similar to: I need the least expensive way to do this

We have several people in our church that recently became disabled. I am thinking of setting up an asterisk server and several phone lined so that they can call in to church during services to listen to the service. The phone lines at church are also used by our private school during the week. So I am thinking that I need to setup a schedule so that the lines used for the service will

Hello, I'm planning to convert my phone system to Asterisk, as I've outgrown my TalkSwitch system. I have a few questions for experienced * users, most of which can be answered yes/no. Current Setup: - Talkswitch 48NLS (4CO/8Ext) phone system. - One CO line, two Vonage lines, one Voicepulse line connected to phone system - A third Vonage line directly connected to a fax machine - A

I've been running with a generic X100P for 5 or so months and every once in a while I have problem receiving faxes. I see that others have the same problems and some worse than I have with these boards so I was wondering if using a Sipura SPA-3000 would be any more reliable. Has anyone had enough experience to tell me if that would definitely fix the random fax error. PS. I have * at

Is there a database of MAC address prefixes used the common VoIP devices. I see the Linksys Sipura devices state with 00:0E. Does the same apply to other Linksys VoIP equipment? Is there some way VoIP equipment allow themselves to be identified by requesting data from some ports?

Quoting Tim Litwiller > to connect to the speaker system I either need to trigger a ring on a > analog line to the phone interface on our speaker system, it picks up > on > the first ring, or we can manully push a button that picks up the > line. > If we do the second we would have to have something in asterisk > connect > it to the conference when it picks up. We just

works with Asterisk. I'm thinking I'd need 2 to support 6-8 lines - Or suggest some other equipment that will provide up to 8 fxo ports and connect to asterisk. for future projects I'd also like something with 2 fxo ports and 4 - 5 fxs ports - I suppose a digium card would do fine for 2 fxo and 2fxs and I could do a sipura 2002 for 2 more.

Should a Linksys Sipura 2102 be configured with nat=yes even if it is on the local network? I have been having some troubles with a Linksys Sipura 2100 series, which suffers from NO AUDIO after a few calls.. Because it is on the same subnet as Asterisk it is configured with nat=no. When you think of it because the Sipura 2100 is a broadband router, the voice part may be considered as being behind

Hello Everyone, I can see that a few people are interested in SIP WIFI phones. I have tested several Linksys 300,and it is OK. More of a toy then a business tool. It a poor built in ear speaker, which makes all calls sound tinny, and the unit is known to hang. I have two Linksys 300's that are fun to play with however, I wont hand them out to users. HOWEVER- The Zultys WIP 2 is an

I have had the opportunity to test many IP phones in the last 6 months and I thought you might enjoy a quick review of what I have found. Grandstream Budgtone 200 - Poor Quality for business use- Looks good, and the handset feels nice, buttons have a decent feel, but the disply is difficult to read when you are not directly over head of the phone, plus the sound quality of the handset, and

does anyone get a hold of the SPA-941 Provisioning Guide? i tried call Sipura's tech support, seems like none of them heard of the term "remote provisioning". they kept refering me to their web site which i've check thoroughly, and could not find any documentations on the SPA-941. finally they gave me a phone number to call, which appears to be a fax machine. that's when i

Has anyone got any information on bulk provisioning of Linksys SPA-941/94s? There is an overview in the admin guide but it refers to a different provisioning guide that I haven't found anywhere. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com

I'm experiencing a problem with a Sipura SPA-941 not available for incoming calls after Asterisk & Freepbx upgrade. I can dial out with the phone gto any other internal or external ext. It is registered with the Asterisk server. When I dial the Sipura directly from any other extension, it goes directly to vm. I have other Sip softphones that are working fine. A sip debug when calling the

Apologies in advance as this is not directly Asterisk related, however I thought I might be able to leverage the experience of particiapants on this listserv for some advice. I have a client who is utilizing Talkswith PBX appliances, which have no native call monitoring/call recording capabilities. They are looking for an additional application, service or appliance that can sit on the LAN, and

https://bugzilla.netfilter.org/show_bug.cgi?id=1475 Bug ID: 1475 Summary: Array of addresses wrongly processed Product: nftables Version: unspecified Hardware: x86_64 OS: All Status: NEW Severity: major Priority: P5 Component: nft Assignee: pablo at netfilter.org

I have a customer looking for an automated way to provide his customers information. He found some windows software called Active Call Center - but I believe that he did the 40 day trial and it crashed his windows machine to much. So he wants something that can do a similar task running on linux. The Users expierience should go something like this One of his customers calls in the IVR asks

I'm wondering if the page application is broken in 1.2.5 The following: exten => 2001,1,Page(SIP/3254105) does strange stuff. The caller's phone immediately drops into the call, while the callee's phone is still ringing. I'd think it was a SIP messaging issue, except that the Dial() command is working fine, which makes me wonder if it's a bug in the Page appplication.

I've a small system that I use to support a number of churches. I provide web and email for them. My current server is running CentOS 6.3 with paired 1TB drives in a RAID1 configuration. It works well. One filesystem is very large, >500GB, and contains numerous large files: SQL, docs, church libraries in ebook and digital form, plus stored videos of church services. My problem is that

I have been asked by my church to recommend a VoIP system which can do the following. They do internet radio shows which are sometimes broadcast on radio. They are looking for a system which does the following for about 5 agents, exactly as they have described it. 1. Take incoming calls 2. Put them on hold if there is no one to handle the call immediately, or transfer them to an available

This is a simple question, but I'm going on the supposition that the only stupid question is the one not asked. 1. I have many sets of 5 proportions that are different from each other (prop.test), and want to know which proportions are different from each other. In other words, I want the equivalent of Tukey's HSD test, but for proportions rather than anova. Here is one of the

*@home has this for it's incoming fax macro --- start snip --- [ext-fax] exten => in_fax,1,GotoIf($[${FAX_RX} = system]?2:analog_fax,1) exten => in_fax,2,Macro(faxreceive) exten => in_fax,3,system(tiff2ps -2eaz -w 8.5 -h 11 ${FAXFILE} | ps2pdf ${FAXFILE}.pdf) exten => in_fax,4,system(mime-construct --to ${EMAILADDR} --subject "Fax from ${CALLERIDNUM}