[asterisk-users] IAX softphone
Hi all, Can some one suggest me an IAX client for Linux and Windows? I used KIAX once, but know it seems complicated to have it working on Ubuntu. Thanks. Ronaldo. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Soft phones.
Hi Ken, Can it be an IAX client? If so, I'd recommend KIAX. I used it once, both on Linux and Windows, and it worked for me. []s Ronaldo. On Thu, Jul 22, 2010 at 4:14 PM, Ken D'Ambrosio k...@jots.org wrote: Hey, all. I'm looking -- if possible -- for a decent, multi-platform soft-phone. Specifically, Linux and Windows; that way, I'll go through the same issues my end users do. I've noticed a couple (e.g., minisip, which seems abandoned, and sip-communicator, which, honestly, is probably a great IM client, but has a confusing interface for actual phone calls). So I'm wondering if anyone has any favorites. Failing multi-platform, I'll stick with Twinkle on Linux, and gladly take suggestions for Windows -- OSS if possible, but payware is acceptable. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Applications
Ok Eric, Thank you again. Ronaldo On 4/14/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Ronaldo Zacarias Afonso wrote: Hi all, I was trying to set up a conference room using the MeetMe application and my asterisk is telling me that there is no MeetMe application available for the extension I've dialed. [Apr 14 20:57:11] WARNING[958] pbx.c: No application 'MeetMe' for extension (internal, 600, 1) So, I issued the command core show applications and it didn't show me a MeetMe application. I'd like to know how I can install that application or what I have to do to make it available for use. Thanks. MeetMe requires Zaptel to be installed. Install Zaptel, then rebuild and install 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
[asterisk-users] Installing Applications
Hi all, I was trying to set up a conference room using the MeetMe application and my asterisk is telling me that there is no MeetMe application available for the extension I've dialed. [Apr 14 20:57:11] WARNING[958] pbx.c: No application 'MeetMe' for extension (internal, 600, 1) So, I issued the command core show applications and it didn't show me a MeetMe application. I'd like to know how I can install that application or what I have to do to make it available for use. Thanks. Ronaldo. ___ --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] SIP: number to names
OK Yuan, What I wanted to know is if the extension I've created is right. exten = 101,1,Dial(SIP/sip:[EMAIL PROTECTED]) Will my asterisk bridge a SIP phone that dialed 101 to the SIP user: [EMAIL PROTECTED] Do I need some think more in order for it to work? Do you have or know any documentation that explains me that? Regards Ronaldo. On 4/13/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Ronaldo Zacarias Afonso [EMAIL PROTECTED] Date: Thu, 12 Apr 2007 11:54:51 -0300 Hi all, Is it possible to configure an extension number to dial a sip address? Nothing prevents you from doing this. Yuan Liu For example: exten = 101,1,Dial(SIP/sip:[EMAIL PROTECTED]) That way I can dial to a sip name using my Hardphone that is not able to dial using names just numbers. Thanks in advance. Ronaldo. (I hope putting my sip address soon here) ___ --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] SIP: number to names
Hi all, Is it possible to configure an extension number to dial a sip address? For example: exten = 101,1,Dial(SIP/sip:[EMAIL PROTECTED]) That way I can dial to a sip name using my Hardphone that is not able to dial using names just numbers. Thanks in advance. Ronaldo. (I hope putting my sip address soon here) ___ --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 and hard phone configuration
Hi, It's really a simple question! I've just started playing with asterisk too, and I think what you want could be found in the 4th chapter of Asterisk: The Future of the Internet. It's a open book you can download from http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11. I hope it'd helped you. Ronaldo. On 4/12/07, Ilya Vishnyakov [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello Asterisk Gurus! I have a very simple question. I've just started playing around with Asterisk and BSD box. I also have grandstream ip phone and installed asterisk from ports. Now I'm on my very first steps to configure Asterisk. The question is: How do I make Asterisk communicate with my Grandstream hard phone? Thank you in advance. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGHovpUZGmaUWxLn8RAn9UAJ94exp6gs2PBWpMDiiNA69Mt78jhgCfYy71 eOq4eOuYi2uDpve+8YM2fp4= =+Jt7 -END PGP SIGNATURE- ___ --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 SIP phones to buy?
Hi Stephen, I'm using Grandstream and I think is a nice phone, but its the only one that I've tried. I bought it to learn voip/asterisk. Just my 2 cents. Good luck. Ronaldo. On 4/11/07, Stephen Bosch [EMAIL PROTECTED] wrote: Stephen Bosch wrote: I need to buy some new phones for our own offices. I've used only Polycom phones until now, but I'd like to broaden my experience. I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I don't want to broaden my knowledge... ...because I like to stay dumb. Of course, that's not what I meant :) -Stephen- ___ --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] Require only GSM Codec
Hi Sanjay, I'm not sure about that, but I think you can configure it in, for example, /etc/asterisk/sip.conf. There is an option that you configure for each channel like: only=gsm It instructs the sip protocol, that only gsm codec must be used. I hope it has helped you. Regards, Ronaldo. On 4/3/07, Sanjay Rajdev [EMAIL PROTECTED] wrote: Hello All, I would like to only use the gsm codec for all the calls, is it possible I want to use minimum possible bandwidth as we have most of calls over Internet. Regards, Sanjay Rajdev ___ --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] Linking incoming calls
Hi all, I just want to know how I can make sure that incoming calls to my asterisk server are being treated by [incoming] section of extension.conf file. Thanks in advance. Ronaldo. ___ --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
