similar to: Can Asterisk do this ?

Hello all, We are looking for some hardware requirements/recommendations to be able to handle a full DS3's worth of TDM -> VoIP traffic. The DS3 would bring 24 calls per T1 x 28 T1s = 672 simultaneous calls. We would then need to convert those calls into G729 SIP VoIP calls to send to our asterisk box over ethernet. Since everything is going in/out of asterisk is 729, and no features

Hello, I've been planning to replace my aging CENTREX switch with a new PBX and am seriously considering Asterisk as my solution. I work at a college, and we currently support just under 300 regular analog lines to the offices and whatnot. I was wondering.. Is asterisk ready for such a job? Would I be making a mistake to deploy this across the campus at this time? It seems that

I was debugging a turnup with Global Crossing the other day and they presented me with a web page that displayed a 'ladder diagram' of a call including a ton of detail all neatly organized in tabs and links so you could drill down to any level of detail needed. The copyright notice says 'Copyright? 2008 Empirix.' Is there any free software available to analyze a pcap or

Good Afternoon, I have an Asterisk box that is acting like it is passing "core show" before every command I type. For example, if I type sip, I will get "No such command 'sip' (type 'core show help sip' for other possible commands). Any ideas? -- -jayson

Hi all, I am handed a project to setup *. The requirement is that it can handle 8 T1s. Half of the calls coming into the system will be routed to SIP extensions (with transcoding). The machine we have in our disposal is a new dual Xeon 3.2gHz server with 2g of ram and an dual 1000mb nic. Voice will be coming in from the PSTN (through 2 quad digium cards) in g711ulaw, and most of the time will

Anybody here having trouble receiving email from the list on Gmail? I havn't received anything since Friday July 29.

Hello - I am looking at connecting 7 - 10 locations together using Asterisk and possibly some VoIP gateway appliances. I need to insure best voice quality as these trunks will be used primarily for customer calls. I am considering implementing a full T1 frame relay circuit to each location which can be done for a reasonable cost. DSL and Cable are currently at each location and setup for

Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050717/311c56ec/attachment.htm

I have compiled zaptel on Mandrake following everything I have always done on Fedora. It is 2.6 udev so... I had to modify the 01-devfs.rules Make linux26 Make Make install... Everything appears to compile correctly but it says module not found when doing "modprobe zaptel" Is this a rights issue? Jordan Novak -------------- next part -------------- An HTML attachment was

I'm trying to setup asterisk to accept call from Teliax, request the 10 digit number from user, then dial it thru the VoIPJet. If I'm not wrong I will be charged by both providers because both connection is active during conversation. So my question is can I set the things so that I pay only to VoIPJet? Specific configuration snippets will be greatly appeciated. Thank you.

Please help me test my setup by dialing 800.564.0215 and listen to the queue for a bit. I have a quad port T1 with NFAS setup. I can dial-out but I cannot dial any 800 numbers (Global Crossing says I need LDS service and that will be a couple weeks) so I cant test it myself. I need at least 24 callers to feel comfortable enough that it is working properly. Thanks, Steve Totaro

I''m writing a quiz application and I need to validate the radio button reponses. I do not have the questions in a database, I just have forms which contain "question1" and then I strip out the "question" portion and just insert the number as the question''s id. So now I''m faced with the dilemma of making sure the radio buttons are pressed. Radio

Hi gang, We are moving our 1.4 asterisk with DAHDI over to 10.0 with SIP. Everything is going nicely except that I can't get NV_FAXDETECT to compile properly into 10.0. Because of this, I will have to have my receptionist manually transfer incoming faxes. Any suggestions? Thanks in Advance Danny Nicholas -------------- next part -------------- An HTML attachment

Hello, I have Asterisk 1.6.2.9-2, the directive "sendrpid" does not work! I placed this in my peer: (sip.conf) sendrpid=yes trustrpid=yes or sendrpid=yes trustrpid=no (and restarted Asterisk) and the line "Remote-Party-ID" does not appear in my sip debug! Please help me, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

We have written some monitoring and stat collection scripts that use asterisk -rx "command" The script runs once a min and logs data and posts any critical notifications. Everything is working well with this method but we get the -- Remote UNIX connection / disconnect message once a min and we would like to suppress it. Is it possible without reducing the verbose logging level.

Hi, I am investigating getting a wifi VoIP phone because its may be a better option than an ATA and a cordless phone.. Does anyone have any experience with the whats out there?? Do they support things like WPA etc?? I have heard the battery life can be a problem.. Is this the case? Thanks..

Can anyone recommend any reading to help me get a better understand of VOIP technology and all the terms used in this discussion thread? I've read many documents on the wiki but I find it a bit lacking. Sincerely, Tyler

I want to know if I buy a Tyan Transport GX28 (B2881) will it work with a TDM400 card? As the expansion slots are only (2) 64-bit 133/100MHz PCI-X. I never tried PCI 2.2 compliant card in a PCI-X slot so I don't know if it can even fit in the slot and if it does will it be seen? Thanks Martin Roy

Off topic but an interesting article http://blogs.zdnet.com/BTL/index.php?p=1306 Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050603/6e1373e7/attachment.htm

Hello there, I'm a new Asterisk user and I wonder if it is possible to associate a voicemail box with a group of users, i.e., a single recorded message is sent to everyone in that group. If so, where can I find more information about that? Thanks in advance, Leo Burd