RE: [asterisk-users] unsubscribe
Disclaimer at the bottom still looks ridiculous even in Spanish... LOL Wiley E. Siler Director of Information Technology 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] www.education2020.com http://www.education2020.com/ Helping students on a mission. Graduation and beyond. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, May 18, 2007 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] unsubscribe Cristian López F. Integración y Tecnología - Terra Chile Phone: (56 2) 330 6966 movil: 56-92401759 E-mail: [EMAIL PROTECTED] Este correo y su contenido solamente interesan a las personas autorizadas de TERRA NETWORKS CHILE. Si usted fue receptor de este correo por error, por favor no lo tome en cuenta y avise al remitente. This message is solely of the interest of TERRA NETWORKS CHILE or its businesses. If you have received this e-mail by error, please ignore it and notify the sender. image001.jpg___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Now
Can someone tell me what is included in this distro? Does it have voicemail, meetme, panel, and IVR? Thanks, Wiley E. Siler Director of Information Technology 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] www.education2020.com http://www.education2020.com/ Helping students on a mission. Graduation and beyond. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom Phones
Can anyone tell me which config file tells the phone what file to load as bootrom.ld? Or is this hardcoded in the phone? I just got a IP501 but I have a bunch of IP500s... Will the bootrom (2.6.2) work OK with both the IP500 and 501? Thanks! Wiley E. Siler Director of Information Technology 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:[EMAIL PROTECTED] www.education2020.com http://www.education2020.com/ Helping students on a mission. Graduation and beyond. attachment: image001.jpg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax Blast over IP?
Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? I use Asterisk now for my phone system. Thanks! Wiley E. Siler Director of Information Technology Education 2020 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] www.education2020.com http://www.education2020.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax Blast over IP?
Basically, I want to send bulk faxes to a list of my clients. It is time consuming for a person to individually fax so a blast type solution seems best. Over IP is of course to save money... Thanks for the link, reading now... Any suggestions for the blast then? Wiley E. Siler Director of Information Technology Education 2020 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:[EMAIL PROTECTED] www.education2020.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Thursday, April 12, 2007 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax Blast over IP? On Thu, 12 Apr 2007, Wiley Siler said something to this effect: Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? Asterisk can send faxes, if you make it interoperate with a few well-known open-source utilities and/or software packages, depending on what precisely you want to do: http://www.voip-info.org/wiki-Asterisk+fax -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax Blast over IP?
Thanks all... Looks like I will have to let them know that FOIP is a no go and that we can automate on Asterisk though... Thanks! Wiley E. Siler Director of Information Technology Education 2020 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:[EMAIL PROTECTED] www.education2020.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Howard Sent: Thursday, April 12, 2007 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax Blast over IP? Wiley Siler wrote: Basically, I want to send bulk faxes to a list of my clients. It is time consuming for a person to individually fax so a blast type solution seems best. Over IP is of course to save money... Thanks for the link, reading now... Any suggestions for the blast then? My suggestions are in the reading material. Basically it boils down to you not using VoIP for fax. Lee. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Thursday, April 12, 2007 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax Blast over IP? On Thu, 12 Apr 2007, Wiley Siler said something to this effect: Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? Asterisk can send faxes, if you make it interoperate with a few well-known open-source utilities and/or software packages, depending on what precisely you want to do: http://www.voip-info.org/wiki-Asterisk+fax -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nufone
Are these guys still around? I cannot get to www.nufone.net or nufone.com Thanks, Wiley ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Nufone
Strange. I can get there too now... Must have been DNS problem Now to figure out where my DID has gone Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Prior Sent: Monday, January 15, 2007 2:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Nufone Wiley Siler wrote: Are these guys still around? I cannot get to _www.nufone.net_ file://www.nufone.net or nufone.com Not only can I get to their website, but yesterday I called their customer service and for the first time ever it was actually answered by a live person. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IVR woes
Hmm... Wouldn't you just place something in t,1,
To catch the timeout event and loop back to the top of the IVR?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert P.
McKenzie
Sent: Thursday, March 09, 2006 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IVR woes
Sean,
Thanks I've made those changes but still the same problem. The call
falls through if nothing is pushed.
-- Executing Set(IAX2/rob-6, TIMEOUT(digit)=5) in new stack
-- Digit timeout set to 5
-- Executing Set(IAX2/rob-6, TIMEOUT(response)=30) in new stack
-- Response timeout set to 30
== Auto fallthrough, channel 'IAX2/rob-6' status is 'UNKNOWN'
-- Hungup 'IAX2/rob-6'
The hangup is still asterisk dropping the call.
Sean Cook wrote:
If memory servers me correctly DigitTimeout and ResponseTimeout are
depricated...
try:
exten = s,13,Set(TIMEOUT(digit)=5)
exten = s,14,Set(TIMEOUT(response)=30)
Sean
Robert P. McKenzie wrote:
Hello all. I'm having a problem debugging an IVR I'm building. I
can't see any reason this shouldn't be working. Firstly the asterisk
version is:
Asterisk SVN-trunk-r7230 built by root @ localhost.localdomain on a
i686 running Linux on 2006-02-17 22:44:48 UTC
Basically the problem is this. While the playbacks are happening you
can push any one of the options and to happily goes off and does it.
However, if you wait until the messages stop playing back it just
hangs up with the error at the bottome of this message.
Any help in finding a solution to this werid problem would be greatly
appreciated.
The IVR context and console logs are:
[lcl-ivr-main]
;;
; ; This is the main number IVR menu system ;
;;
exten = s,1,Answer exten = s,2,NoOp exten = s,3,NoOp exten =
s,4,NoOp exten = s,5,Wait(1) exten =
s,6,Background(LCL/prompt-00) exten =
s,7,Background(LCL/prompt-01) exten =
s,8,Background(LCL/prompt-02) exten =
s,9,Background(LCL/prompt-03) exten =
s,10,Background(LCL/prompt-04) exten =
s,11,Background(LCL/prompt-05) exten =
s,12,Background(LCL/prompt-09) exten = s,13,DigitTimeout,5 exten =
s,14,ResponseTimeout,30
; exten = _1,1,Background(LCL/prompt-20) ; Sales exten =
_1,2,Dial(${SALES}|40|trwo) exten = _1,3,Voicemail([EMAIL PROTECTED])
exten = _1,103,Voicemail([EMAIL PROTECTED]) exten = _1,4,Hangup
; exten = _2,1,Background(LCL/prompt-30) ; Support exten =
_2,2,Dial(${SUPPORT}|40|trwo) exten =
_2,3,Voicemail([EMAIL PROTECTED]) exten =
_2,103,Voicemail([EMAIL PROTECTED]) exten = _2,4,Hangup
; exten = _3,1,Background(LCL/prompt-40) ; Accounts exten =
_3,2,Dial(${ACCOUNTS}|40|trwo) exten =
_3,3,Voicemail([EMAIL PROTECTED]) exten =
_3,103,Voicemail([EMAIL PROTECTED]) exten = _3,4,Hangup
; exten = _4,1,Background(LCL/prompt-50) ; Reception exten =
_4,2,Dial(${RECEPTION}|40|trwo) exten =
_4,3,Voicemail([EMAIL PROTECTED]) exten =
_4,103,Voicemail([EMAIL PROTECTED]) exten = _4,4,Hangup
; exten = _5,1,NoOp ; Dial Extension ; exten =
_6,1,Goto(lcl-ivr-menu,s,7) ; Play menu again ; exten =
i,1,Goto(lcl-ivr-menu,s,7) ; Return to menu after a time out exten =
t,1,Goto(lcl-ivr-menu,s,7) ; Return to menu after a time out
Here is he asterisk console output:
-- Accepting AUTHENTICATED call from xx.xx.xx.xx:
requested format = unknown, requested prefs = (), actual format =
ulaw, host prefs = (ulaw|alaw|gsm), priority = mine
-- Executing Goto(IAX2/rob-5, lcl-ivr-main|s|1) in new stack --
Goto (lcl-ivr-main,s,1) -- Executing Answer(IAX2/rob-5, ) in new
stack -- Executing NoOp(IAX2/rob-5, ) in new stack -- Executing
NoOp(IAX2/rob-5, ) in new stack -- Executing NoOp(IAX2/rob-5,
) in new stack -- Executing Wait(IAX2/rob-5,
1) in new stack -- Executing BackGround(IAX2/rob-5,
LCL/prompt-00) in new stack -- Playing 'LCL/prompt-00' (language
'en') -- Executing BackGround(IAX2/rob-5, LCL/prompt-01) in new
stack -- Playing 'LCL/prompt-01' (language 'en') -- Executing
BackGround(IAX2/rob-5, LCL/prompt-02) in new stack -- Playing
'LCL/prompt-02' (language 'en') -- Executing BackGround(IAX2/rob-5,
LCL/prompt-03) in new stack -- Playing 'LCL/prompt-03' (language
'en') -- Executing BackGround(IAX2/rob-5, LCL/prompt-04) in new
stack -- Playing 'LCL/prompt-04' (language 'en') -- Executing
BackGround(IAX2/rob-5, LCL/prompt-05) in new stack -- Playing
'LCL/prompt-05' (language 'en') -- Executing BackGround(IAX2/rob-5,
LCL/prompt-09) in new stack -- Playing 'LCL/prompt-09' (language
'en') -- Executing DigitTimeout(IAX2/rob-5, 5) in new stack --
Set Digit Timeout to 5 -- Executing ResponseTimeout(IAX2/rob-5,
30) in new stack
-- Set Response Timeout to 30 == Auto fallthrough, channel
'IAX2/rob-5' status is 'UNKNOWN' -- Hungup 'IAX2/rob-5'
That hangup is Asterisk just dumping out..
___
--Bandwidth
RE: [Asterisk-Users] Asterisk 1.2.3 Released - Critical Update
Hmm... And Nufone is down suddenly Coincidence or other? Stated reason was multiple hardware failure. Somehow I am betting anyone with this problem already noticed too... W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Development Team Sent: Wednesday, January 25, 2006 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk 1.2.3 Released - Critical Update Asterisk 1.2.3 has been released and contains a number of bug fixes. One of the fixes is for a critical bug introduced in version 1.2.2 that will cause an Asterisk server to stop processing calls correctly when the server's clock reaches January 25th, 2006 (today). It is vital to upgrade all 1.2.2 servers with this release as soon as possible. Thank you for your support for Asterisk! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.2.3 Released - Critical Update
Excellent to know. Fortunately for me I don't have any scheduled use on my DID from them today. Phew... I am surprised that hot swaps are not more common practice. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, January 25, 2006 3:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk 1.2.3 Released - Critical Update Not. Jer would have had * corrected in a nano-jiffy. Given the timeframe stated, sounds like they had a backbone device failure and waiting for parts to show up (or hard drive failure, or something like that). Hmm... And Nufone is down suddenly Coincidence or other? Stated reason was multiple hardware failure. Somehow I am betting anyone with this problem already noticed too... W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Development Team Sent: Wednesday, January 25, 2006 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk 1.2.3 Released - Critical Update Asterisk 1.2.3 has been released and contains a number of bug fixes. One of the fixes is for a critical bug introduced in version 1.2.2 that will cause an Asterisk server to stop processing calls correctly when the server's clock reaches January 25th, 2006 (today). It is vital to upgrade all 1.2.2 servers with this release as soon as possible. Thank you for your support for Asterisk! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.2.3 Released - Critical Update
I am sure. My meaning was not that they should not have posted. I was just commenting that people in position most assuredly knew SOMETHING was up. I would have people banging down my door if I had services die like that. Hopefully that update will fix your problem. I feel for you dude. W From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rusty DekemaSent: Wednesday, January 25, 2006 3:32 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Asterisk 1.2.3 Released - Critical Update I, for one, am glad this message was posted because I was about to call Digium and try to RMA my TDM400B card. Calls using it (in and out) have stopped working as of today (although pure-VoIP calls seem to work fine) for absolutely no reason that I can ascertain. I am about to upgrade to 1.2.3 and I suspect that it will solve the problem.-Rusty On 1/25/06, Wiley Siler [EMAIL PROTECTED] wrote: Hmm... And Nufone is down suddenlyCoincidence or other?Stated reason was multiple hardware failure. Somehow I am betting anyone with this problem already noticed too...W-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of AsteriskDevelopment TeamSent: Wednesday, January 25, 2006 12:48 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk 1.2.3 Released - Critical UpdateAsterisk 1.2.3 has been released and contains a number of bug fixes. Oneof the fixes is for a critical bug introduced in version 1.2.2 that will cause an Asterisk server to stop processing calls correctly when theserver's clock reaches January 25th, 2006 (today). It is vital toupgrade all 1.2.2 servers with this release as soon as possible.Thank you for your support for Asterisk! ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.2.3 Released - Critical Update
I am going to assume the best and hope it was a an issue of testing code missed at release. How or why that would be the case is beyond me but I sure hope that is the case. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Technical Support Sent: Wednesday, January 25, 2006 4:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk 1.2.3 Released - Critical Update Why would the software halt on that date? Is there a time bomb in Asterisk? I can't imagine what legit piece of code would be checking for a particular date -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, January 25, 2006 5:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk 1.2.3 Released - Critical Update Not. Jer would have had * corrected in a nano-jiffy. Given the timeframe stated, sounds like they had a backbone device failure and waiting for parts to show up (or hard drive failure, or something like that). Hmm... And Nufone is down suddenly Coincidence or other? Stated reason was multiple hardware failure. Somehow I am betting anyone with this problem already noticed too... W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Development Team Sent: Wednesday, January 25, 2006 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk 1.2.3 Released - Critical Update Asterisk 1.2.3 has been released and contains a number of bug fixes. One of the fixes is for a critical bug introduced in version 1.2.2 that will cause an Asterisk server to stop processing calls correctly when the server's clock reaches January 25th, 2006 (today). It is vital to upgrade all 1.2.2 servers with this release as soon as possible. Thank you for your support for Asterisk! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom 301 DTMF
All my Polycoms are set to... dtmfmode=rfc2833 Should solve your problems. Best configuration is through the config files and using an FTP or TFTP server. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Michaelson Sent: Wednesday, January 18, 2006 3:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom 301 DTMF Just got a Polycom 301 and I'm configuring. Examples given in wiki recommend using dtmfmode=inband, so that's what I set in sip.conf for this phone, as I have for various other IP phones on my network. But the telephone does not seem to send DTMF tones up thru the network (although I hear them in the handset when I bang the buttons). Also, I can't seem to find a corresponding parameter in the web-based config pages of the phone. Can anyone give me some hints about how best to configure this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] experiences with teliax, voipjet or junction networks?
VoipJet has been great to me for dial time. Nufone.net is where I get my inward dialing for my VoIP. Also good experience so far. Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, January 17, 2006 4:45 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] experiences with teliax,voipjet or junction networks? We are looking for SIP trunks for our * pbx for our business. Being able to port our numbers is an absolute requirement. teliax can do it, but I am unsure of the others. Anyone have experiences (good, bad) with the above mentioned providers to share? Eg reliability, quality, etc. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FXS or VOIP
Think of it this way. VoIP phones allow you to place a phone anywhere that a network connection exists. Your Asterisk box will be on the network and will be easily accessible. FXO and analog phones require point to point termination. Phone to FXO. Period. What a pain! VoIP phones are relatively cheap and look/work really nice. Just buy a 4 port FXO card from Digium and connect your 4 analog lines to the * box. Or you can even contact me off list if you want to buy my old one. I just moved to PRI. Get a analog to SIP gateway (Sipura SPA-1001 for example) and connect your fax into the system. Just the ease of use alone is worth using VoIP phones. A single computer can handle a HUGE amount of VoIP phones. The phone connects to your network and talks to the asterisk server over the network. I have 20 phones on my network with no issue at all. Others have many many more. Contact me off list if you want some newbie help. I have done this a couple of times and am more than willing to help out a first timer. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Freeze Sent: Wednesday, January 11, 2006 2:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FXS or VOIP On 1/11/06, William Boehlke [EMAIL PROTECTED] wrote: You can save a little money with analog phones however if that saving is not an issue business class VoIP phones from providers like Polycom and Cisco have more features and much of the time better call quality. Thanks William for the response. That is good news about the phone quality. From what I have read, I think the overall cost would still be cheaper with a voip solution, even if the phones are more. A 4 line FXS card is about $3-400 (I think). If I understand this correctly, even if I have only 4 lines incoming, I need an FXS homerun to each phone. So for 5 phones, I would need 2 cards. And, the O'Reilly book says that I should not put 2 cards in the same box, so I would need another computer. I was hoping a single computer could handle up to 10 voip phones. Am I deluding myself? Jim Hi I am setting up a phone system for a small office. The office will have 5-8 phones and a fax line. There are 4 hunt lines coming into the office. We have made no hardware purchase yet. Being an asterisk newbie, before I suscribed to this list I just assumed that I would buy voip phones and connect all the phones to a private ethernet network. However, I see many people inquiring about FXS cards. Is there any reason why I would need to consider using analog phones and FXS cards? Seems to me the cheapest way is with voip phones and voice quality should be good since the phones are on a private network that only has voice traffic. -- Jim Freeze ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FXS or VOIP
Or FXS... Whatever. The point is port connect directly. No one spam me on this one... 8) Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Wednesday, January 11, 2006 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] FXS or VOIP Think of it this way. VoIP phones allow you to place a phone anywhere that a network connection exists. Your Asterisk box will be on the network and will be easily accessible. FXO and analog phones require point to point termination. Phone to FXO. Period. What a pain! VoIP phones are relatively cheap and look/work really nice. Just buy a 4 port FXO card from Digium and connect your 4 analog lines to the * box. Or you can even contact me off list if you want to buy my old one. I just moved to PRI. Get a analog to SIP gateway (Sipura SPA-1001 for example) and connect your fax into the system. Just the ease of use alone is worth using VoIP phones. A single computer can handle a HUGE amount of VoIP phones. The phone connects to your network and talks to the asterisk server over the network. I have 20 phones on my network with no issue at all. Others have many many more. Contact me off list if you want some newbie help. I have done this a couple of times and am more than willing to help out a first timer. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Freeze Sent: Wednesday, January 11, 2006 2:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FXS or VOIP On 1/11/06, William Boehlke [EMAIL PROTECTED] wrote: You can save a little money with analog phones however if that saving is not an issue business class VoIP phones from providers like Polycom and Cisco have more features and much of the time better call quality. Thanks William for the response. That is good news about the phone quality. From what I have read, I think the overall cost would still be cheaper with a voip solution, even if the phones are more. A 4 line FXS card is about $3-400 (I think). If I understand this correctly, even if I have only 4 lines incoming, I need an FXS homerun to each phone. So for 5 phones, I would need 2 cards. And, the O'Reilly book says that I should not put 2 cards in the same box, so I would need another computer. I was hoping a single computer could handle up to 10 voip phones. Am I deluding myself? Jim Hi I am setting up a phone system for a small office. The office will have 5-8 phones and a fax line. There are 4 hunt lines coming into the office. We have made no hardware purchase yet. Being an asterisk newbie, before I suscribed to this list I just assumed that I would buy voip phones and connect all the phones to a private ethernet network. However, I see many people inquiring about FXS cards. Is there any reason why I would need to consider using analog phones and FXS cards? Seems to me the cheapest way is with voip phones and voice quality should be good since the phones are on a private network that only has voice traffic. -- Jim Freeze ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FXS or VOIP
I think what he means is that an * server can support hundreds of phones because the server connects to the network via a NIC. Port count becomes irrelevant when you thing about VoIP phones connecting to a VoIP server. They connect over the network not point to point. It is just a matter of bandwidth and network topography at that point. If your desire it to connect a bunch of analog phones then you can use ATAs. I would recommend that you just replace them though. You should only need to worry about your POTS lines at this point. If you have 4 POTS lines, a single 4 port TDM card will suffice. Cheers, W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Freeze Sent: Wednesday, January 11, 2006 2:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FXS or VOIP On 1/11/06, William Boehlke [EMAIL PROTECTED] wrote: A single computer will handle hundreds of telephones. Just get a card with more ports, or use an external gateway. I am sorry, I don't understand. Are you talking about analog FXS phones? All the PCI cards I have seen have a max of 4 FXS lines and the external boxes seem very expensive. -- Jim Freeze ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: RE: [Asterisk-Users] FXS or VOIP
If you are doing a whole bunch of fax machines then the external solution Brian spoke of is probably best. If we are talking 4 fax machines, you can get a TDM card and 4 FXS modules and connect then right to the * box. However, you have no growth space at that point without hardware changes. Is that an issue or will your needs remain static? My PRI line works just fine for fax. I am configured thusly... PRI T1 (T100P) --- * Server (NIC) --- Network --- SIP ATA --- Fax Machine This works flawlessly and is a great solution when you only need a couple of fax machines. Asterisk is capable of routing fax from an analog device to a PRI T1 if that is your question. As long as you have a TDM card or ATAs for your analog devices, they will be connected to the * box. PRI has worked fine for me with FAX so there is no need to connect to the PSTN just for fax. YMMV but I bet it is fine. How many fax lines do you need? W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Crew Sent: Wednesday, January 11, 2006 3:43 PM To: asterisk-users@lists.digium.com Subject: Re: RE: [Asterisk-Users] FXS or VOIP What about fax machines working over a PRI T1 line? Run FXS ports to each fax machine and the TDM card will convert the digital T1 to analog for faxing? I have no POTS lines, just a T1 (PRI soon if I find out I can use asterisk for regular POTS-type faxing). Begin Original Message From: Wiley Siler [EMAIL PROTECTED] Sent: Wed, 11 Jan 2006 15:20:03 -0700 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] FXS or VOIP I think what he means is that an * server can support hundreds of phones because the server connects to the network via a NIC. Port count becomes irrelevant when you thing about VoIP phones connecting to a VoIP server. They connect over the network not point to point. It is just a matter of bandwidth and network topography at that point. If your desire it to connect a bunch of analog phones then you can use ATAs. I would recommend that you just replace them though. You should only need to worry about your POTS lines at this point. If you have 4 POTS lines, a single 4 port TDM card will suffice. Cheers, W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Freeze Sent: Wednesday, January 11, 2006 2:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FXS or VOIP On 1/11/06, William Boehlke [EMAIL PROTECTED] wrote: A single computer will handle hundreds of telephones. Just get a card with more ports, or use an external gateway. I am sorry, I don't understand. Are you talking about analog FXS phones? All the PCI cards I have seen have a max of 4 FXS lines and the external boxes seem very expensive. -- Jim Freeze ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users End Original Message Sent by Go2net Mail! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FXS or VOIP
The consensus is usually a big NO though some have made more than 2 cards work. I ran my system with 2 four port TDM cards and it worked fine. Others have had nothing but problems. This has to do with IRQs and the PCI bus if memory serves. A quick search should yield info on IRQ and TDM cards. Cheers, W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philip Edelbrock Sent: Wednesday, January 11, 2006 3:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FXS or VOIP Jim Freeze wrote: [...] So for 5 phones, I would need 2 cards. And, the O'Reilly book says that I should not put 2 cards in the same box, so I would need another computer. [...] Whoa, I'm confused. Can't you use as many cards as you have slots? We've got just one 4-port card, but I've always assumed it was just a matter of purchasing and installing more to get 8 or 12 lines? Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialer
If this or any other example is available, I would be most thankful to have it. I got the go ahead on this project to day so now I have to start seeing how to do this. Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe Sent: Tuesday, January 03, 2006 5:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dialer I'm supposed to have a mostly canned script that will do this done already. It will pull the list of people to call out of a db and play them the file specified in the db table. Contact me offlist if you're interested. It will be done real soon but I'm not done testing yet. Darren Wiebe [EMAIL PROTECTED] Kerry Garrison wrote: You actually aren't far from it. If the system only needs to play the same file to each person, a simple script can be used to pull from a database and create call files. Asterisk will use the call files to place the calls and play a sound. A few minutes of searching on that should get you started. I haven't seen anyone else have a canned script ready to go, but would like to know if anyone does. -Kerry *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Wiley Siler *Sent:* Tuesday, January 03, 2006 3:32 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] Dialer Hello All, I am having trouble finding a specific * piece of software so I thought I would see If you guys can help me get my terminology clear. First off let me premise this with no, this is absolutely not for doing call marketing. I need to make my Asterisk box call a group of people and play them a message. My company deals with education so we need to do follow ups if students are not logging on. We do this manually now but it would be easier and cheaper to just play them a message. What is the term I should be looking for? I keep thinking auto dialer or something like that but I am not quite getting there. Any help would be appreciated. I have been learning a bit of Perl so I was thinking I could auto generate and AGI file and then just do a Play() of the mp3 when they pick up at the other end? Seems a little kludge though. Thanks, Wiley --- - ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialer
Just to make it easy, I will be reading the caller list from a another server via a web page, parsing it and dialing. After each pass, I just post back to the server web page and it updates the other system. Our tech just needs to review the log once daily. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Higley Sent: Friday, January 06, 2006 11:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Dialer A really neat thing about this, you could make it interactive, and also post the response back from each user on if they accepted it or not. and then call them back in 5 min again :) LOL But someone could be seeing what the system is doing realtime... ./Ben Hello All, I am having trouble finding a specific * piece of software so I thought I would see If you guys can help me get my terminology clear. First off let me premise this with no, this is absolutely not for doing call marketing. I need to make my Asterisk box call a group of people and play them a message. My company deals with education so we need to do follow ups if students are not logging on. We do this manually now but it would be easier and cheaper to just play them a message. What is the term I should be looking for? I keep thinking auto dialer or something like that but I am not quite getting there. Any help would be appreciated. I have been learning a bit of Perl so I was thinking I could auto generate and AGI file and then just do a Play() of the mp3 when they pick up at the other end? Seems a little kludge though. Thanks, Wiley -- - - ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialer
Very cool! Is this something you can share the code? Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel Sent: Friday, January 06, 2006 2:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Dialer On Fri, 2006-01-06 at 11:45 -0700, Wiley Siler wrote: Just to make it easy, I will be reading the caller list from a another server via a web page, parsing it and dialing. After each pass, I just post back to the server web page and it updates the other system. Our tech just needs to review the log once daily. That is basically what I did for a customer. I have a DB that is filtered pursuant to 47 CFR 64.1200 and 16 CFR 310 (US federal laws concerning these types of systems -- not calling to the US, dont worry about it). I wrote some tools to make that a snap. I then have 1-N clients pull from the DB servier via HTTP to get the next number to dial and context to goto. The dialplan updates the DB via HTTP so the status of a given number is known and prevents duplicate calls. I added answering machine detection to my asterisk server and a few other things to make the dialing slightly better. The way it works they can have many many calling systems if they need, nothing has to be local to each other. Reports can be generated off any data that is available (timestamps of events, status of calls, etc). This is perfect for dr appt reminders, batch calls saying 'your product has been shipped' etc. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialer
Title: Dialer Hello All, I am having trouble finding a specific * piece of software so I thought I would see If you guys can help me get my terminology clear. First off let me premise this with no, this is absolutely not for doing call marketing. I need to make my Asterisk box call a group of people and play them a message. My company deals with education so we need to do follow ups if students are not logging on. We do this manually now but it would be easier and cheaper to just play them a message. What is the term I should be looking for? I keep thinking auto dialer or something like that but I am not quite getting there. Any help would be appreciated. I have been learning a bit of Perl so I was thinking I could auto generate and AGI file and then just do a Play() of the mp3 when they pick up at the other end? Seems a little kludge though. Thanks, Wiley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best Switch for VOIP Applications
What is your port density requirement? For 24 ports the LinkSys SRW2024 is awesome. They street for less than $500 and have good QoS. For a smaller switch, they make a 12 port variant. Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of calvis Sent: Monday, December 05, 2005 3:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Best Switch for VOIP Applications I need to replace my switch. Does anyone have any recommendations for a switch that is VoIP friendly? I want it to be a managed gigabyte switch. There are lots of brands out there, but would prefer some recommendations from the list. -Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best Switch for VOIP Applications
Cisco owns Linksys so they have some good features now. 64 VLANs, 8 port trunking groups, console port, 802.1p CoS support Four Quality of Service egress queues per port let you prioritize traffic via 802.1p. http://www1.linksys.com/products/product.asp?grid=35scid=40prid=673 This can be found for close to $400. Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: Monday, December 05, 2005 3:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Best Switch for VOIP Applications Wiley Siler wrote: What is your port density requirement? For 24 ports the LinkSys SRW2024 is awesome. They street for less than $500 and have good QoS. For a smaller switch, they make a 12 port variant. Does the SRW2024 support port mirroring? I was shopping around, but couldn't find any Linksys switch that support port mirroring. I ended with the DLINK DES-1226G which retails for a lot less than the SRW2024 (over here we can get it for US$300) and has VLAN (port-based or 802.1q) and port mirroring. Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of calvis Sent: Monday, December 05, 2005 3:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Best Switch for VOIP Applications I need to replace my switch. Does anyone have any recommendations for a switch that is VoIP friendly? I want it to be a managed gigabyte switch. There are lots of brands out there, but would prefer some recommendations from the list. -Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] which hardware should i use??????????
I recommend checking the following site... www.voip-info.org Lots of info for you there... By VoIP phones, I think you are meaning soft phones which are software based. You will need a headset for the PC that runs the software phone. Usually Logitech or Plantronics at about $50 a headset. If you want a hardware phone on a budget then Snom and Grandstream are popular. The Asterisk servers only need a card if they will be connected to the PSTN. So for a bunch of POTS line you want a TDM card or a channel bank and T1 card. The T1 card for a real newbie would be the Digium brand. A more advanced user might consider a Sangoma. Good luck and hope you are doing well in Pakistan. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ishtiaq ahmed Sent: Wednesday, October 12, 2005 1:08 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] which hardware should i use?? hy all i need a suggesion on what hardware should i use for the following case study i have five offices each will be having 35 to 45 extensions. if i will be using voip fones for those extensions( either it is iax or sip ) which one will be better and cheaper what should i use. all the five offices will be connected through asterisk servers ( one in each of the offices ). now the confusion is that is there any hardware needed to connect the voip fones to the asterisk server( how we can connect them to asterisk server ). and for outgoing calls to the pstn network which card should be used. plzz guide me thouroly about the hardware. i have asked a lot of people every one is giving his own suggestion. so i thought to ask from the official mailing list. i hope that i will be getting a good response. i live in pakistan. __ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WCFXO and T1 PRI Card?
Title: WCFXO and T1 PRI Card? Can I have a TDM400 and a T100P in the same machine? I am using AAH and trying to combine two boxes. If so, can anyone tell me the proper config for zaptel.conf and zapata.conf? Thanks! Wiley ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WCFXO and T1 PRI Card?
Title: WCFXO and T1 PRI Card? Well, partial success so far Here is my ztcfg SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01)Channel 02: FXS Kewlstart (Default) (Slaves: 02)Channel 03: FXS Kewlstart (Default) (Slaves: 03)Channel 04: FXS Kewlstart (Default) (Slaves: 04)Channel 05: Individual Clear channel (Default) (Slaves: 05)Channel 06: Individual Clear channel (Default) (Slaves: 06)Channel 07: Individual Clear channel (Default) (Slaves: 07)Channel 08: Individual Clear channel (Default) (Slaves: 08)Channel 09: Individual Clear channel (Default) (Slaves: 09)Channel 10: Individual Clear channel (Default) (Slaves: 10)Channel 11: Individual Clear channel (Default) (Slaves: 11)Channel 12: Individual Clear channel (Default) (Slaves: 12)Channel 13: Individual Clear channel (Default) (Slaves: 13)Channel 14: Individual Clear channel (Default) (Slaves: 14)Channel 15: Individual Clear channel (Default) (Slaves: 15)Channel 16: Individual Clear channel (Default) (Slaves: 16)Channel 17: Individual Clear channel (Default) (Slaves: 17)Channel 18: Individual Clear channel (Default) (Slaves: 18)Channel 19: Individual Clear channel (Default) (Slaves: 19)Channel 20: Individual Clear channel (Default) (Slaves: 20)Channel 21: Individual Clear channel (Default) (Slaves: 21)Channel 22: Individual Clear channel (Default) (Slaves: 22)Channel 23: Individual Clear channel (Default) (Slaves: 23)Channel 24: Individual Clear channel (Default) (Slaves: 24)Channel 25: Individual Clear channel (Default) (Slaves: 25)Channel 26: Individual Clear channel (Default) (Slaves: 26)Channel 27: Individual Clear channel (Default) (Slaves: 27)Channel 28: D-channel (Default) (Slaves: 28) 28 channels configured. However, asterisk will not start. I thought zapta.conf was OK but maybe I am wrong... Anyone able to throw me a hint on this one? Thanks,Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley SilerSent: Thursday, October 06, 2005 8:53 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] WCFXO and T1 PRI Card? Can I have a TDM400 and a T100P in the same machine? I am using AAH and trying to combine two boxes. If so, can anyone tell me the proper config for zaptel.conf and zapata.conf? Thanks! Wiley ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WCFXO and T1 PRI Card?
Title: WCFXO and T1 PRI Card? I am getting an error about a broken pipe when I run asterisk -vvvc It reads zapata.conf as Found then dumps this error about a broken sound pipe? W From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley SilerSent: Thursday, October 06, 2005 9:10 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] WCFXO and T1 PRI Card? Well, partial success so far Here is my ztcfg SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01)Channel 02: FXS Kewlstart (Default) (Slaves: 02)Channel 03: FXS Kewlstart (Default) (Slaves: 03)Channel 04: FXS Kewlstart (Default) (Slaves: 04)Channel 05: Individual Clear channel (Default) (Slaves: 05)Channel 06: Individual Clear channel (Default) (Slaves: 06)Channel 07: Individual Clear channel (Default) (Slaves: 07)Channel 08: Individual Clear channel (Default) (Slaves: 08)Channel 09: Individual Clear channel (Default) (Slaves: 09)Channel 10: Individual Clear channel (Default) (Slaves: 10)Channel 11: Individual Clear channel (Default) (Slaves: 11)Channel 12: Individual Clear channel (Default) (Slaves: 12)Channel 13: Individual Clear channel (Default) (Slaves: 13)Channel 14: Individual Clear channel (Default) (Slaves: 14)Channel 15: Individual Clear channel (Default) (Slaves: 15)Channel 16: Individual Clear channel (Default) (Slaves: 16)Channel 17: Individual Clear channel (Default) (Slaves: 17)Channel 18: Individual Clear channel (Default) (Slaves: 18)Channel 19: Individual Clear channel (Default) (Slaves: 19)Channel 20: Individual Clear channel (Default) (Slaves: 20)Channel 21: Individual Clear channel (Default) (Slaves: 21)Channel 22: Individual Clear channel (Default) (Slaves: 22)Channel 23: Individual Clear channel (Default) (Slaves: 23)Channel 24: Individual Clear channel (Default) (Slaves: 24)Channel 25: Individual Clear channel (Default) (Slaves: 25)Channel 26: Individual Clear channel (Default) (Slaves: 26)Channel 27: Individual Clear channel (Default) (Slaves: 27)Channel 28: D-channel (Default) (Slaves: 28) 28 channels configured. However, asterisk will not start. I thought zapta.conf was OK but maybe I am wrong... Anyone able to throw me a hint on this one? Thanks,Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley SilerSent: Thursday, October 06, 2005 8:53 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] WCFXO and T1 PRI Card? Can I have a TDM400 and a T100P in the same machine? I am using AAH and trying to combine two boxes. If so, can anyone tell me the proper config for zaptel.conf and zapata.conf? Thanks! Wiley ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Config PolyCom SoundStation 4000 help
No doc but I can tell you that the easiest thing to do is use a config file and ftp if you have the ability. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of kurt x Sent: Wednesday, October 05, 2005 2:13 PM To: Asterisk Subject: [Asterisk-Users] Config PolyCom SoundStation 4000 help I am trying to get a IP 4000 to register to Asterisk. I can make outbound calls from the IP 4000 but not to it. When I implement sip show peers it lists the extension but with no IP address (unspecified). I am configuring the phone via the web interface. I am not using ftp or tftp to configure the phone. Does anyone have a doc explaining how to get the phone to register to asterisk. Thanks, Kurt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [EMAIL PROTECTED] Questions
Michael, This is the list for Asterisk not Asterisk at Home. That list list can be found at the same place you downloaded the AAH software. www.voip-info.org is the location of Asterisk Wiki. Now. On to your problems. 1. No one can help you with that problem since we don know what kind of phone or what kind of modem. you cannot use just any modem. It has to be either a Digium TDM card or a clone with the Tiger chipset. Connection after that is a matter of FSO versus FXO. Google for that if explanationneeded... 2. Go to Wiki I listed above and you can find how to setup paging. Then you just need to create an extension in the AAH setup that does what you want. And finally... If you are just getting started, you have about a 2 week learning curve so immerse yourself in the Wiki and all will be well... Good luck, Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael JanofskySent: Wednesday, September 28, 2005 4:01 PMTo: Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] [EMAIL PROTECTED] Questions Hi! I am new to this, and have been googling for hours to solve this issue. First question: How do I go about setting up my POTS line through a modem that is in my server? I can also not get my "hardphone" to connect to asterisk. The softphones can talk to each other and through a GoIAX VoIP line. Second Question: (No direct how-to or answer online) How do I set up [EMAIL PROTECTED] to page through the soundcard in the server. I would like to dial a code from a phone or selected phones (say 1234 as the code) and have it play a beep and page the speakers. Thanks, Michael PS If this should not be here, would someone willing to help contact me off-list? Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Buy a digium hardware
Assuming you can purchase online, just go to voipsupply.com. http://www.voipsupply.com/index.php?manufacturers_id=13 The switch between analog and digital makes a huge difference to port density. With an analog TDM card you can get 4 FXO/FXS ports per card. With a digital T1PRI card, you can get 4 T1 spans with 23 voice channels each. If you are going to use a lot of analog ports (more than 8) then youmay benefit from moving to a channel bank and installing a PRI card to the Asterisk box. You can find more info at... http://www.voip-info.org/ Cheers, Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leopoldo Rodríguez HSent: Monday, September 19, 2005 8:24 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Buy a digium hardware Where i can buy a digium hardware TDM400P in Mexicois there a hardware with more than 4 FXS/FXO ports (8, 12, 24)? that is supported by Asterisk*RegardsLeopoldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] wiki down?
I got right in just fine... W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francesco Peeters Sent: Friday, September 16, 2005 10:26 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] wiki down? I'm unable to connect to voip-info.org... Anybody else have the same issues, ro is it just me? -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] LiveVOIP - I win :)
LOL - Congrats! $30 down... Let's see... how much to go? W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Monday, September 12, 2005 1:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] LiveVOIP - I win :) A few months ago, the friendly folks from liveVOIP went under. We had some discussion on how to limit our losses, and my recommendation was a chargeback, since FTTP Services -- their CC merchant -- wasn't affected by the bankruptcy, as far as we could tell. Today, I received this from my CC company: http://muware.com/asterisk/livevoip.pdf Anyone else got lucky? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] First PRI Installed - WOOT
Title: First PRI Installed - WOOT Today I got my first PRI installed. It literally took less than 5 minutes and the circuit was up and we were making calls. The T100P is performing excellent. The Linux/Asterisk box is running well and the quality is great. The line is from MCI and they did a great job. I know this is not the usual banter but I just thought I would share a good experience and throw out some props to Digium and Mark. I love it when things work well and work the first time. Cheers, Wiley ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voice over atlantic
Pay the license fee and get the GSM codec would probably be best. The fee is nominal and the codec is a good one... $0.02 W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Hajek Sent: Thursday, September 08, 2005 1:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] voice over atlantic Hi- I'm using IAX between two boxes, where one box is located in US and the second in Europe. I'm trying to achieve the best voice quality and mainly reliability between these boxes and looking for hints and experience of others. Facts: - Asterisk 1.0.7 - RTT varies from 130-170 ms, depends on time and actual Internet throughput Questions: - What is the sugested codec for such setup? Now I'm using ULAW, but realizing it may not be the best choice. Sometimes I can hear broken audio. Maybe speex is better choice? - Jitter buffer, yes/no? What are the suggested values. Currently I'm using these values: jitterbuffer=yes dropcount=10 maxjitterbuffer=500 maxexcessbuffer=300 minexcessbuffer=20 jittershrinkrate=2 - Trunking? Is it reliable enough? Thanks for any hints. -- David ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voice over atlantic
http://www.digium.com/index.php?menu=product_detailcategory=extrasprod uct=G729 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Hajek Sent: Thursday, September 08, 2005 2:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] voice over atlantic Probably missing something here. Never heard of GSM commercial licence for asterisk. Do you have any URLs? Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Thursday, September 08, 2005 11:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] voice over atlantic Pay the license fee and get the GSM codec would probably be best. The fee is nominal and the codec is a good one... $0.02 W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Hajek Sent: Thursday, September 08, 2005 1:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] voice over atlantic Hi- I'm using IAX between two boxes, where one box is located in US and the second in Europe. I'm trying to achieve the best voice quality and mainly reliability between these boxes and looking for hints and experience of others. Facts: - Asterisk 1.0.7 - RTT varies from 130-170 ms, depends on time and actual Internet throughput Questions: - What is the sugested codec for such setup? Now I'm using ULAW, but realizing it may not be the best choice. Sometimes I can hear broken audio. Maybe speex is better choice? - Jitter buffer, yes/no? What are the suggested values. Currently I'm using these values: jitterbuffer=yes dropcount=10 maxjitterbuffer=500 maxexcessbuffer=300 minexcessbuffer=20 jittershrinkrate=2 - Trunking? Is it reliable enough? Thanks for any hints. -- David ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voice over atlantic
Ooops... meant G729 but seems like other suggestion of GSM might do... W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Hajek Sent: Thursday, September 08, 2005 3:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] voice over atlantic Yep. Thats G729, not GSM. Btw, GSM codec implemented in Asterisk is EFR? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Friday, September 09, 2005 12:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] voice over atlantic http://www.digium.com/index.php?menu=product_detailcategory=e xtrasprod uct=G729 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Hajek Sent: Thursday, September 08, 2005 2:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] voice over atlantic Probably missing something here. Never heard of GSM commercial licence for asterisk. Do you have any URLs? Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Thursday, September 08, 2005 11:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] voice over atlantic Pay the license fee and get the GSM codec would probably be best. The fee is nominal and the codec is a good one... $0.02 W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Hajek Sent: Thursday, September 08, 2005 1:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] voice over atlantic Hi- I'm using IAX between two boxes, where one box is located in US and the second in Europe. I'm trying to achieve the best voice quality and mainly reliability between these boxes and looking for hints and experience of others. Facts: - Asterisk 1.0.7 - RTT varies from 130-170 ms, depends on time and actual Internet throughput Questions: - What is the sugested codec for such setup? Now I'm using ULAW, but realizing it may not be the best choice. Sometimes I can hear broken audio. Maybe speex is better choice? - Jitter buffer, yes/no? What are the suggested values. Currently I'm using these values: jitterbuffer=yes dropcount=10 maxjitterbuffer=500 maxexcessbuffer=300 minexcessbuffer=20 jittershrinkrate=2 - Trunking? Is it reliable enough? Thanks for any hints. -- David ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sending fax
Google can translate if that helps...
w
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
ShipmanSent: Thursday, September 08, 2005 4:44 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] sending fax
Thanks, but I can't read
Spanish.
Chris
- Original Message -
From:
Il
Neofita
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, September 05, 2005 2:14
PM
Subject: Re: [Asterisk-Users] sending
fax
Hi,I found on a forum a script that "emulate a hylafax"
this is the linkhttp://www.vocesuip.com/viewtopic.php?p=2423You
can use the WHFC in order to send a fax to asterisk.
On 9/5/05, Harald
Klein [EMAIL PROTECTED]
wrote:
Hi
Chris, Hi Arne,Am 5.9.2005 schrieb "Chris Shipman" [EMAIL PROTECTED]:I'veseen
some programs that install as a printer and create an image. However
this would be to cumbersome for your average user.It would need to
be able to print to as local printer and then send
outAsterisk.What about:Client with Postscript
printer driver Some kind of a printing system (samba with lpr[ng] and/or
cups etc.) toaccess the fax-printer via
smb/cifs/lpr/ipp/whatever..Output filter for the fax-printer to convert
Postscript to tiff andgeneratea call file with App txfax...
The problem is to tell the printer the number to fax to...You
can grep in the Postscript file for a predefined string (for example"Fax
Recpient Nr") and generate some matching templates in your office
suite..Search for HylaFax solutions, they are pretty much the
same...HariChris- Original
Message -From: "Arne Morten Johansen" [EMAIL PROTECTED]To: "Asterisk Users
Mailing List - Non-Commercial Discussion"asterisk-users@lists.digium.comSent:
Monday, September 05, 2005 6:27 AM Subject: SV: [Asterisk-Users]
sending fax What about faxing yourself if you
don't have a scanner? -Opprinnelig
melding- Fra: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]
På vegne av Johan vanTongeren Sendt: 5. september 2005
09:11 Til: Asterisk Users Mailing List - Non-Commercial
Discussion Emne: RE: [Asterisk-Users] sending
fax [macro-fax-dialing] exten =
s,1,SetCIDNum(0${CALLERIDNUM}) exten =
s,2,Dial(Zap/g${ARG2}/${ARG1},20,,t) exten =
s,3,Goto(900) exten = s,103,Goto(900) exten
= s,900,Busy exten = s,901,Hangup
-Oorspronkelijk bericht- Van: [EMAIL PROTECTED]
[mailto:
[EMAIL PROTECTED]] Namens Chris
Shipman Verzonden: maandag 5 september 2005 7:22
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: [Asterisk-Users] sending fax I've read
alot on the wiki about sending and receiving faxes thru
asterisk. I've gotten the receive to work
great.My question is how does one send
a fax? I see lots of instructions about how to send
the image to asterisk by email,
etc.The problem is how
doesone make the image of the fax to
begin with? Has anyone come
up with a good solution for this?
Regards,
Chris
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RE: [Asterisk-Users] ipvolution t1 cards
Last time I talked to them, it was supposedly going to be released in June... Then July,... Then August... These are still vaporware as far as I can tell... If anyone knows of anything different, I would love to hear it... W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Thursday, September 01, 2005 12:24 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ipvolution t1 cards On Thursday 01 September 2005 14:27, Trey Scarborough wrote: Has any one used the Ipvolution tdm120 cards i am intrested to know how well it works and how well the on board dsp's work. I wasn't aware that they were in production yet. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Phone advise
I have one and it is absolutely awesome. Works great and the quality of Polycom conference phones is excellent regardless of protocol. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of kurt x Sent: Friday, August 26, 2005 9:50 AM To: Asterisk Subject: [Asterisk-Users] Polycom Phone advise I would like to know if any body is using the Polycom Soundstation IP 4000 SIP conference phone with Asterisk. I am thinking of purchasing one. Kurt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PCI 2.3
Title: PCI 2.3 Hello All, Anyone know if this is backwards compatible with 2.2? Here is the spec from the Mobo I am looking at. Five 32-bit v2.3 Master PCI bus slots (support 3.3V/5V PCI bus interface). Thanks! Wley ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP providers -- California, U.S.
Bad URL... Too many R's in there... Correct... http://www.voipzoneenterprise.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Watters Sent: Thursday, August 25, 2005 10:29 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] VoIP providers -- California, U.S. http://www.voipzoneenterrprise.com DID's in 92% plus of the USA, can provide full Enterprise solutions from SIP2.0 to Internet access. BRW -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leon Sun Sent: Thursday, August 25, 2005 10:04 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] VoIP providers -- California, U.S. If you want SIP phone PBX hosting or residential partitioning, I can't help. If you want traffic termination(National and International), we can do it. Regards Leon Sun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jennyw Sent: Thursday, August 25, 2005 12:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] VoIP providers -- California, U.S. Hi, Just wondering if people could suggest a good VoIP provider that can service the San Francisco Bay Area and the Los Angeles area. I've tried race.com (recommended to me) but they're kind of hard to get ahold of. Any other suggestions? This is for a business, so reliability is key. I did see the recent thread about this, and while I saw a few mentioned, I didn't see anything about how reliable the different vendors are, or whether people are using them for business or personal use. Thanks! Jen ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared
Just because you cannot get it to work does not mean that IT does not work. Just using the right motherboard is not enough. Did you check for IRQ problems? You don't mention whether you have checked for this. Look for a thread called "Asterisk-Users Small office setupusing analog lines w Asterisk" in the archive via Google. use site:lists.digium.com Try all the things listed in that thread. Do you have a network that is capable of VoIP? Are you using hubs when you should be using switches? There is a major difference and hubs WILL NOT work reliably with VoIP. Are you using QoS on your switches if you have lots of network traffic? If you are using your own Distro and installing from scratch, try to use Asterisk at Home just to see if you still have the same problem. I am putting my money on an IRQ issue myself. W From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of canuck15Sent: Wednesday, August 24, 2005 1:38 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared I came into this with my eyes wide open. I have read ABSOLUTELY EVERYTHING there is to be found on the net about avoiding echo problems BEFORE I even attempted to create a production system. Since lots of people are apparently using this in production environments now I just assumed that echo IS avoidable. As others have recommended, I created a test system with the proposed production parts. I bought a couple different SIP phones to try and a Digium TDM01B card. I am using an older PIII 1Ghz system with 815chipset (PCI Rev2.2) with 256MBfor my test system. The only thing that will be different on a production system is that I will be using a newer chipset PC with faster processor and 512MB. Probably Intel 7505, 7210, or 7211chipsets which seem to be the most compatible with Asterisk. My problem is that I cannot eliminate echo no matter what I try. I seriously doubt that a newer chipset faster PC with more memory will eliminate or even reduce my echo problems based on what I have read.I am not about to drop more cash to try and find out. Essentially, my findingsare that Asterisk is NOT production capable for my configuration which is via FXO and PSTN. That is probably THE most common configuration so if itis not production capable like that itisn't production capable period as far as I'm concerned. What a disappointment :(. Unless I am missing something I am sure that many many people with a similar configuration in a production environment have the same problem. Perhaps they are just living with it?? For me it is just as unacceptable on an Asterisk system as it is on a traditional PBX. Some calls are ok and some are not. No correlation to local, long distance, time of day. There always seems to be some echo. Sometimes it is worse than other times. Again, no correlation to local, long distance, time of day. Tried connecting to ATA adapter and using VoIP provider instead to see if the telco was causing the problem. That did not change anything. Still the same general echo problem The things I have tried includein no particular order and not limited to are: *Buy latest TDM400P withlatest FXO module *Ensurecopper connection to analog telco lines and telco are not causing problems including running a separate shielded line to the demarc AND having the telco guy come out and test the levels, impedance etc. *Adjust RX/TX levels as per Asterisk Wiki using the quick Ztmonitor method and by using the detailed Ztmonitor method via a Telco 102milliwatt test phone #. The end result was RX=8.0, TX=-1.0. Since I still have echo problems I have tried all sort of other settings without success. *After ALL of the above, try every possible combination of all of the following on Asterisk v1.0.9: echocancel (off, on, 128, 256, 16, 32, 64), echowhenbridged (on, off), echotraining (off, on, 800), Mark 2(default, aggressive,CVS head developments, bugs.digium.com patches, adjust threshold level as per wiki etc. etc.) *Make sure echotraining line is before FXO channel assignment in zapata.conf file *Run fxotune which did not find a need to adjust the FXO levels (1=0,0,0,0,0,0,0,0) Based on all the above testing the best settings were pretty much in line with what most people are finding. echocancel=on. echowhenbridged=on, echotraining=800, Mark 2 echo canceller, aggressive cancellation OFF, bugs.digium.com #2820 patch, RX=8.0, TX=-1.0. Still have echo. Aggressive mode helps a bit but then the other persons voice get's cut offa lotespecially when I talkand the cutting in and out of the canceller is more noticeable and objectionableingeneral thanif Aggressive is turned off. Ihave two SIP phones. An Aastra 9133i and a Grandstream GXP2000. Echo problem is the same on both phones. I am located within a metropolitan area in Canada. Any comments and/or suggestions would be greatly appreciated as I am pretty
RE: [Asterisk-Users] Small office setup/using analog lines w/ Asterisk
What good does RAID give you on writes? None whatsoever. RAID only helps performance on reading. Come again? Writing to multiple hard drives in parallel is way faster than writing the same file to one HDD. You should Google the words RAID and Write Performance. I assume you must have meant certain RAID levels are better than others. If that was your meaning then you would be correct. Cheers, Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.0.7 won't run after upgrade to FC4
Did you recompile everything * after your upgrade? W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Stahl Sent: Monday, August 22, 2005 10:32 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk 1.0.7 won't run after upgrade to FC4 I just upgraded to Fedora Core 4 and Asterisk won't run any more. When launching asterisk, I get asterisk: error while loading shared libraries: libssl.so.4: cannot open shared object file: No such file or directory. A quick search (find / -name libssl.so.4) for the file shows the file nowhere on my system. However, when I yum provides libssl.so.4, yum tells me that openssl contains the file I want, and that openssl097a.i386 is ALREADY INSTALLED! What now? How do I get libssl.so.4 if the providing package is already installed? Help! -Bob- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CRM software
Title: CRM software Go look at the Asterisk @ Home install to see how they got Sugar CRM integrated. It is a good start point and you can build from there. W From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee ArcherSent: Thursday, August 18, 2005 8:29 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] CRM software Can anyone recommend CRM software with a link into Asterisk? I would like a pop up on caller ID if possible. I've played with the FOP and SugarCRM but can't get them working together. Regards Lee ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IP Cop as a firewall and QOS
There are a dozen Linux based methods ranging from. Personally I like the Mandrake offering called Multi-Network Firewall. It is pretty turnkey and they have it available for download. It also supports bonding which allows you to use multiple nics bonded together and views as one connection. http://www.mandriva.com/business/mnf2 Other than that, like I said, there are dozens... W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo Jojo Sent: Wednesday, August 17, 2005 3:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IP Cop as a firewall and QOS We are looking for a good firewall replacement which will basically do pot blocking and QOS. Our current solution just plain stinks.. We basically need to handle the traffic of a few web servers, mail server and asterisk box. The most traffic this device will need to handle is what can be shoved through a T1. I don't mind buying an appliance to get something solid but IP Cop just looks better than he appliances I see out there. I am only concerned if it is stable for a production environment. It says it's designed for a SOHO environment, we are doing a bit more than that. Will this thing hold up? Can it be trusted? Anyone using this for QOS and Asterisk in a production setup. Any thoughts or suggestions or warnings would be appreciated! Thanks! -- Start Your Own Internet Service! http://www.YourOwnISP.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How many TDM22P Card can be used on the same PC ?
Just Google the archive on 'IRQ issues'. You can pretty much bet that 6 TDM cards on 6 PCI slots would suck hugely. Unless echo is your goal, you are not going to be pleased. If you have to use 24 existing POTS lines, look into a channel bank and interface it to a T1 card. If you are planning new, just get a PRI T1 and be done with it. Cheers, W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, August 17, 2005 4:38 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How many TDM22P Card can be used on the same PC ? Is it possible to use 24 FXS/FXO lines(on 6 PCI slots) at the same time on the same PC? I wonder for sound quality and power issues. Can anyone convince me that I can(not) use 6 TDM22P cards? Thanks in advance. BDM. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Plantronics USB Headsets Audio 45
I use a DSP 500 and I love it. Great sound, good price. IaxComm is hands down the best softphone I have found. As you can guess it is for IAX though... Cheers, W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Monday, August 15, 2005 10:20 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Plantronics USB Headsets Audio 45 Anybody using Plantronics USB headsets? What softphone are you using and whats your overall experience? Any comments/suggestions? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] intel 875P chipset ok?
I think the easiest way to tell if you don't get an answer is to see if it uses IRQ sharing and if it allows you to assign IRQs individually. A check of the BIOS instructions for that Mobo should be available at the manufacturer. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robbie Hughes Sent: Tuesday, August 16, 2005 3:49 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] intel 875P chipset ok? Does anyone know if the te110p would have any problems running on one of these chipsets? Need new server quickly and the acer altos g310 boxes look relatively good... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TAFM
Also check out this getting started page http://www.oneunified.net/support/asterisk/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kris Edwards Sent: Tuesday, August 16, 2005 9:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TAFM Many of your questions have most likely already been answered either on this list or on the wiki http://www.voip-info.org. Might want to check there if you're just looking for a basic overview of how things work and the various config files. On 8/16/05, Il Neofita [EMAIL PROTECTED] wrote: Hi, I installed this program but I am not able to configure, it does not want to work. Someone can help me? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ita erat quando hic adveni ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Advice on old polycom ip 500
And no RJ45 connectors? Doesn't sound like an IP phone at all. Sure you did not get a phone for a Polycom PBX solution of some sort? W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Tuesday, August 16, 2005 9:38 AM To: Asterisk-Users Subject: [Asterisk-Users] Advice on old polycom ip 500 I have some IP 500s that I bought used, but the connectors are different than the new ones. There is a Modem/Power RJ11, a Line RJ11, and Handset and headset connector. Does anyone know how they work? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell Poweredge 1400
Alejandro... Go search the archive... There are tons of posts regarding Dell equipment Here is how to do so if you do not know... Go to www.google.com Enter the following... site:lists.digium.com Dell Poweredge Thanks, W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Acosta Sent: Monday, August 15, 2005 12:07 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Dell Poweredge 1400 I think this email got mixed with other emails thks. Hi all, In this moment I have the opportunity to install asterisk in Poweredge 1400 Dell Server (PIII, 2 GB of RAM). I wonder if any of you have any experience running asterisk (+ Digium cards) on this kind of hardware, any comment about know problems or good experiences are welcome. Thanks in advance. Alejandro Acosta,- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Firewall will definatelyincrease jittersinyourvoice conversation
Typically a hardware firewall is specialized and uses ASICs. Because the solution utilizes specialized chips tailored to the task, this is considered a hardware based solution. Of course software is involved but it too is specialized and is even proprietary in nature. A software firewall, be it BlackICE or even a Linux on PC uses no specialized hardware. Thus the software designation. It runs on pretty much any x86 hardware (Linux at least) and is not proprietary in nature. That is the general meaning when people say hardware or software firewall. Sure, both technically use some form of hardware and software. But the specialization of that hardware is what makes it designated as hardware based or software based. There have been countless arguments over firewalls in the software vs. hardware arena. At this point and time, I can say I feel that both have great purpose and functionality. I prefer my Pix because I use VPN tunnels to certain sites that have Cisco on the other side and it makes things easier. The configuration of my firewall is also very simplified with my Pix. I ran a Linux firewall for quite a while and I loved it. With the amount of power available to the modern (or even somewhat outdated) PC, you can leverage plenty of performance out of a marginal box. So, to each there own! Use what works best for you application. Great points on single entry point being easier BTW. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Travers Sent: Saturday, August 13, 2005 3:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Firewall will definatelyincrease jittersinyourvoice conversation Wiley Siler wrote: The question was not can I secure a Linux box without a hardware firewall. The question (or statement really) was will a firewall add jitter and lower performance. A good firewall architecture w/QoS will actually prevent jitter and increase performance, I might add. That answer is obviously a big NO. Can you secure a Linux (or even Windows) machine by closing ports? Sure. It helps immensely. However, an advantage of hardware is that you are physically separating the traffic from the end point. The analogy I would use here is that you could purchase a safe for each person in your house and have them each keep all their valuables in it, but it is often cheaper and easier to focus on securing entrence-points. The same is doubly true for office buildings, and also quite true for computer networks. I typically use used P1's running Linux for firewalls. They work great and have all the capabilities I need including QoS and secure management. Sure, all the ports closed on a Linux box can protect that machine. However, having only web (for example) traffic going to your Apache server is really beneficial. The server can focus on delivering pages and not spend any CPU cycles on is this a good packet? Should I drop it?. A firewall (software or hardware) should also be able to better deal with DOS and things of that nature. Port securing does nothing to assist with DOS. DOS doesn't include a TCP/IP stack does it? ;-) By Things of that nature are you including CP/M? Actually port securing can provide some measure of protection against DoS attacks in that fewer services are available to attack. However, you are correct that this protection is probably insignificant. So... You are totally right, you can secure a box that way. However, a firewall (be it software or hardware) is far superior a method. When you say software or hardware I assume you mean hardware like PIX and software like BlackIce. I am not sure where a stripped down Linux version running on a P1 which does firewalling and only firewalling fits in. I call that type of system a hardware firewall simply because it is a dedicated piece of hardware which does perimiter control and only perimiter control. Where VOIP is concerned, use a dedicated firewall system with QoS capabilities. Period. (Yes it is possible to run such a system on Windows, but I certainly don't advise it.) I prefer the hardware method myself as it is a matter of management and additional features. However, for some, a software method may be better. I ran Mandrake SNF (a shorewall implementation) for a long time so I have been there. Considering you can run a Linux firewall on a 386 machine worth $20 makes the fact that so many people don't have firewalls seem just ridiculous. Bear in mind that finding replacement parts (NIC's etc) for your 386 may not be trivial. That is why I use P1's with PCI slots... Also it is often impossible to get OpenGK to compile on such a machine due to memory limitations (my P1 firewall even has this problem and it has a whopping 32MB RAM). So the older you go, the less functionality you may be able to add. Best Wishes, Chris Travers Metatron Technology
RE: [Asterisk-Users] Firewall will definatelyincreasejitters inyourvoice conversation
Do you mean this occurs when traffic is passed over an IPSec tunnel or that it occurs anytime a tunnel is use on a machine that also is passing VoIP traffic (outside the tunnel)? I assume you must mean over the tunnel but I am curious... Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Connolly Sent: Saturday, August 13, 2005 3:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Firewall will definatelyincreasejitters inyourvoice conversation On that note... IPSec tunnels seem to reek havoc on the echo canceling/training process. Anytime our Cisco PIX loads up, the echo complaints start coming in. Stay away from the IPSec tunnels. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Travers Sent: Saturday, August 13, 2005 5:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Firewall will definately increasejitters inyourvoice conversation Rich Adamson wrote: That's a crack of crap sold by the marketing (not sales) people selling firewalls. If you know what you're doing, one can very easily secure any linux system to function on the Internet (etc) without a firewall. It all depends on your level of knowledge/skills on how to disable those items that are not really needed in your environment. Start with a 'netstat -a' to identify those ports that are listening, and shut those items down that you don't want exposed. You can do the same for any MS system as well. But you still want a firewall here especially if you have several VOIP systems which could be making independent connections to the internet. The firewall in this case will hopefully not only do things like VPN for securing your data in trasit between your office and a remote one, but it will also provide a platform for QoS/traffic shaping. To avoid the firewall here is actually *asking* for sound quality problems in addition to the fact that you no longer have the entrence point to your network secured. Now to your point Almost any Linux system can be configured (if you know what you are doing) to perform all these firewalling functions. Just add an extra network card, put it on the perimeter of your network, set up iptables, traffic shaping, uninstall unnecessary software, use Netstat to doublecheck listening ports, etc. and you have your firewall. A firewall doesn't have to be expensive but some form of perimiter control is very helpful in these cases. Best Wishes, Chris Travers Metatron Technology Consulting ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Firewall will definately increasejittersinyourvoice conversation
Yes all firewalls are software running on a piece of hardware. Pretty semantic though. Not all hardware is create equal though. As long as there is a firewall, then all is well. That is the point regardless of what flavor you like. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Esben Stien Sent: Thursday, August 11, 2005 5:31 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Firewall will definately increasejittersinyourvoice conversation Wiley Siler [EMAIL PROTECTED] writes: firewall (be it software or hardware) It's software in the hardware. I prefer the hardware method myself as it is a matter of management and additional features. I think you will look long for a dedicated filter module that has more features than netfilter;). Using netfilter is way more powerful, in my opinion. Some dedicated filter rack modules ship with linux/netfilter now, though. -- Esben Stien is [EMAIL PROTECTED] s a http://www. s tn m irc://irc. b - i . e/%23contact [sip|iax]: e e jid:b0ef@n n ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Firewall will definately increase jittersinyourvoice conversation
The question was not can I secure a Linux box without a hardware firewall. The question (or statement really) was will a firewall add jitter and lower performance. That answer is obviously a big NO. Can you secure a Linux (or even Windows) machine by closing ports? Sure. It helps immensely. However, an advantage of hardware is that you are physically separating the traffic from the end point. Sure, all the ports closed on a Linux box can protect that machine. However, having only web (for example) traffic going to your Apache server is really beneficial. The server can focus on delivering pages and not spend any CPU cycles on is this a good packet? Should I drop it?. A firewall (software or hardware) should also be able to better deal with DOS and things of that nature. Port securing does nothing to assist with DOS. So... You are totally right, you can secure a box that way. However, a firewall (be it software or hardware) is far superior a method. I prefer the hardware method myself as it is a matter of management and additional features. However, for some, a software method may be better. I ran Mandrake SNF (a shorewall implementation) for a long time so I have been there. Considering you can run a Linux firewall on a 386 machine worth $20 makes the fact that so many people don't have firewalls seem just ridiculous. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, August 10, 2005 8:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Firewall will definately increase jittersinyourvoice conversation That's a crack of crap sold by the marketing (not sales) people selling firewalls. If you know what you're doing, one can very easily secure any linux system to function on the Internet (etc) without a firewall. It all depends on your level of knowledge/skills on how to disable those items that are not really needed in your environment. Start with a 'netstat -a' to identify those ports that are listening, and shut those items down that you don't want exposed. You can do the same for any MS system as well. Wiley is definitely right. It would be dangerous not to have a firewall for security reasons. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Wednesday, August 10, 2005 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Firewall will definately increase jitters inyourvoice conversation Lokesh, While adding a firewall may add a tiny bit of latency (non-noticeable by the way) it in no way means you are gonna get jitter. An over utilized data line might cause that but a firewall in and of itself will not. I use a Pix to route my VoIP to an ITSP and I could not be happier. To say that using a firewall causes high latency is incorrect. Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lokesh kumar Sent: Wednesday, August 10, 2005 10:57 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Firewall will definately increase jitters in yourvoice conversation Hi, If you will put firewall, then i think you will get high latency and consequently you will hear voice jitter in your conversation. so avoid putting firewall. Regards Lokesh Portugal mail [EMAIL PROTECTED] Send a rakhi to your brother, buy gifts and win attractive prizes. Log on to http://in.promos.yahoo.com/rakhi/index.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com
RE: [Asterisk-Users] will a firewall slow down asterisk?
That should not be a problem. My users conference using a voip line from an ITSP so at any time there may be 4-8 calls passing over the firewall and terminating in the MeetMe conference. It works great. I would recommend Pix BTW. Linksys would be my next rec. But hey, they are both Cisco now... 8) Cheers, W From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven LangleySent: Wednesday, August 10, 2005 2:12 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] will a firewall slow down asterisk? Hi there I am in the process of setting up a production Asterisk server, which will mainly be used for meetme conferencing. I am considering running a firewall, but wondering whether this will slow Asterisk down if all packets are being scanned. Any ideas? Many thanks Steven Langley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Firewall will definately increase jitters in yourvoice conversation
Lokesh, While adding a firewall may add a tiny bit of latency (non-noticeable by the way) it in no way means you are gonna get jitter. An over utilized data line might cause that but a firewall in and of itself will not. I use a Pix to route my VoIP to an ITSP and I could not be happier. To say that using a firewall causes high latency is incorrect. Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lokesh kumar Sent: Wednesday, August 10, 2005 10:57 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Firewall will definately increase jitters in yourvoice conversation Hi, If you will put firewall, then i think you will get high latency and consequently you will hear voice jitter in your conversation. so avoid putting firewall. Regards Lokesh Portugal mail [EMAIL PROTECTED] Send a rakhi to your brother, buy gifts and win attractive prizes. Log on to http://in.promos.yahoo.com/rakhi/index.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Firewall will definately increase jittersinyourvoice conversation
Absolutely. Lokesh, I suggest you go to the Wiki and check out the security issues inherint in the implementation of SIP in Asterisk. http://voip-info.org/tiki-index.php?page=Asterisk%20security http://voip-info.org/tiki-index.php?page=Asterisk+security+dialplan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Wednesday, August 10, 2005 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Firewall will definately increase jittersinyourvoice conversation Wiley is definitely right. It would be dangerous not to have a firewall for security reasons. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Wednesday, August 10, 2005 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Firewall will definately increase jitters inyourvoice conversation Lokesh, While adding a firewall may add a tiny bit of latency (non-noticeable by the way) it in no way means you are gonna get jitter. An over utilized data line might cause that but a firewall in and of itself will not. I use a Pix to route my VoIP to an ITSP and I could not be happier. To say that using a firewall causes high latency is incorrect. Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lokesh kumar Sent: Wednesday, August 10, 2005 10:57 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Firewall will definately increase jitters in yourvoice conversation Hi, If you will put firewall, then i think you will get high latency and consequently you will hear voice jitter in your conversation. so avoid putting firewall. Regards Lokesh Portugal mail [EMAIL PROTECTED] Send a rakhi to your brother, buy gifts and win attractive prizes. Log on to http://in.promos.yahoo.com/rakhi/index.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] does SIP works behind the NAT
Go to the wiki and search on SIP and NAT www.voip-info.org -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jonny hashem Sent: Wednesday, August 10, 2005 1:24 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] does SIP works behind the NAT i tried to connect 2 iax servers and it worked well but when i tried to connect 2 Sip servers in the iax configuration (one behind the NAT and the other with real IP) it failed and give me this: Aug 10 23:25:47 WARNING[28013]: chan_sip.c:843 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Response) i think it is a NAT problem but i want to know how to fix it. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] First PRI
Excellent info everyone. Thank you!! W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jj Sent: Tuesday, August 09, 2005 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] First PRI Use NI2 anytime it is availabel. It will deliver calling name. NI1 will only deliver calling number. Also most COs will support NI2 with no tweaks much better than NI1 or any of the others. NI1 was created to solve configuration issues between systems. did a pretty good job. But as new features were added and more knowledge gained NI2 came into being. If memory serves there is a 3 coming/ rumored?? On Aug 9, 2005, at 9:25 AM, Tom Hayden wrote: They let you chose your protocol? Nice guys, I've never been asked - just told. I don't know any major advantages between the different signalling formats, though, I don't think there really are any major differences. I've had no problems with ni1 and ni2 with Asterisk. -- Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net On 8/9/05, Wiley Siler [EMAIL PROTECTED] wrote: Hello All, I am getting my first PRI installed in a couple of weeks and I wanted to ask for a little advice. I have a single span Digium card I will be using for the install. Id there a benefit to which protocol I use? When asked, I told them to set it up as NI2. The PRI is through MCI and will be used for local and long distance with DIDs and features like CallerID, etc. Any advice would be appreciated. Thanks! Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] First PRI
Title: First PRI Hello All, I am getting my first PRI installed in a couple of weeks and I wanted to ask for a little advice. I have a single span Digium card I will be using for the install. Id there a benefit to which protocol I use? When asked, I told them to set it up as NI2. The PRI is through MCI and will be used for local and long distance with DIDs and features like CallerID, etc. Any advice would be appreciated. Thanks! Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk - Firewall/Nat - Internet - Firewall/Nat - Softphone/hardphone
Switch to IAXCOMM and use an IAX extension. Problem solved. W From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin KronstadSent: Friday, August 05, 2005 7:03 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] Asterisk - Firewall/Nat - Internet - Firewall/Nat - Softphone/hardphone Hi! The bandwith is not the problem, uploadspeed is about 400 kbits. I think I found the solution, I need to have a Proxy in the middle, or set up a IAX2 client and server at each end I will be testng this next week. BR Martin Kronstad What is the upload speed on B? Looks to me as you have bandwidth problem! Martin Kronstad wrote: Hi! Problem: I can_t hear what the people at Location B i saying, they hear me but I do not hear them. They can call, I can call. Just no sound. My current setup is: Softphones/Hardphones(Location A) - Asterisk - Firewall/Nat - Internet - Firewall/Nat - Softphone/hardphone(Location B) I am having problems with sound, I have opened the following ports: Location A: 10 000 - 20 000 (TCP and UDP) 5060 (TCP and UDP) 8000 (TCP and UDP) Location B: 8000 (TCP and UDP) 5060 (TCP and UDP) I am using [EMAIL PROTECTED] 1.3 , and xlite as softphone. I have tried to set the softphone I have set the extention parameters(in sip.conf) to: ;; Location A [200] username=200 type=friend secret=1234 record_out=On-Demand record_in=On-Demand qualify=no port=5060 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid="Location A" 200 ;; Location B [201] username=201 type=friend secret=1234 record_out=On-Demand record_in=On-Demand qualify=no port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid="Location B" 201 My sip.conf : port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) externip=80.202.50.16 disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown language=no Best Regard Martin Kronstad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Full T38 sip Faxing now Available
Caveat Emptor Considering how he been as a list participant, I would be wary but it is your dime... Hope it works out... W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Wednesday, August 03, 2005 1:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Full T38 sip Faxing now Available As I understand it, you sent non-Asterisk-related commercial announcement to the Asterisk Users' mail list. What made you think that that wouldn't be considered to be Spam? I obviously can't comment on your service, as I'm unlikely to become a customer. I think you are being a bit tough here. I think the message was well meaning and well intended. I won't exclude his services just for that, especially as I need T.38 faxing to complete our installation. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Making a call on Asterisk... new thread or not?
You may want to try a little research here... www.voip-info.org www.digium.com Google: site:lists.digium.com asterisk process -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Karl Sent: Tuesday, August 02, 2005 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Making a call on Asterisk... new thread or not? Hello, Does anyone know how Asterisk manages calls on a system? More specifically, does it spawn a thread off of the asterisk program... are they separate processes? We're trying to see what kind of system load the PBX will create when calls are put through. Thanks, Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WHat does it take
It would not hurt for you to realize that this is the Asterisk list and not the Asterisk @ Home forum. AAH is a specifically configured turn key product that someone was nice enough to package for people who dont want to hand code their configs. Thusly, it is not really something that people on this list (most at least) bother with. You need to go to the AAH website on source forge and find the link to the forum. That being said, I will throw you a bone. Assuming you have all 4 ports for that card, did you try to make concurrent outgoing calls? Did one fail while the other works just fine? AAH at one point came pre-configured for one line, not 4 so you had to adjust your zaptel.conf lines. Make sure you have the latest version. And that your channels are there. As to AMP, there is a link to the documentation on the AAH site. Read it. You are probably not setting up your trunks right. Info on Asterisk that can be found at the Wiki. www.voip-info.org. Cheers, Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim King Sent: Tuesday, August 02, 2005 8:29 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] WHat does it take How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re: [Asterisk-Users] WHat does it take
Go into the CLI on the box and type:
sip show users
sip show peers
Did you get two lists? One that shows the sip accounts and the other
that shows the registered sip accounts?
What does this show in the CLI: zap show channels
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim King
Sent: Tuesday, August 02, 2005 2:54 PM
To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: Re: [Asterisk-Users] WHat does it take
The busy extensions are from dialing any local extensions from one to
another. I cant seem to post the configs because I used asterisk at home
and
the post becomes too big. I have ZAP and SIP extensions configured and
no
matter what they always transfer straight to voicemail.
Extensions.conf
; Asterisk Management Portal (AMP)
; Copyright (C) 2004 Coalescent Systems Inc
; dialparties.agi (http://www.sprackett.com/asterisk/)
; Asterisk::AGI (http://asterisk.gnuinter.net/)
; gsm
(http://www.ibiblio.org/pub/Linux/utils/compress/!INDEX.short.html)
; loligo sounds (http://www.loligo.com/asterisk/sounds/)
; mpg123 (http://voip-info.org/wiki-Asterisk+config+musiconhold.conf)
; include extension contexts generated from AMP
#include extensions_additional.conf
; Customizations to this dialplan should be made in
extensions_custom.conf
; See extensions_custom.conf.sample for an example
#include extensions_custom.conf
[from-trunk]; just
an
alias since VoIP shouldn't be called PSTN
include = from-pstn
[from-pstn]
include = from-pstn-custom ; create this context in
extensions_custom.conf to include customizations
include = ext-did
include = from-pstn-timecheck ; this has to be included
otherwise
it overrides ext-did
[from-pstn-timecheck]
exten = .,1,Goto(s,1) ; catch-all matching for calls that have
DID
info (if a DID route hasn't matched them)
exten = s,1,GotoIf($[${IN_OVERRIDE} =
forcereghours]?from-pstn-reghours,s,1:)
exten = s,2,GotoIf($[${IN_OVERRIDE} =
forceafthours]?from-pstn-afthours,s,1:)
exten =
s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1:)
exten = s,4,Goto(from-pstn-afthours,s,1)
[from-pstn-reghours]
exten = s,1,GotoIf($[${FAX_RX} =
disabled]?from-pstn-reghours-nofax,s,1:2)
; if fax detection is disabled, then jump to from-pstn-nofax - else
continue
exten = s,2,Answer
exten = s,3,Wait(1)
exten = s,4,SetVar(intype=${INCOMING})
exten = s,5,Cut(intype=intype,-,1)
exten = s,6,GotoIf($[${intype} = EXT]?7:9) ; If INCOMING
starts
with EXT, then assume its an extension
exten = s,7,Wait(3)
;wait 3 more second to make sure this isn't a fax before dialing someone
exten = s,8,Goto(ext-local,${INCOMING:4},1)
exten = s,9,GotoIf($[${intype} = GRP]?10:12) ; If INCOMING starts
with
GRP, then assume its a ring group
exten = s,10,Wait(3)
exten = s,11,Goto(ext-group,${INCOMING:4},1)
exten = s,12,GotoIf($[${intype} = QUE]?13:15)
exten = s,13,Wait(3)
exten = s,14,Goto(ext-queues,${INCOMING:4},1)
exten = s,15,Goto(${INCOMING},s,1) ; not EXT or GR1
-
it's an auto attendant
exten = fax,1,Goto(ext-fax,in_fax,1)
exten = h,1,Hangup
[from-pstn-reghours-nofax]
exten = s,1,SetVar(intype=${INCOMING})
exten = s,2,Cut(intype=intype,-,1)
exten = s,3,GotoIf($[${intype} = EXT]?4:5) ; If INCOMING
starts
with EXT, then assume its an extension
exten = s,4,Goto(ext-local,${INCOMING:4},1)
exten = s,5,GotoIf($[${intype} = GRP]?6:7) ; If INCOMING starts
with
GRP, then assume its a ring group
exten = s,6,Goto(ext-group,${INCOMING:4},1)
exten = s,7,GotoIf($[${intype} = QUE]?8:11) ;queue
exten = s,8,Answer
;
answer call before queue
exten = s,9,Wait(1)
exten = s,10,Goto(ext-queues,${INCOMING:4},1)
exten = s,11,Answer
;
answer call before auto attendant
exten = s,12,Wait(1)
exten = s,13,Goto(${INCOMING},s,1) ; not
EXT or
GR1 - it's an auto attendant
exten = fax,1,Goto(ext-fax,in_fax,1)
exten = h,1,Hangup
[from-pstn-afthours]
exten = s,1,GotoIf($[${FAX_RX} =
disabled]?from-pstn-afthours-nofax,s,1:2)
; if fax detection is disabled, then jump to from-pstn-nofax - else
continue
exten = s,2,Answer
exten = s,3,Wait(1)
exten = s,4,SetVar(intype=${AFTER_INCOMING})
exten = s,5,Cut(intype=intype,-,1)
exten = s,6,GotoIf($[${intype} = EXT]?7:9) ; If INCOMING
starts
with EXT, then assume its an extension
exten = s,7,Wait(3)
;wait 3 more second to make sure this isn't a fax before dialing someone
exten = s,8,Goto(ext-local,${AFTER_INCOMING:4},1)
exten = s,9,GotoIf($[${intype} = GRP]?10:12) ; If INCOMING starts
with
GRP, then assume its a ring group
exten = s,10,Wait(3)
exten = s,11,Goto(ext-group,${AFTER_INCOMING:4},1)
exten = s,12,GotoIf($[${intype} = QUE]?13:15)
exten = s,13,Wait(3)
exten = s,14,Goto(ext-queues,${AFTER_INCOMING:4},1)
exten = s,15,Goto(${AFTER_INCOMING},s,1) ;
RE: [Asterisk-Users] [Asterisk-Dev] Asterisk 1.2 Release Plans
Something very very different. AAH is a package of Asterisk (1.0.7 I think) and AMP and FOP and other tools... Asterisk is the core software that runs in AAH. So, Asterisk is the REAL software nuts and bolts while AAH is a nice packaging of tools with Asterisk as the core. You should go to digium.com and asterisk.org. It would surely benefit your overall understanding of the product... Hope that helps... W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Stone Sent: Wednesday, July 27, 2005 4:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] [Asterisk-Dev] Asterisk 1.2 Release Plans The current version up for download is 1.3 how does that mesh with a potential release of 1.2, in the future, when 1.3 is out on http://asteriskathome.sourceforge.net already? Or is the asteriskathome project something different? On Wed, 2005-07-27 at 23:18 +0200, TWV wrote: What are all these astonishing new features and improvements? Can you please give us an overview? Thanks! -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Kevin P. Fleming Verzonden: dinsdag 26 juli 2005 18:15 Aan: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Developers Mailing List Onderwerp: [Asterisk-Users] [Asterisk-Dev] Asterisk 1.2 Release Plans As previously mentioned on the lists by Olle Johannson, we are actively trying to get Asterisk in shape for a 1.2 release within the next 60 days. To accomplish this, we need a few things to happen: 1) A feature freeze - This will occur at the end of this month, with no new feature submissions accepted after July 31st. Any _pending_ feature patches in Mantis that have passed architecture review and functionality testing before August 1st can be accepted into 1.2, if they make it through the remainder of the review processes and are able to be merged before August 15th. 2) Progress on open bugs - There are a number of bugs open in Mantis that are waiting for the poster to provide additional information, test results, call traces, etc. We would much prefer to not release 1.2 with suspected problems already identified, but we cannot solve them without adequate input from you. If you have an open bug and are not in a position to continue providing assistance in solving it, please post a message to the mailing lists asking for volunteers to help replicate the problem so it can get resolved. 3) Testing - We need a _lot_ of help testing. If you have not previously tested CVS HEAD, please download it, read the UPGRADE.txt file and install it on one or more systems to play around with. Please do _not_ put it into a production environment unless you are willing to accept the consequences of that action. If you do find a bug or other issue, when you open a bug in Mantis, please try to provide _all_ the configuration information, call traces, etc. that the bug guidelines request, so that we don't waste 3-4 days just going back and forth requesting more information from you. If possible, join the #asterisk or #asterisk-dev IRC channel to find out exactly what debugging information will be required and how to produce it, if you don't already have that knowledge. 4) Release Candidates - I will produce the first release candidate on August 20th, with followup versions produced every week until we deem the release ready for public consumption. I expect it will require at least three -RC releases for us to get things in shape, so that means that 1.2 itself may be ready by September 15th. We are very thankful for the community's help and support, and we want Asterisk 1.2 to be as important a release as 1.0 itself was. The number of new features, performance improvements, bug fixes and interoperability enhancements in CVS HEAD is astonishing, and a very large percentage of them came directly from community contributions. We hope that all of the 'non-developers' in the community will be able to help us 'shake out' the bugs and problems remaining in the code, so we can be assured of the most stable 1.2 release possible :-) ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
RE: [Asterisk-Users] Opteron Hardware with Asterisk
Did you build it using the 64 bit CentOS or another Distro? Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Dobrin Sent: Friday, July 22, 2005 4:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Opteron Hardware with Asterisk I have asterisk running on dual 244's. Everything works fine, the only special issue i had was installing the g729a codec (required a very tiny tweak to the asterisk Maiefile). Unfortunately, the system doesn't get a huge amount of traffic, so I can't testify to capacity. Running 1.0.8, btw. Asterisk Supporter wrote: Anyone running Asterisk on dual Opteron Server? Are there any special issues in a 64 bit environment and what is the capacity curve like? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1 - incomplete calls
Do some debug on the calls and see what you get. W From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JOAO CARLOS MOURA Sent: Thursday, July 21, 2005 2:56 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] T1 - incomplete calls Hi All Help. We are using a T1 with Paetec Telecom in the Miami area, with a Digium card into our Asterisk software, and in the last week we are experience a large quantities of incomplete calls, even local and international, what do you think, the problem are into the T1 or into our configuration? Here our configuration Zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us === Zapata.conf [channels] language=en signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=200 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=000 busydetect=yes busycount=5 group=1 callgroup=1 pickupgroup=1 callreturn=yes context=pstn channel = 1-23 Thank you João Carlos Moura NiNeTel Telecommunications 7382 N.W. 35 Terrace Miami, FL 33122 USA João Carlos Moura NiNeTel Telecommunications 7382 N.W. 35 Terrace Miami, FL 33122 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Free Music
For the fella who wanted MOH music. Royalty free stuff can be found here.. The Acoustic Guitar is a nice collection http://www.freeplaymusic.com/ Cheers, W ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] So you all think VoIP sypply is warm and fuzzy
Right, so before resolution can be attempted or had, you lash out on the forum. If they had told you to stuff it or had just ignored you, you might have something to complain about. You are pissed that the ATA is not web configurable? How in the hell is that VoipSupplys fault? You bought it! You specifically set out to bash VoipSupply with your comments, most of which seem ill informed and a couple just asinine. I personally hope no one answers your questions. Your complaint was hardly to hear from others about VoipSupply. That could have been accomplished with a simple Google search of the list archive. As it is, this is just really unnecessary and testament to your lack of patience. So take a valium and let the vendor try and fix the problem. Then tell your story here. W From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael D Schelin Sent: Monday, July 18, 2005 5:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] So you all think VoIP sypply is warm and fuzzy I was waiting for everyone to reply so here is mine.. Check out the Mediatrix web site. There are no downloads or lists of resellers who might have this provisioning software that is normally included with purchase. You may be right that it is a refurb but every indication points that it is not. I have contacted both companies and I'm waiting for replys. I'm on the west coast and it took over 7 days to get here. I am a little pissed when all other ATA's are configurable from their built in web server. And Yes, I'm self serving as well as mostly everybody I've ran into in this business. This unit was purchased for testing. Because of the timezone problem, When I get the product from UPS it's too late to call Canada or FL. when all I need is a simple download to correct the problem. Is it too much to expect everything in the box when you purchase it? Or have a web site with these free included software so if this happens we don't wast our valuable time. By the way I did get an email from VOip Supply asking me to wait until morning so they could find the software. This is at 2:30 PST. This complaint was to hear from others about VoIP Supply and their business practices. I wanted to get feedback ether way, or maybe a contact name so I can get this paper weight working and tested. Has anyone used the 2102? Please let me know. Michael D. Schelin Shelltel JD Austin wrote: Michael D Schelin wrote: Here is a letter I sent them for my $150 paper weight. Dear Voipsupply, As a small service provider, using you company for the first time, I'm very disappointed that you have removed the configuration CD that should have been shipped with the Mediatrix 2102 just to get a few more bucks. I have contacted mediatrix and they have informed me that the CD's is shipped in every 2102. If I don't here back from you shortly and receive the configuration program that should have shipped, I will return it back to you for a full refund and express my views to the Voip community. As of now I've herd of nothing but good things about your customer support. I've called and left messages to your support team. I waited 7 days for this unit and have no way to configure it. Email me the CD. Michael D. Schelin Owner Shelltel Are you sure you didn't buy a refurbished model? I hear they sell a lot of refurbished equiptment, I've purchased some of it myself. Everything I've purchased from them worked without issue. None however came with an installation CD. A few things had to be reset to clear settings though. Anything I needed was freely available. Since you know how to contact Medatrix, perhaps you can download the software or get a CD from them. JD -- JD AustinTwin Geckos Technology Services LLCemail: [EMAIL PROTECTED]http://www.twingeckos.comphone/fax: 480.288.8195 ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best VoIP provider
Hmmm.. My nufone account is still running although it had problems yesterday. Seshu, try contacting Jeremy at nufone dot com. I think that is his email at least. Last name should be Macnamera (sp?) I think. You can search the archive for his name along with nufone.com if you need other contact info. I am sure it is there somewhere. Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Tuesday, July 19, 2005 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Best VoIP provider It does not look like Nufone is still in business, judging from the content on their site, which is very little. There is not even a configuration document to download, to connect to their network. The rates file is only for US/Canada calling. No international rates on this rates.csv file. I have signed up with a $5.00 account with them way back in November 2004. After signup, I havent received any email or anything of that sort, explaining to me how to connect to their network. The only email address I see on their site is [EMAIL PROTECTED], there is no support related contact information on the site, which does not inspire much confidence. Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Tuesday, July 19, 2005 12:33 PM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Best VoIP provider That's odd -- they used to be here: http://www.nufone.net/rates.csv Of course, you can't rely on that. -Original Message- From: Chris Mason (Lists) [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 19, 2005 6:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Best VoIP provider Madhawa Jayanath wrote: o Bernie, 1) best results www.nufone.net 2) low cost www.voipjet.com Anyone able to find NuFone's rates? I have been looking for them on their site. I need international rates and UK Mobile. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Free Music for MOH from Digium?
freeplay.com i think. i will vverify for the url tomorrow at work. the acoustic guitar stuff is nice... Cheers, W From: [EMAIL PROTECTED] on behalf of Jim Archer Sent: Tue 7/19/2005 5:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Free Music for MOH from Digium? Hi All... I installed the Debian Sarge Asterisk package and in the docs it had the licensing terms for the MOH, explaing that Digium (or someone) had licensed the mucic for distribution as MOH only. That's fine, but I can't find the music! Does anyone know where it can be found? Is there another source of free MOH that sounds good with Asterisk? Thanks... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Teliax to VoIPJet
Use to providers for the call, pay two providers for the call. You have two call legs so you are using two channels bridged at your * box. You will have to pay for those to legs... W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of code select Sent: Monday, July 18, 2005 6:34 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Teliax to VoIPJet 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. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Teliax to VoIPJet
This sounds like DISA which is great for saving bucks on LD if used right... You will still need two channels and thus it will still cost for both legs... Nature of the beast... W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Monday, July 18, 2005 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Teliax to VoIPJet HUH? Why? If you are having Cellphones dialed for the user its one thing but what is the goal On 7/18/05, code select [EMAIL PROTECTED] wrote: 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. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codecs and bandwidth
I assume ISDN accomplishes this since the PRI is set to use channel 24 for signaling. Your 64K channels is data and the control overhead is sent on the signaling channel. Actually, everything I have seen is around 80K full duplex for a uLaw channel with overhead. That is point to point... W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Pushor Sent: Monday, July 18, 2005 9:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Codecs and bandwidth Hi Friends, Something I'd like to shed some light on if possible - how is it that a single ISDN conversation only uses 64K for bidirectional communication (using ulaw, correct?), but on several occasions now have seen references to ulaw voip conversations using 64K per side of the conversation, plus packet overhead (http://www.zytrax.com/tech/protocols/voip_rates.htm - seems to be down now - plus other references) for a total of over 128K per ulaw 'full duplex' voice conversation? Thanks Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I ne ed?
Let me expand on the bandwidth point HTH made and maybe shed light on your requirements A 100baseT switched (no hubs) network has a lot of bandwidth when you think in terms of VoIP. The uLaw stream (uncompressed) from an IP500 phone to the Asterisk box is not going to take more than 80K of bandwith from the bandwidth pool. That means 60 phones ALL in a single call would only be using around 5 megs of throughput. At that point packet scheduling becomes far more important than bandwidth. Gigabit is nice but the value of QoS in comparison is very evident. If cost becomes a driving factor, you may want to focus on upgrading port count and remain at 100baseT instead of going to Gigabit. A properly configured 100baseT network with good QoS rules will yield great performance over an unregulated 100baseT network. Do you know your real traffic needs? I would check how much traffic is via user download, www browsing, streaming, email, etc, etc... You may find that some simple rules save you quite a bit of cash. Just a thought and alternative... Gigabit is also very tempting so that whole spiel may have been for not. 8) Also, pay heed to the PoE stuff you are hearing about. I may be wrong but I am pretty sure you want to be careful what you connect to a PoE port. Otherwise you wind up with fried PoE injectors and end devices. I believe PoE ports would only be used for a PoE phone in essence. Just as a reminder and warning. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Thursday, July 14, 2005 8:47 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I ne ed? But currently, I only have one ethernet jack per office. Routing another 60 or so ports would add a very substantial expense in both cabling and backbone expansion (what category ethernet is required, BTW?). Most decent phones have an ethernet passthrough (2 port) so you can plug in your PC. As long as your LAN is decent (Cat5 100baseT switched) the overhead using VoIP is negligible. I have used the 3Com NJ wall jacks with good success: http://www.3com.com/products/en_US/detail.jsp?tab=featurespathtype=purc hase sku=3CNJ90 It's basically a 4 port switch that you replace your wall jack with. I used the NJ200, it allows you to set priority per port, although I think they are discontinued now. In combination with a 3Com power over Ethernet injector, I was able to expand a 24 port LAN to a 96 port LAN with a per-port cost of $62 Cdn. And, 24 ports of those 96 are PoE, so I can plug my phones right in to port 1 and they power up, no external power supply needed. hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need?
Depends on what you mean by expanding your network. Do you need a bunch of new routers? Probably not. Do you need to consider port count at every station? Absolutely. However, there is good news and bad news. The good news is that most of the phones that are being recommended to you actually have a pass through port on them so you can connect your PC right to the phone. So, in a situation where there is one port and one user, you have no problem adding the phone to the scenario. You just connect the phone to the wall and connect the PC to the switch port on the phone. The bad news is that is does create a small amount of cable clutter and it does limit the speed of the PC in question to 100Mbit. So if you go this route, don't run out and get all new Gigi NICs for those PCs. They would just get pushed back down to 100Mbit once you connect them to the phone port. In situations where you have many users in a room and limited ports, the cost effective method will be to just add a switch in that room. Make sure it is a real switch though. Hubs are VoIP performance killers. If you can get new pulls in place then do so, the benefit long term is there. You might frame it as an upgrade to the 4 wire Cat-3 stuff you were asking about. Cat5 is sufficient for most cases and supports up to 100Mbit. Go to Cat5E (Cat Five Big E as it is known) which has a 350MHz frequency and you can support Gigi over copper. I would check around for bids on the cabling. $60 a drop is very reasonable and would put you at $3600 in cabling. A worthwhile endeavor if you can slide it into the budget. If not, cautious use of local switches can accomplish the task. Again, look for QoS capable switches or at the minimum CoS (Class of Service). I doubt your Cat-3 will be of any practical use to you going into the future. I would have them pull it out when they install the new drops or tuck it out of the way. I assume your comdial phones are proprietary and connected to your Comdial PBX. That being the case, they probably would not be reusable (see the Wiki). However, you may be able to sell them off to get s few dollars back on your upgrade. Cheers, Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need?
No idea on the phone ports but I doubt it as 100Mbit is sufficient and the parts are cheap for the makers of the phones. Not a bad switch but since you get 4 ports (one is used for connect to wall) you may want to just up for the 8 port unless you know only two people will use each switch. You won't regret it. I also prefer Linksys for most small switches but that decision is mostly a matter of preference and features. Look out for duplex issues, make sure you get FULL. Also beware the hub as stated before. For most Gigabit parts it seems to be a non-worry but just to throw that out there. Adding switches like that only offer a problem if you consider backplane speed. The backplane of a Netgear 5 port is not going to be as high as that of a 24 or 40 port HP. As example, I have a 100baseT HP 40 port that has a 9 gig backplane so you can see how port count is important compared to backplane bandwidth. But since the port count is lower on those switches, there really is less contention for the bandwidth in most cases. The fact that you picked a switch that honors packet priority seems like a good step. Other than that, as long as you are not daisy chaining these (serial) then all should be well. Connect one port of you main switch to one end user switch and only one switch. Never string one room to the next room to the next room, etc, etc. You may get a little bottle neck if two users on an end switch were both pulling from an internal resource but I doubt your users will notice. Sorry if any of that is just way to obvious and you already knew them. I just like to throw those things out there just in case. Cheers, Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need?
You clipped the original so there are some other things that need to be known. How many users are being supported again? The biggest hits to servers seems to be due to transcoding in most cases. Look on the wiki for an explanation of server sizing and decide based upon how you will connect your users to dial tone. A good bet is to figure on a dual processor machine with 1 GB (or 2 is better at current RAM costs) of RAM, and at least 80GB HDD. You may want something that has drive redundancy via RAID (most would I think) so factor that into the cost. How many phones do you need? What features? Great phones can be had for $150-200. Do you need one line LED only? Get the Polycom IP300. Need multi-line? Get the Polycom IP501. (best value in my opinion but others are great too) How many ports for your Gigabit network? How many replacement NICs? How many switches at what size? I assume you can price that yourself? Do you really even need this? I think moving to Gigibit is great but you may want to make sure you focus more on QoS enabled switches. Plenty come in Gigabit capable and with high port density. Linksys has some nice managed switch at 24 ports with gigabit and QoS. How many lines do you need to have dialing at once? Do you need 40 hard lines? Do you need 23 (PRI T1)? Figure a PRI T1 at $600-800 per month (may be different in your region) Software phones are pretty much free at this point. USB Headsets are around $50 for good ones from Logitech or Plantronics (my fave). PCI cards are pretty much decided like this. From what I have seen, Sangoma cards require more technical savvy than Digium cards so plan on getting some Digium cards unless you want to deal with the learning curve. Others may disagree but that is my opinion. Examples of the card costs can be seen at voipsupply.com and other places. They make multiport cards that support up to 4 T1s on one card if you need more than 23 channels of voice. I think most of the pricing is really something you can do yourself if you just answer the questions above and go from there. I don't think anyone here will be able to give you a budget number. You will need to start a tally sheet and go from there. Post your numbers when you are done and I bet someone will double check them for you. Until then, you need do a little more of the calculation on your own. You don't need to understand telecom to get this budget completed. You just need to know what your basic requirements are. Answer the questions above and you will know all of your hardware and line provisioning. Factor in an additional 10% for cost overrun and you should be good. Don't forget to add any consulting fees you feel you may need (figure a couple grand at worst). Hope that helps! Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Pastore Sent: Wednesday, July 13, 2005 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need? Thanks for all the great replies. I guess I over-asked my question (since so many kept popping up). For now, what I really need to determine is what I need to budget for a full implementation. Unfortunately, I don't have time now to do testing and analysis... I just need to get my budget submitted. So I'm trying to figure out what all I'll need to buy and budget for. Obviously this is pretty hard, since I understand so little about telecom. So that said... Can anyone help me in determining what all I will need? The only thing I really need is one ballpark figure for a grand total cash outlay. However, it it is too low, I may be hosed. If it is too big, the project may be cut out of the budget. So I'd like to get within, say $5K of the actual expected cost. The items I had identified in my original post were: - A server, running Debian Linux or OS X (our preferred operating systems here) - A good network. We're on switched 100 Base-T, but will move to gigabit next year. - A T1 or some dedicated channels of a T1 - Gateway PCI cards or devices (in the case of OS X, only devices I guess) - VOIP phones or phone software (I'd like to use software and USB handsets) Are there more things I need? Or does someone have a rough estimate of what it costs to implement an Asterisk system in a small business? We have about 50 users and currently have something like 20 POTS lines coming into our PBX. Thanks again. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options
RE: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need?
I think one thing you may want to remember is that porting numbers to a VoIP provider can make them EXTREMELY hard to ever port back to a normal telco provider. Also, if there is ever a problem with the VoIP provider (which has been common lately) then you are in deep trouble. For a mission critical install, it is highly recommended that you get land lines pulled in via whatever means meets your needs and that you use VoIP providers only as a backup to your hardwired system. Saving bucks on long distance is great but betting the farm on a zystem that provides dial tone from purely VoIP can be highly dangerous. My $0.02 would be to get a PRI (or 3 or whatever you need) and use VoIP for cost savings on LD as applicable and as a backup to PRI failure. If you want a purely VoIP solution, you should go to an Avaya or other hosted VoIP PBX solution because the model for Asterisk doesn't really support hosted services. Asterisk IS the host of your services. You just need to connect it to all the correct systems. Again, just my $0.02. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sahil Gupta Sent: Wednesday, July 13, 2005 11:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need? Why not look at getting a provider that can port your numbers to their network and buying the DID's off them over VoIP? Regards, Sahil Gupta VoiceValley On Wed, 13 Jul 2005, Ed Pastore wrote: Thanks for all the great replies. I guess I over-asked my question (since so many kept popping up). For now, what I really need to determine is what I need to budget for a full implementation. Unfortunately, I don't have time now to do testing and analysis... I just need to get my budget submitted. So I'm trying to figure out what all I'll need to buy and budget for. Obviously this is pretty hard, since I understand so little about telecom. So that said... Can anyone help me in determining what all I will need? The only thing I really need is one ballpark figure for a grand total cash outlay. However, it it is too low, I may be hosed. If it is too big, the project may be cut out of the budget. So I'd like to get within, say $5K of the actual expected cost. The items I had identified in my original post were: - A server, running Debian Linux or OS X (our preferred operating systems here) - A good network. We're on switched 100 Base-T, but will move to gigabit next year. - A T1 or some dedicated channels of a T1 - Gateway PCI cards or devices (in the case of OS X, only devices I guess) - VOIP phones or phone software (I'd like to use software and USB handsets) Are there more things I need? Or does someone have a rough estimate of what it costs to implement an Asterisk system in a small business? We have about 50 users and currently have something like 20 POTS lines coming into our PBX. Thanks again. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need?
Sounds like a PRI T1 will be fine for you to start with. It offers you 23 voice channels (one channel is used for signaling). That means you can get a single Digium T1 card for around $600 or you can get a quad T1 card for around $220 (with echo cancellation). If there is no move to expand, then get the regular T1 card and save some cash. New equipment is always coming out so by the time you are ready to expand there may be something new out. Example: http://www.voipsupply.com/product_info.php?cPath=99_103products_id=415 The Voice T1 card is and isn't like a normal T1. I am sure you are thinking T1 = Internet. Well, T1 can equal phone channels to. In this case, your PRI is delivering voice channels so the card will be your server's interface to the telco side. Once the card is connected to the PRI T1, Asterisk will take care of routing any calls it receives from your internal users. It will receive the calls from the internal users via the NIC in the server. The NIC acts as the user side of this whole shebang. Phones or SIP/IAX devices on the same LAN as the NIC of the server will be able to connect to the NIC via standard TC/IP address you assign when you build the server. Protocols are taken care of automatically, just config the box accordingly. Dial plans allow users connected to your PBX to route without the user knowing squat. The system knows that number 18001234567 should go to the PRI and it routes it such. In essence, aside from being an application server (voicemail, IVR, etc) Asterisk is also a router. That being said, it also means that an * box can split data and voice if configured properly. Some T1 providers will offer you a split T1 that has 512K data for instance along with 15 voice channels (1 channel for signaling) or whatever permutation of the bandwidth you choose. For ease of use, I would recommend you stick to just voice over your T1. So the path of a call looks like this User Handset -- Your Network -- NIC on * PBX -- Dial Plan on PBX Parses -- Sends Call Out of the PRI via the T1 Card Regarding remote access, there are several ways. You can allow VPN into your network then your users can connect just like they were local. You can use IAX protocol devices like the IAXy to connect them with an adapter and a hard phone. Generally speaking, if you expose a SIP port to the internet (security caution BTW) then you can have your users connect from anywhere. Just remember that there are security issues. VPN is the best method for security. Your numbers will come from whomever your get your PRI from. MCI or anyone like that can offer you something. DIDs are really really cheap so don't worry too much about that. Just tell your salesman how many you want. T1 lines are Digital. That says it all. Better quality of sound (usually) and more features with more control. Keeping your 20 lines is an option of course. You would need a channel bank and a T1 card. The channel bank would accept the analog POTS lines and allow you to connect your * server to it via a T1 interface. So that would be. POTS Lines -- Channel Bank -- T1 Card on Asterisk Box Example: http://www.voipsupply.com/product_info.php?products_id=922 There are PCI cards for POTS but they only support 4 lines per card. By the time you get to 20 lines your server will be in IRQ hell. Better not to deal with it. The channel bank is a viable solution but like all other solutions it comes at a cost for hardware. Ease of setup and the fact that you do not have to wait 30 days for a T1 install make it a nice option for some though. You can have as many DIDs as you want on a digital system line the PRI. It works like so. There are 23 channels. No channel is tied to any one particular phone number. The information on what number is being called is passed to the PRI which passes it along to the T1 card. So the channel number in use is irrelevant. All we need to know is who is calling and for what number. The T1 card passes the data to Asterisk which uses its dialing rules to decide who gets the call. Maybe the number in question is support so you send it to a queue. Maybe it is for someone's direct line so you sent it to their desk. The options are pretty endless. The only catch is this 50 DIDs does not equal 50 calls at once. Something to remember. Only 23 of the 50 DIDs could ever possibly be in use at once. Equally important, how you set your hunt groups upstream will matter when it comes to line usage. If you get a lot of calls on an 800 number, setting all 23 lines as huntable would leave you with no outward dialing if you got really busy. That being the case, you would set a hunt of 20 for the 800 number for instance and leave 3 out. That way 3 lines would be available for your DID pool or for someone to make an outgoing call. BTW - I am in no way a telco expert so if I made a mistake, someone on the list is sure to jump on it and correct me. They always do...
RE: [Asterisk-Users] Any suggestions for an IP phone?
Great points but I think the ease of config on Polycom via FTP along with the ease up firmware updates is a real winning combination. I have yet to need the kind diagnostics you refer to while troubleshooting. I copy a valid config, change the values as needed and load it to the FTP server. Boot the phone and we have tone. No tone? Check the config file and that is about it. For the average Joe, that would suffice. At least for me anyways... Cheers, W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, July 13, 2005 12:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Any suggestions for an IP phone? On Jul 13, 2005, at 8:44 AM, dbruce wrote: Feature for feature, the polycom phones are much better than the cisco phones. Presence: Polycom = yes, Cisco = no Messaging: Polycom = yes, Cisco = no Microbrowser: Polycom = yes (on IP600, xml), cisco = yes (on 7940 and 7960 - cmxml - much harder to program) Auto-Answer: Polycom = yes (via configuration files and Alert-Info header), cisco = yes (through a manually configured second line) Call Appearances: Polcom = yes (up to 8 per configured line), cisco = yes (up to 2 per configured line) Ringtones: Polcom = yes, Cisco = yes Upgradable Firmware: Polcom = yes (simple procedure), Cisco = yes (not so simple, sometimes very complex procedure Sound Quality: Polycom = excellent, Cisco = excellent Configuration: Polycom = very comprehensive, Cisco = basic Price: Polycom = $300USD (IP600), Cisco = $320USD(7960) First we are a Polycom shop and have been very happy with them. But one area which is lacking that Cisco is MUCH better at is diagnostics. By telnetting into a Cisco you have access to everything on the phone. You can also view layer 2 info and much more which you cannot do on a Polycom. When troubleshooting this is vtial information to have. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need?
Hello and welcome... Most of what you want to know is available on the wiki located here... http://voip-info.org/tiki-index.php Just scroll down to the All Things Voip section. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Pastore Sent: Tuesday, July 12, 2005 2:33 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Systems Admin; Telecom Newbie - What do I need? Hi, folks. I am planning on implementing Asterisk in 2006, and need to budget for it now, so I need to know what I'll need to get. My company has about 50 users, and is currently languishing on a very old Comdial PBX. All of our client computers are Macs; our servers are mostly OS X, with a couple Debians and a Red Hat. I am thoroughly experienced at systems administration, and can figure out most everything I need on the computer hardware and software side, but I am a complete telecom newbie and get lost when trying to figure out what else I will need. Here's what I think I need: - A server, running Debian Linux or OS X (our preferred operating systems here) - A good network. We're on switched 100 Base-T, but will move to gigabit next year. - A T1 or some dedicated channels of a T1 - Gateway PCI cards or devices (in the case of OS X, only devices I guess) - VOIP phones or phone software (I'd like to use software and USB handsets) Here's what I don't get: 1. How do I route between the internet and the telco network? (I said I was a telecom newbie, right?) I mean, if someone dials a phone number, what tells it to route to my gateway device? Do I need service from a telecom company? I need to get the phone numbers from somewhere, right? 2. Does my network need to be VOIP capable? I see some network switches which route additional layers of ethernet, including in some cases VOIP. Do I really need that? Or will any gigabit switches do the trick? If so, what's up with those VOIP switches? Is that just marketing? They sure cost a lot more. 3. What do I need in a T1? I currently have one T1 from Sprint, going into a Cisco router, which then goes to my firewall, then to my network. If I want, say 30 channels of another T1 for VOIP... can I just buy another Sprint T1? And where does additional hardware fit into that route in order to split out the VOIP channels from the data channels? 4. Do I pretty much need a vendor for implementation help, if this is all new to me? Or is there a path I can follow that will help me get through this? 5. What am I not asking that I should be? :) Any help, input, suggestions, etc. would be welcome. (But please no vendor calls yet... I'm in early budgeting, and will just ignore vendor input until I know more.) Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Question about Polycom SoundPoint 500
Read directly off of one of my phones power supplies... AC Adaptor I.T.E Power Supply Model: AD41-1200400DU Input: 120VAC 60Hz 200mA Output: 12VDC 400mA P/N: GJE-AD41-995 LEVEL 3 Outer ring is negative, inner is positive There should be more in the manual for that phone I assume Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Jones Sent: Monday, July 11, 2005 3:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Question about Polycom SoundPoint 500 Hi Folks; I just bought a Polycom SoundPoint 500 off of ebay after having spent way too much time trying to get updated sip images for our cisco phones. The phone I bought didn't have an AC power adapter; Could someone please tell me the volts amps that the dc plug that comes with the phone puts out? Thanks! Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Epia C3 Linux
Yep, along with 6 other distros. W From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JD Austin Sent: Tuesday, July 05, 2005 5:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Epia C3 Linux Tried knoppix? Wiley Siler wrote: OK. Something is truly rotten in Denmark. I took the 2.5 inch drive out altogether and setup a regular 3.5 IDE drive with a CDROM. BIOS recognizes both. Try to install Redhat 9, it dies. Fedora Core 3 dies, kernel panic. How in Zeus Red Ripe Ass did you guys get this to install? Am I going to have to make a custom kernel? To recap This is a Via Mini-ITX board 800MHz Samuel 2 Processor AKA E-Series C3 (not Eden, this one has a fan) Thanks to all, Wiley PS. AstLinux bombed too From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Wiley Siler Sent: Tuesday, July 05, 2005 4:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have attempted FC3, RedHat 9, Mandriva 10, Gentoo 2005.0, Gentoo 2004.3, Debian Mini, Debian woody, and Windows XP. Nothing will install. All see the HDD. All attempt partitioning (XPO seemingly completes), none will install the OS. BIOS posts the correct HDD and all the installers see the HDD. All bomb out immediately after attempting to partition with the exception of Gentoo. The LIVECD will allow me to set a partition table but it dies when I attempt to apply filesystem ext3 to the root partition. I am officially stumped. Thanks for all the input everyone! Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux It installed directly from the FC3 dvd, no changes...no external drivers required From: Wiley Siler [mailto:[EMAIL PROTECTED]] Sent: Friday, July 01, 2005 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux Did it require any special work or did you just download the ISO for FC3 and install? Thanks, Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have Fedora Core 3 running great on an Epia mobo From: Wiley Siler [mailto:[EMAIL PROTECTED]] Sent: Friday, July 01, 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Epia C3 Linux Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD AustinTwin Geckos Technology Services LLCemail: [EMAIL PROTECTED]http://www.twingeckos.comphone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Epia C3 Linux
Rob, How in the world did you know that I just ran the memtest86 and it is nothing but error after error. Switched out the ram and I am getting no errors on memtest86 now. I am back in the saddle. Fedora Core 3 is installing as we speak Thank you! Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Thomas Sent: Tuesday, July 05, 2005 6:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux Sounds to me like bad RAM. Try running memtest (your Fedora CD has it, just type memtest at the cd boot prompt) --Rob From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Wednesday, 6 July 2005 10:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux OK. Something is truly rotten in Denmark. I took the 2.5 inch drive out altogether and setup a regular 3.5 IDE drive with a CDROM. BIOS recognizes both. Try to install Redhat 9, it dies. Fedora Core 3 dies, kernel panic. How in Zeus Red Ripe Ass did you guys get this to install? Am I going to have to make a custom kernel? To recap This is a Via Mini-ITX board 800MHz Samuel 2 Processor AKA E-Series C3 (not Eden, this one has a fan) Thanks to all, Wiley PS. AstLinux bombed too From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Tuesday, July 05, 2005 4:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have attempted FC3, RedHat 9, Mandriva 10, Gentoo 2005.0, Gentoo 2004.3, Debian Mini, Debian woody, and Windows XP. Nothing will install. All see the HDD. All attempt partitioning (XPO seemingly completes), none will install the OS. BIOS posts the correct HDD and all the installers see the HDD. All bomb out immediately after attempting to partition with the exception of Gentoo. The LIVECD will allow me to set a partition table but it dies when I attempt to apply filesystem ext3 to the root partition. I am officially stumped. Thanks for all the input everyone! Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux It installed directly from the FC3 dvd, no changes...no external drivers required From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux Did it require any special work or did you just download the ISO for FC3 and install? Thanks, Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have Fedora Core 3 running great on an Epia mobo From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Epia C3 Linux Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Epia C3 Linux
This did wind up being a matter of memory... Thanks, Wiley W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Wednesday, July 06, 2005 10:14 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Epia C3 Linux On Tue, Jul 05, 2005 at 08:39:15PM -0400, Michael Stahl wrote: Take a look at the via arena web site. Your processor may look like a 586 to the installer but may not support all of the instructions (causing a crash). The via arena site gives instructions on how to compile and get it installed on your processor! (I have the C3 Nehemiah processor so I didn't need to recompile) You'd expect it to blow up with Illegal instruction then and not with a segfault. If you fear this may be a 386 issue, get the Debian Sarge netinst. It has only i386 kernel. Or try current Rapid, which will also give you an Asterisk installation. But my suspect here is the memory: have you tried memtest? a number of of installers and live-cds now come with it as a boot option. Also note that most installers have a shell available on an alternative terminal (usually console no. 2). It used to be very limited, but the one on current debian (sarge) installer is actually quite usable and even has tab completion for path names (thanks busybox). -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Epia C3 Linux
I have attempted FC3, RedHat 9, Mandriva 10, Gentoo 2005.0, Gentoo 2004.3, Debian Mini, Debian woody, and Windows XP. Nothing will install. All see the HDD. All attempt partitioning (XPO seemingly completes), none will install the OS. BIOS posts the correct HDD and all the installers see the HDD. All bomb out immediately after attempting to partition with the exception of Gentoo. The LIVECD will allow me to set a partition table but it dies when I attempt to apply filesystem ext3 to the root partition. I am officially stumped. Thanks for all the input everyone! Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux It installed directly from the FC3 dvd, no changes...no external drivers required From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux Did it require any special work or did you just download the ISO for FC3 and install? Thanks, Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have Fedora Core 3 running great on an Epia mobo From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Epia C3 Linux Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Epia C3 Linux
OH, yes, the error is always Segementation Fault when I try to write the ext3. W From: Wiley Siler Sent: Tuesday, July 05, 2005 4:53 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Epia C3 Linux I have attempted FC3, RedHat 9, Mandriva 10, Gentoo 2005.0, Gentoo 2004.3, Debian Mini, Debian woody, and Windows XP. Nothing will install. All see the HDD. All attempt partitioning (XPO seemingly completes), none will install the OS. BIOS posts the correct HDD and all the installers see the HDD. All bomb out immediately after attempting to partition with the exception of Gentoo. The LIVECD will allow me to set a partition table but it dies when I attempt to apply filesystem ext3 to the root partition. I am officially stumped. Thanks for all the input everyone! Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux It installed directly from the FC3 dvd, no changes...no external drivers required From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux Did it require any special work or did you just download the ISO for FC3 and install? Thanks, Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have Fedora Core 3 running great on an Epia mobo From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Epia C3 Linux Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Epia C3 Linux
OK. Something is truly rotten in Denmark. I took the 2.5 inch drive out altogether and setup a regular 3.5 IDE drive with a CDROM. BIOS recognizes both. Try to install Redhat 9, it dies. Fedora Core 3 dies, kernel panic. How in Zeus Red Ripe Ass did you guys get this to install? Am I going to have to make a custom kernel? To recap This is a Via Mini-ITX board 800MHz Samuel 2 Processor AKA E-Series C3 (not Eden, this one has a fan) Thanks to all, Wiley PS. AstLinux bombed too From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Tuesday, July 05, 2005 4:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have attempted FC3, RedHat 9, Mandriva 10, Gentoo 2005.0, Gentoo 2004.3, Debian Mini, Debian woody, and Windows XP. Nothing will install. All see the HDD. All attempt partitioning (XPO seemingly completes), none will install the OS. BIOS posts the correct HDD and all the installers see the HDD. All bomb out immediately after attempting to partition with the exception of Gentoo. The LIVECD will allow me to set a partition table but it dies when I attempt to apply filesystem ext3 to the root partition. I am officially stumped. Thanks for all the input everyone! Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux It installed directly from the FC3 dvd, no changes...no external drivers required From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux Did it require any special work or did you just download the ISO for FC3 and install? Thanks, Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have Fedora Core 3 running great on an Epia mobo From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Epia C3 Linux Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Epia C3 Linux
Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Epia C3 Linux
Oliver, Thanks for the response! Do you know where I can find an example of how to do this? I have never had to install a custom kernel before. Thanks! Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oliver Rath Sent: Friday, July 01, 2005 10:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Epia C3 Linux Wiley Siler wrote: Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? You have to compile without mmx and sse, best 586compatible, because linux is recognizing C3 as PIII, what is definitly wrong. Hth, Oliver ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Epia C3 Linux
Did it require any special work or did you just download the ISO for FC3 and install? Thanks, Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have Fedora Core 3 running great on an Epia mobo From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Epia C3 Linux Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Epia C3 Linux
I just tried Fedora Core CD1 and it died on autopartitioning. W From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux It installed directly from the FC3 dvd, no changes...no external drivers required From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux Did it require any special work or did you just download the ISO for FC3 and install? Thanks, Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have Fedora Core 3 running great on an Epia mobo From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Epia C3 Linux Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Quality of provider: VocTel
Wow, found the papers out at Broadband. Even more shocking than expected!! Papers located at bottom of page here if anyone wants them: http://www.broadbandreports.com/forum/remark,13748234~mode=flat~days=999 9~start=20 I long wondered what the link between Brandon and Pamela was. These guys are a regular little criminal clan with Pam playing the part of Ma Barker Joop must have been the fast talking boyfriend? He was a snake oil salesman if ever there was one, that is for sure. They should all be prosecuted for fraud but I doubt they will be... They got several people for multi-thousand dollar pre-purchases. Hopefully, those people are smart enough to file a complaint and get the fraud charges rolling... Here is hoping... Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Thursday, June 30, 2005 10:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Quality of provider: VocTel Livevoip was HARDLY big. They had one server total if you read their bankruptcy papers. /b --- Anakin: You're either with me, or you're my enemy. Obi-Wan: Only a Sith could be an absolutist. On Jun 29, 2005, at 10:36 PM, Michael Stahl wrote: Any users of the VocTel VOIP service? (Canadian) How have you found the quality (Choppy / smooth audio)? Any problems registering? (I have been unable to register for hours) After reading about the collapse of a big USA VOIP provider, I'm curious Thanks, OCG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
