Re: [asterisk-users] Console softphone
You can install a browser softphone on the same server and make calls from any browser. Ted - Original Message - From: Julien Claassen To: asterisk users mailinglist Subject: [asterisk-users] Console softphone Date: Thu, 28 Aug 2008 11:16:59 +0200 (CEST) Hello all! Is there a way to (mis)use asterisk itself as a softphone? Can I make a call from within the CLI? Can asterisk from itself produce a ringtone? Or can I bind a system-command to incoming calls? Any help is sincerely appreciated! Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nothing says Labor Day like 500hp of American muscle Visit OnCars.com today. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco vs Asterisk
I'm a CCIE and CCVP. I have worked in the Cisco TSBU on both CCM and Telepresence systems I have two IP patents for the VoiP Lite protocols and have been designing and building OSS IPBXs for companies including Google going back to 2001. I'm not mentioning any of that to be jerk I mentioned it to say I'm as qualified as anyone to to compare the CCM and OSS servers. The only fair way to compare the two is a list of weights features, for example if cost is your biggest feature then OSS is better, if support is your biggest feature than Cisco wins. When a customer is comparing the costly (TCO) and best supported systems in the world with hundreds of thousands installed systems for the large global companies on the planted backed by 54,000 employees and over $25b in the bank vs, a FREE system with one layer of support maybe two layers of support, the features don't even come in the evaluation in my opinion. I once asked a manager why did you buy the CCM and he said no one ever got fired for buying Cisco if anything wrong, If push the OSS and it goes I could loose my job. I would get a list of the important features, because there is no answer to your question of which is better. - Original Message - From: Benoit Plessis To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco vs Asterisk Date: Tue, 22 Jul 2008 15:10:50 +0200 voip crazy a écrit : Hello all, A client of us, is thinking to migrate their actual PBX to a Cisco CallManager. We want to sell him an asterisk box to complement the Cisco PBX. I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) Has asterisk all the functionalities to replace a CIsco Unity server? Which functionalities Cisco Unity has than asterisk could cover? How could asterisk complement the Cisco Call Manager funcionalities? To answer your questions, one would need to know what exactly are all the functionalities of a Cisco Unity server, and more specificaly, what are the needs of your client. But i'm pretty sure the voip-info wiki can answer the asterisk part... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Be Yourself @ mail.com! Choose From 200+ Email Addresses Get a Free Account at www.mail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco vs Asterisk
My son owns compoanyn here in San Jose and when a customers says they want Cisco be provides Cisco phones with OSS PBX, it seems to work the lower cost and Cisco phone on the desktop. - Original Message - From: Steve Totaro To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco vs Asterisk Date: Tue, 22 Jul 2008 08:59:24 -0400 On Tue, Jul 22, 2008 at 8:52 AM, voip crazy wrote: Hello all, A client of us, is thinking to migrate their actual PBX to a Cisco CallManager. We want to sell him an asterisk box to complement the Cisco PBX. I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) Has asterisk all the functionalities to replace a CIsco Unity server? Which functionalities Cisco Unity has than asterisk could cover? How could asterisk complement the Cisco Call Manager funcionalities? Thanks. VoipCrazy. You said migrate to a Cisco, what do they have now? Sell them all Cisco. You will make more money and great residual income for MACs ;-) Anyways, you could ditch the Cisco entirely and use Asterisk. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Be Yourself @ mail.com! Choose From 200+ Email Addresses Get a Free Account at www.mail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Toll Free International Number
voxbone.com - Original Message - From: Larry Costigan To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Toll Free International Number Date: Tue, 15 Jul 2008 14:22:49 -0400 Hello All, I am looking to find a way to provide international toll free access to our Knoxville, TN (USA) office from our customers in the UK and in Australia, and when I talked with ATT I was surprised to find out how expensive they are... Surely, other businesses are not paying this much - are they?!?! Can someone in this good group please help me with some advice as to who can provide affordable and reliable international toll free service for a better price than ATT? Thanks in advance,Larry CostiganFood Donation Connection (Asterisk fan and ABE user) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Be Yourself @ mail.com! Choose From 200+ Email Addresses Get a Free Account at www.mail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distinctive Ring for SIP?
exten = s,1,Wait(1)
exten = s,n,Set(Var_Alert=${SIP_HEADER(Alert-Info)})
exten = s,n,GotoIf($[${Var_Alert} = http://127.0.0.1/Bellcore-dr3]?2,1)
;goto ext 2 if distinctive ring
exten = s,n,Dial(SIP/1)
- Original Message -
From: Steve
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Distinctive Ring for SIP?
Date: Tue, 8 Jul 2008 19:04:31 -0500
Hello,
I've done a bit of research, though obviously in the wrong places. :)
I'm looking to set up some type of distinctive ring for SIP phones
based
on the context in which the SIP phone is being called from. If that
sounds confusing it isn't.
I have two FXOs connected to POTS, so in essence a caller could be
dialing us at one of two phone numbers. Each of those FXOs has its
own
context in extensions.conf. The contexts essentially grab the
incoming
call and ring a bunch of single line SIP phones. If no one answers
then
the call is sent to voicemail depending on the number that the caller
dialed. The problem is that when the SIP phones ring, there's no way
for us humans to tell which number the caller actually dialed.
My initial thought was to make some type of distinctive ring to the
SIP
phones to indicate which number the caller dialed, much in the same
way
that the telco does it when someone has the distintive ring service.
Another idea would be to set the callerid on the SIP phones to
indicate
which line was dialed. The drawback there is that we lose the
incoming
callerid from POTS. Maybe set one of line of the callerid to the
incoming line and leave the other as the callerid number?
Does anyone have experience with setting distinctive ring in SIP in
such
a way?
Steve
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