

Hi Just 2 issues I have with SPA841. 1. I autodial extension 600 then inside an AGI wait for more digits. The digits are transmitted correctly to * but they do not show up on the SPA841 display, only the 600. How do I set the 841 is show the digits after the 600# 2. Is the SPA841 pixel display backlit? Master

Why is using ulaw or alaw an unlikely scenario? I wouldn't use anything but ulaw\alaw. The Bells can compete on price and will if they have to. Where they CAN'T compete is quality. If there were something better than 711, I'd offer that. Well, there is 722, but not many things support it. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ----- Original

I know the subject of faxing has been covered in some detail, but I was wondering if anyone has a hardware configuration similar to ours that has faxes working successfully and would be willing to share any settings/insight. We are unable to fax reliably with a Sipura 2100 connected to Asterisk. We do not route calls over the Internet and our network has very low latency. The Asterisk

Hi all, The problem is on the volume of the voice sent by the SPA-841. I think the echo cancel algorithm sets a limit to the microphone when detects sounds or noise from the earphone. This problem generates an oscillation on the voice volume sent by the phone and even turns it off completely for very little lapses of time making the communication very uncomfortable. I manage three different

Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server -- #Joseph

Hallo! I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out to be unreliable and never shipped. Yesterday I went looking for alternative suppliers and found the Linksys SPA3000 device. It's a different box, but the specs look very similar. Is this the same device? Has anyone used this Linksys SPA3000 successfully with Asterisk? Thanks, Frank

On Thu, 2006-03-02 at 11:42 -0600, Jordan Novak wrote: > Does anyone have a way to do wake calls? > > > > Jordan Novak > > Communications Technician > > Logistics Health Inc. You could use cron and /var/spool/asterisk/outgoing scripts to dial numbers, etc... > Can you elaborate, I am fairly new to Linux and a phone guy to boot. I am looking for a way for the

I am trying to use a the SIP tapi from www.enum.at <http://www.enum.at/> . This works fine from all kinds of applications which support TAPI, like outlook and Dialer Pro. However when making tapi controlled calls, the signaling to and from PSTN seems to fail. I have used the digium hardware ISDN PRI boards, but also a SIP gateway. Both result in a audio message from asterisk

I have a Sipura SPA-841. It's developed a nasty habit. At random times, it likes to dial my cell phone voicemail number and play my messages to anybody who happens to be within earshot. Any clues where to look at what's going on? My voice mail number (extension 220 in my dialplan) is the only number being dialed. When this happens, show channels looks like this: IAX2/NuFone-1

Can someone give me an idea of the processor power I will need for 1 x TDM240 with 2xquad FXO's and 8 sip phones/ATA's on a quite 100Mbit LAN. The machine we have available of hand is a P4 1GHz with 768MB RAM. Tx Marnus van Niekerk -- "Opportunity is missed by most people because it is dressed in overalls and looks like work." Thomas Alva Edison - Inventor of 1093

Guys. I just unpacked on of the new spa841 I orderd and I was changing the ringtone (and listening to the options) when suddently the phone stopped playing back the tones and now the phone doesn't ring, speaker doesn't work and no ringtone play can be heard. Has anybody had this kind of problems?

Hi All; Any one can suggest a TFTP server to be installed in Fedora (same machine that Asterisk is installed) to be used for Cisco IP Phones to download the required firmware and configuration files. Thanks for the help in advance. Regards Bilal

I'm setting up a new AAH 1.0 box to replace my AAH 0.6 box. Though on the new box, I've installed a generic ebay X100P. I don't have my livevoip or voicepulse accounts set up yet on the new box (can both boxes be registered at the same time?). I've set up one IP phone (SPA841) with the new box. I have my SBC POTS line plugged into the fxo card. I set up everything in AMP.

Hi all. I posted this earlier but never got any advice that helped. If anyone knows how to get this going, I'd appreciate some advice. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten =>

Hi Everyone, Does anyone know of any ATAs that can do proxy failover without using SRV. I don't want to rely on dns if at all possible. Basically, I have Asterisk boxes in two different data centers and I need ATAs to be able to uses the server at DC2 if DC1 goes down. The servers are already in a HA setup at each datacenter. I am looking for added protection if one of the datacenters

Hey guys. If my Asterisk box connects to the PSTN using SIP and IP over ethernet and doesn't require any authentication, what sort of a trunk would need to be created? Thanks, -- Nick e: nick.hoffman@altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality

I have installed several call centers in the netherlands with the eyebeam softphone (from the counterpath guys) It is not free, but very stable, and pretty easy to use. It works great with asterisk (specially the presence option, so agents can see whether somebody is actually ready to take a call). In combination with sennheiser headset CC series, I have had no complaints. We also use a tapi

Phil I have very good experience with the vegasteam ATA's devices.(you might also want to look @ sipura ATA's, since vegastream is doing an oem on there boxes) They support modem until v.90 speeds and faxes for g3. They are expensive, and again, work great and configure very easy joash ________________________________ From: asterisk-users-bounces@lists.digium.com

via google, I found the reference regarding digit maps for the Polycom phones: http://lists.digium.com/pipermail/asterisk-users/2005-January/082884.html But I don't see any instruction for prepending digits to the number dialed. Does anyone know how to prepend a digit to the number dialed (from the Polycom side, not Asterisk)? I can do this pretty easily on a Sipura. i.e. Say I want to