similar to: soft ATA on linux with zaptel?

How Magic Jack can only charge $20 per year? do they have a call limit? do they have a call duration limit or limit of minutes per day?, Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090207/974cf988/attachment.htm

Can Asterisk be configured to hang up if another phone picks up? I'm a bit lost as far as terminology goes, but here's my setup. At home, I have asterisk answering calls from the pstn and sending them through to a sip extension or voicemail. All that is working fine. The box running Asterisk isn't on 24/7 so I have a secondary phone connected to the line as well. If Asterisk is

While I myself am not a MagicJack user, I'm curious as to whether anyone here has managed to heist their MagicJack account's sip credentials, and use them to terminate calls using asterisk. Apparently it's as simple as sniffing the SIP credentials. If so, said person would enjoy unlimited termination for $20 year while retaining the flexibility of setting their CallerID to a

Anyone have much luck with these on ATA's? I have a few sites that use them succesfully with multi-port Audiocodes boxes, but just connected ten machines to Linksys 2102s and they are very flaky. Using u-law on a 100Mb switched network that is barely utilized, then out a T1 on a Sangoma card. Perhaps there is some tuning on the Linksys or the credit card machine itself? Going to look

was breezing over http://voxilla.com/ Looks like a new ATA from the founder of Komodo Technology (aka the Cisco 186) Sipura SPA 2000 http://www.sipura.com/products/spa2000.htm to join the others Cisco ATA 186/188 http://www.cisco.com/warp/public/cc/pd/as/180/186/ 8x8 DTA-310 http://www.8x8.com/products/home_office/dta-310/index.asp.html Grandstream HandyTone 286

I am using asterisk 1.4.10 and seem to be having a problem with blind transfer. This could very well be a pebkac problem but I'm not sure. A call comes in on a Zap channel and answered just find by a context that does a Goto which calls a macro (seems convoluted now that I look at it) to do some CID bookkeeping but that ultimately dials all of the phones interested in calls from the Zap

I was looking for something to connect a couple of POTS handsets to my asterisk server and found this on ebay http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItem&category=162&item=5162868118 &rd=1 The documentation says that it does SIP - therefore will it work in an asterisk environment. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus

Me again... I have service with a company that does not allow for a BYOD plan. They will not give out credential or server info either. Is it possible to run the FXS port of the ATA to an FXO port in *? The service I have is throug Broadvox Direct using the Mediatrix 2102. I have tried this using loop start and kewl start. The * box sees the incoming ring, picks up and starts my dial plan. But

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

> -----Original Message----- > From: Brian J. Murrell [mailto:brian@interlinx.bc.ca] > > - Single Sign-On via Kerberos > > OK. Actually I understood this feature. I am just wondering how it > applies in an MS network. SSO to all of what? If my DCs are my > file/printer server(s) (let's say I mirror the data contents > of my PDC to > my BDC as well --

Currently running two POTS lines into an asterisk system. Analog and SIP on premises. Being in the sticks, the POTS service is abysmal for quality, especially in the rain. Recently, cable has become available with VOIP phone. The cost savings are attractive as it can replace several independent services for TV and internet (currently satellite). But, I cannot get much out of them, regarding

Hi, I'm putting together a fax solution for my company that utilizes T.38. I wanted to throw out my plan and get some feedback if anyone is doing something similar or sees a blatant problem with it. We're currently rolling out SPA-942 phones for the standard desk phone with vanilla Asterisk 1.4.15 (just upgraded it today) on the back end. Most calls for satellite offices are handled by

Hi, I've brought this up in the past and there was a good discussion - am wondering if there have been any new developments. Our dialtone service, like I am sure is true for most ITSPs, touts the ability to drop your POTs lines for significant savings. For businesses we have a low-cost Atom based PBX and a "fax relay" setup locally with hylafax/iaxmodem to solve that issue,

I have running * without any Digium (or any other) hardware. Now I need to connect analog FAX machine to it. I think that cheapest and easiest way is to buy ATA. Please correct me if I'm wrong. Now, which ATA should I buy? Local dealer sells those four. I can buy something else (if there is any reason for it), but I prefer something of this. One more question, can I plug two lines in any of

Hi ;) I'm a french newbie and i installed asterisk 1 day ago. I've got an ATA186 and a computer with Sjphone installed. If i want to call the sjphone from the ata or call the ata from de sjphone everything is ok. My problem is ,that i can't call the voicemail or any other phone number ..as 600 for exemple from the ata or the jphone. I don't know why but i looked after a long

Greetings. I have a scenario I would like some advice on. I have a 100,000 square foot building that we will be moving some work crews into. It has offices on each end of the building and a fiber line between them. I currently have an asterisk 1.2 system in place and about 30 phones. My problem is they want a few fax machines out in the warehouse area where I currently have no wiring for

Hello, I've spent a couple days searching and posted into the forum with no luck, apologies to anyone who reads the Digium forums for the cross-post. I'm having a problem with an asterisk set up where I have a TDM402B connected to a POTS line. Also connected to the POTS line are plain telephones, non SIP, just plain old telephones. When one of the normal handsets goes off-hook,

I have had very bad experiences with IAXYs so far.. I have pulled them and will be attempting a refund shortly. Bad audio, overheating and shutting down until allowed to cool, etc. make it unusable in a business environment. That said, is there a low-mid priced solution for a remote office to connect to a home office runing asterisk? There seems to be a hole in the market for 6-8 person

https://bugs.freedesktop.org/show_bug.cgi?id=47306 Bug #: 47306 Summary: segfault in nouveau_fence_update Classification: Unclassified Product: xorg Version: unspecified Platform: x86 (IA32) OS/Version: Linux (All) Status: NEW Severity: blocker Priority: medium Component: Driver/nouveau

Here is the setup: POTS -> PRI -> Asterisk -> ATA (Fax) The ATA is set to only 711. Asterisk's sip.conf sets this device to only 711. Yet, faxing works less than 50% of the time. I cannot possibly be the only person with this problem. I can't even get app_rxfax to work. It answers but rxfax drops the call 5 seconds into it. If anyone out there has near 100% success with