Re: [asterisk-users] SIP Softphone
On 08/06/2014 22:01, Mark Robinson wrote: Hello, can someone recommend a good and free Softphone for Windows which does not display advertisments inside the program? Has anyone tried MicroSIP? http://www.microsip.org/ -- Regards, Giles Coochey, CCNP, CCNA, CCNAS NetSecSpec Ltd +44 (0) 8444 780677 +44 (0) 7983 877438 http://www.coochey.net http://www.netsecspec.co.uk gi...@coochey.net smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] SIP Softphone
On 09-06-14 08:52, Giles Coochey wrote: On 08/06/2014 22:01, Mark Robinson wrote: Hello, can someone recommend a good and free Softphone for Windows which does not display advertisments inside the program? Has anyone tried MicroSIP? http://www.microsip.org/ Nope but if it doesn't meet your needs then maybe have a look at Jitsi https://jitsi.org/ or Linphone https://www.linphone.org/ I prefer the client to have at least the following features: Security: - TLS - SRTP - ZRTP Codecs: - G722 - G729 Fight NAT (if IPv6 is not an option): - STUN - TURN - ICE Cheers, Patrick -- _ -- 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
[asterisk-users] SIP Softphone
Hello, can someone recommend a good and free Softphone for Windows which does not display advertisments inside the program? We have X-Lite but free version display advertisments. thanks. -- _ -- 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] SIP Softphone
Zoiper gsm -Original Message- From: Mark Robinson vsysnetw...@gmail.com Sender: asterisk-users-bounces@lists.digium.comDate: Sun, 8 Jun 2014 17:01:54 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] SIP Softphone -- _ -- 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] SIP Softphone
have you tried zoiper or 3cx? On 9/6/2014 00:01, Mark Robinson wrote: Hello, can someone recommend a good and free Softphone for Windows which does not display advertisments inside the program? We have X-Lite but free version display advertisments. thanks. -- _ -- 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] SIP Softphone
thanks Guys. I like Zoiper. Will try it. On Sun, Jun 8, 2014 at 5:05 PM, binary dreamer.bin...@gmail.com wrote: have you tried zoiper or 3cx? On 9/6/2014 00:01, Mark Robinson wrote: Hello, can someone recommend a good and free Softphone for Windows which does not display advertisments inside the program? We have X-Lite but free version display advertisments. thanks. -- _ -- 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] SIP Softphone
Ekiga works well. William Hetherington w - www.willwh.com t - @wmwh On 8 Jun 2014 14:02, Mark Robinson vsysnetw...@gmail.com wrote: Hello, can someone recommend a good and free Softphone for Windows which does not display advertisments inside the program? We have X-Lite but free version display advertisments. thanks. -- _ -- 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
[asterisk-users] sip softphone for PDA window mobile 2003 / 5.0 ?
are there any good softphone on PDA window mobile 2003 / 5.0 ? tried sjphone, sound quality is unacceptable. Mario ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Softphone on Thinclient?
Is anyone doing this or has anyone tried? The thin clients are running WindowsCE, a browser, and 300mhz. They are Wyse units. I wonder if anyone has any practical advise or can recommend the best phone or method to load a stable softphone on one of these boxes? Thanks, Steve Totaro ___ --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 Softphone on Thinclient?
Hi Steve, We are running X-Lite on Wyse V90 terminals. They have Windows XP Embedded, 800mhz CPU and 512 MB Ram. We use USB headsets because the onboard audio chip is very poor on voice quality. I guess X-Lite has Windows CE version. Check on www.counterpath.com. Idris -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 20, 2006 1:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP Softphone on Thinclient? Is anyone doing this or has anyone tried? The thin clients are running WindowsCE, a browser, and 300mhz. They are Wyse units. I wonder if anyone has any practical advise or can recommend the best phone or method to load a stable softphone on one of these boxes? Thanks, Steve Totaro ___ --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] SIP Softphone on Thinclient?
I currently use NTAVO thin clients w/ Thinstation and I would love to put a soft phone on them, but I don't think that would work well (they use RDP), or do you all know if there is a smooth way to make the interface work? I don't really picture my users switching between an RDP session X-Windows (i.e. ALT-F3/ALT-F4) On 6/20/06, Idris AVCI [EMAIL PROTECTED] wrote: Hi Steve, We are running X-Lite on Wyse V90 terminals. They have Windows XP Embedded, 800mhz CPU and 512 MB Ram. We use USB headsets because the onboard audio chip is very poor on voice quality. I guess X-Lite has Windows CE version. Check on www.counterpath.com. Idris -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 20, 2006 1:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP Softphone on Thinclient? Is anyone doing this or has anyone tried? The thin clients are running WindowsCE, a browser, and 300mhz. They are Wyse units. I wonder if anyone has any practical advise or can recommend the best phone or method to load a stable softphone on one of these boxes? Thanks, Steve Totaro ___ --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] SIP Softphone on Thinclient?
I currently use NTAVO thin clients w/ Thinstation and I would love to put a soft phone on them, but I don't think that would work well (they use RDP), or do you all know if there is a smooth way to make the interface work? I don't really picture my users switching between an RDP session X-Windows (i.e. ALT-F3/ALT-F4) I have compilled for Thinstation softphone named KIAX. Switch beetwen RDP session and softphone doing like ALT-F3/ALT-F4. ___ --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 Softphone on Thinclient?
Steve Totaro wrote: Is anyone doing this or has anyone tried? The thin clients are running WindowsCE, a browser, and 300mhz. They are Wyse units. I wonder if anyone has any practical advise or can recommend the best phone or method to load a stable softphone on one of these boxes? Thanks, Steve Totaro ___ --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 We have both kphone and xlite running on thinterms using LTSP nad running them as a local app, however it uses portaudio with OSS and i have noticed that different audio modules/soundcards give very different audio quality. eg CMIPCI = very good VIX82XX = very poor Bails ___ --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 Softphone on Thinclient?
Vitaly, That is good news, but I'm afraid that switching between screens will be a bit too much for my end users to handle. On 6/20/06, bails [EMAIL PROTECTED] wrote: Steve Totaro wrote: Is anyone doing this or has anyone tried? The thin clients are running WindowsCE, a browser, and 300mhz. They are Wyse units. I wonder if anyone has any practical advise or can recommend the best phone or method to load a stable softphone on one of these boxes? Thanks, Steve Totaro ___ --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 We have both kphone and xlite running on thinterms using LTSP nad running them as a local app, however it uses portaudio with OSS and i have noticed that different audio modules/soundcards give very different audio quality. eg CMIPCI = very good VIX82XX = very poor Bails ___ --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] SIP Softphone on Thinclient?
Steve Totaro a écrit : Is anyone doing this or has anyone tried? The thin clients are running WindowsCE, a browser, and 300mhz. They are Wyse units. I wonder if anyone has any practical advise or can recommend the best phone or method to load a stable softphone on one of these boxes? May I advertise MozIAX (moziax.mozdev.org) ? It is well suited to thin client environment, because the user interface (Firefox extension) and the engine (iax and sound management) communicate through network, so you can run the UI on the server, and the engine on the thin client, and you don't need to run a network sound system on the thin client. I think it gives better sound quality. Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --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 Softphone on Thinclient?
2006/6/20, Jean-Denis Girard [EMAIL PROTECTED]: May I advertise MozIAX (moziax.mozdev.org) ?It is well suited to thin client environment, because the user interface(Firefox extension) and the engine (iax and sound management) communicate through network, so you can run the UI on the server, andthe engine on the thin client, and you don't need to run a network soundsystem on the thin client. I think it gives better sound quality. Jean-Denis, could you elaborate ?Do you mean that :1. you don't need to install anything on your thin client 2. mozIAX is capable to use local sound resourcesCheers ___ --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 Softphone on Thinclient?
Olivier Krief a écrit : 2006/6/20, Jean-Denis Girard [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: May I advertise MozIAX (moziax.mozdev.org http://moziax.mozdev.org) ? It is well suited to thin client environment, because the user interface (Firefox extension) and the engine (iax and sound management) communicate through network, so you can run the UI on the server, and the engine on the thin client, and you don't need to run a network sound system on the thin client. I think it gives better sound quality. Jean-Denis, could you elaborate ? Do you mean that : 1. you don't need to install anything on your thin client 2. mozIAX is capable to use local sound resources Hi Olivier, What I called the engine above is network_client, a small self contained program (written in C, no dependency on whatever libs), which takes care of IAX and sound management directly, thanks to libiaxclient. network_client can run as a local application on the thin client, so it uses local sound system. Then Firefox (and MozIAX, as an extension) is run on the server (display exported to thin client). The FF extension and network_client communicate via network when needed (eg. new call coming in, or user dialing a number). I've been using MozIAX like that for some time in an LTSP environment with a few clients: iax packets and sound are managed locally on the thin client, so sound quality isn't impacted by another layer (sound server). The original question was about a win ce environment: I don't see any reason why it would not work there too. Should you have more questions about moziax, its mailing would be more appropriate: http://moziax.mozdev.org/list.html. Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --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 Softphone on Thinclient?
Martin Joseph a écrit : This looks very interesting to me, and I am interested in an OSX version of it? Is there some way I can help to build that version? Hi Marty, Yes, I'm very interested in having a OSX version ! May I suggest you have a look at the moziax mailing list, available at http://moziax.mozdev.org/list.html, and continue the discussion there. OSX support has been discussed a few times, shouldn't be difficult to build, and I think someone even made it but didn't submit changes back to the project. See http://www.mozdev.org/pipermail/moziax/2005-October/06.html Please do not hesitate to ask more info if needed; I'm really looking forward to have OSX support, so I will try to help. Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --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 Softphone or API which supports QoS (DiffServ/DSCP) needed
Dear All! I need a SIP softphone or API with QoS (DiffServ/DSCP) to integrate it with custom Windows application. Any suggestions? Thanks in advance! Alex ___ --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 softphone with subscription/hint support?
Hi there, for testing purposes I am searching for a freely available softphone that supports SIP subscriptions and display the status of a few of these via e.g. a simulated LED. I know about * EyeBeam (not free) * SNOM softphone (needs Win XP and has old firmware) Are there other softphones with this feature set around (that aren't fixed to one specific VoIP operator)? Philipp ___ --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] SIP SoftPhone for debuging
Hi, I`m looking for SIP SoftPhone for debuging some situations in my SIP VoIP network. Requirements: - it mustn`t be registered with any registrar/proxy/anything - it must be able to send INVITE msgs without registration, so it must accept characters @ and . - it must be able to receive responses from called UAs (if they send some of course) - win32 platform Is it possible ? I tried X-Lite, but i`m not able to send invite messages without registration to registrar and i`m also not able to call other character than digits. Thanks for pointing me to the right direction. -b ___ 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] SIP SoftPhone for debuging
barney wrote: I`m looking for SIP SoftPhone for debuging some situations in my SIP VoIP network. Requirements: - it mustn`t be registered with any registrar/proxy/anything - it must be able to send INVITE msgs without registration, so it must accept characters @ and . - it must be able to receive responses from called UAs (if they send some of course) - win32 platform Is it possible ? I tried X-Lite, but i`m not able to send invite messages without registration to registrar and i`m also not able to call other character than digits. Thanks for pointing me to the right direction. Why not use sipsak? http://sipsak.org/ Regards, Daniel smime.p7s Description: S/MIME Cryptographic Signature ___ 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] SIP Softphone for testing with Asterisk
Also try the snom 360 soft phone emulation: http://snom.com/snom360softphone.html. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of raymond Sent: Thursday, April 07, 2005 6:01 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP Softphone for testing with Asterisk Hi all, I had just set up my asterisk server. Can anybody know that is there any sip softphone for testing with asterisk? (I had download some from internet but I think all are preconfig to certain server). Cheers. Raymond ___ 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] SIP Softphone for testing with Asterisk
Rod Bacon [EMAIL PROTECTED] writes: find firefly one of the best Be aware that firefly is not free software. -- Esben Stien is [EMAIL PROTECTED] http://www. irc://irc./%23contact [sip|iax]: jid:b0ef@ ___ 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] SIP Softphone for testing with Asterisk
Hello Rod, I try firefly but got problem of "Sip registratons failed for network (503)" Is it possible for you to advise me your config? Below is what I put on sip.conf [34169788]type=friendusername=34169788secret=password88host=dynamiccanreinvite=nocontext=internaldisallow=alldtmfmode=rfc2833qualify=4permit=0.0.0.0/0.0.0.0 asterisk1*CLI sip show peersName/username Host Dyn Nat ACL Mask Port Status 301/301 (Unspecified) D 255.255.255.255 0 Unmonitoredphone2/kissops (Unspecified) D 255.255.255.255 0 Unmonitored34169788/341697 (Unspecified) D A 255.255.255.255 0 UNKNOWN Thanks. Raymond - Original Message - From: "Rod Bacon" [EMAIL PROTECTED] To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com Sent: Thursday, April 07, 2005 12:02 PM Subject: Re: [Asterisk-Users] SIP Softphone for testing with Asterisk I've tested about a dozen of them, and find firefly one of the best (others have more features, but I find firefly is a good mix of quality/features/performance). Make sure you get the third-party firefly though, not the one that's limited to virbiage. Try here... http://www.virbiage.com/firefly/download/firefly-thirdparty.exe - Original Message - From: "raymond" [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, April 07, 2005 1:41 PM Subject: [Asterisk-Users] SIP Softphone for testing with AsteriskHi all, I had just set up my asterisk server. Can anybody know that is there any sip softphone for testing with asterisk? (I had download some from internet but I think all are preconfig to certain server). Cheers. Raymond ___ 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] SIP Softphone for testing with Asterisk
Hi Ray I`m using SJphone softphones with my * , working fine.. Imran On Apr 7, 2005 8:41 AM, raymond [EMAIL PROTECTED] wrote: Hi all, I had just set up my asterisk server. Can anybody know that is there any sip softphone for testing with asterisk? (I had download some from internet but I think all are preconfig to certain server). Cheers. Raymond ___ 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] SIP Softphone for testing with Asterisk
On Wed, 2005-04-06 at 22:41, raymond wrote: Hi all, I had just set up my asterisk server. Can anybody know that is there any sip softphone for testing with asterisk? (I had download some from internet but I think all are preconfig to certain server). I'm using xten x-lite on windows and linux [1] : You can download from www.xten.com [1] The linux beta version can be downloaded from http://support.xten.com you must have to register as forum user and sent a message to the forum admin to be accepted as beta tester. Once you have accepted, you can have access to the beta testers forum. Otherwise, http://xten.com/apps/xprolinuxbeta/xlite-linux-23.bz2 have the last beta. Don't forget to register and report any bugs. Cheers. Raymond ___ 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 -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.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] SIP Softphone for testing with Asterisk
Hi all, I had just set up my asterisk server. Can anybody know that is there any sip softphone for testing with asterisk? (I had download some from internet but I think all are preconfig to certain server). Cheers. Raymond ___ 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] SIP Softphone for testing with Asterisk
raymond wrote: Hi all, I had just set up my asterisk server. Can anybody know that is there any sip softphone for testing with asterisk? (I had download some from internet but I think all are preconfig to certain server). That isn't a very good question. Did you google for asterisk soft phone? I got 78,000 responses, with virtually all of the first page being the information you are looking for. This is a very high-volume list. Your questions will get more friendly responses if it appears you have done the slightest amount of work on your own to see what you might find. B. ___ 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] SIP Softphone for testing with Asterisk
I've tested about a dozen of them, and find firefly one of the best (others have more features, but I find firefly is a good mix of quality/features/performance). Make sure you get the third-party firefly though, not the one that's limited to virbiage. Try here... http://www.virbiage.com/firefly/download/firefly-thirdparty.exe - Original Message - From: raymond [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, April 07, 2005 1:41 PM Subject: [Asterisk-Users] SIP Softphone for testing with Asterisk Hi all, I had just set up my asterisk server. Can anybody know that is there any sip softphone for testing with asterisk? (I had download some from internet but I think all are preconfig to certain server). Cheers. Raymond ___ 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] SIP Softphone
iH i received the card earlier this week but could not get the FXS port to work. The FXO port did work for a short time but now fails. (no lights on the card) and reports a number of errors to syslog concerning power up failures. would like to return the card and get a replacement thanks - hcir fyi this is my second email On Jun 28, 2004, at 1:16 PM, Rodrigo P. Telles wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Arve, I've been using kphone (http://www.wirlab.net/kphone/) with success. It's simple and works fine :-) []'s Arve Rasmussen wrote: | Hi, | | What is the best SIP softphone to use with Asterisk? | | I have a hard time finding OpenSource SIP soft phone. | | Regards | | Arve5 | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFA4Iq4iLK8unYgEMQRAmDpAJ49AAIqNUN5t1uhvPL0dwt/bub8PgCeOkZn QhjZWvivXvlwdYCO+mz0tWE= =Kslk -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Softphone
Hi, What is the best SIP softphone to use with Asterisk? I have a hard time finding OpenSource SIP soft phone. Regards Arve5 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SIP Softphone
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Arve, I've been using kphone (http://www.wirlab.net/kphone/) with success. It's simple and works fine :-) []'s Arve Rasmussen wrote: | Hi, | | What is the best SIP softphone to use with Asterisk? | | I have a hard time finding OpenSource SIP soft phone. | | Regards | | Arve5 | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFA4Iq4iLK8unYgEMQRAmDpAJ49AAIqNUN5t1uhvPL0dwt/bub8PgCeOkZn QhjZWvivXvlwdYCO+mz0tWE= =Kslk -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SIP Softphone
On Mon, 2004-06-28 at 16:16, Rodrigo P. Telles wrote: I've been using kphone (http://www.wirlab.net/kphone/) with success. It's simple and works fine :-) kphone only supports inband DTMF and so will only support DTMF when using ulaw or alaw. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Softphone
Hi, I am new in VOIP area, so any help is really appreciated. I setup asterisk at home and I am trying softphone. I download SJphone from SJlabs and I can place calls. Question is, how can I make a call to that softphone What would be config in asterisk and in softphone. I am trying to use SIP. Also, can I make call outside trough modem on Linux? and how. Thank you for any help Drazen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP softphone volume control?
Hi, Where are the settings to access the demo server at Digium? I would like to setup and test x-lite as well with a running asterisk until i get my box up and running. Thanks -- Original Message -- From: Chris Albertson [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Wed, 8 Oct 2003 10:53:12 -0700 (PDT) I went back to the example system direct from CVS with small additions to sip.conf and extnsion.conf needed to make one xten X-Lite phone work. I can dail in and hear the anouncements, call the demo server at Digium. The audio quality I hear comming from Asterisk back to X-Lite is good (9 on a 10 scale) but the sound volume comming from the X-Lite extension is very low even hard to hear. I know about the mic. level adjustment on X-Lite and I've got it set high almost to the point of clipping I appears that the Asterisk server is somehow scaling the sound down. Is this adjustable? Some way to set a per extension gain cotrol? = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP softphone volume control?
I went back to the example system direct from CVS with small additions to sip.conf and extnsion.conf needed to make one xten X-Lite phone work. I can dail in and hear the anouncements, call the demo server at Digium. The audio quality I hear comming from Asterisk back to X-Lite is good (9 on a 10 scale) but the sound volume comming from the X-Lite extension is very low even hard to hear. I know about the mic. level adjustment on X-Lite and I've got it set high almost to the point of clipping I appears that the Asterisk server is somehow scaling the sound down. Is this adjustable? Some way to set a per extension gain cotrol? = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP softphone volume control?
I went back to the example system direct from CVS with smalladditions to sip.conf and extnsion.conf needed to make onexten X-Lite phone work. I can dail in and hear the anouncements,call the demo server at Digium. The audio quality I hearcomming from Asterisk back to X-Lite is good (9 on a 10 scale)but the sound volume comming from the X-Lite extension is very loweven hard to hear. I know about the mic. level adjustment onX-Lite and I've got it set high almost to the point of "clipping"I appears that the Asterisk server is somehow scaling the sound down.Is this adjustable? Some way to set a per extension gain cotrol? Chris, Try adjusting both Input and Output from the WindowsVolume control. Many Window programs change these settings.
Re: [Asterisk-Users] SIP Softphone Echo!!
It is the responsibility of your device (SJpnone, mic/speaker pc) to handle it's half of the echo problem (prevent what is playing in the speaker from being picked up in the microphone. I played with sjphone months ago with mic speakers and experienced the same trouble. I would expect it to work much much better with headphones instead of pc speakers (the test licence expired before I could check this out). Are there any parameters I can use in the sip.conf or anywhere else to reduce the echo?? When making a PSTN call from the phone on the S100U there is no echo at all.. ( I have echocancel=yes and echocancelwhenbridged=yes in zapata.conf ) You are handling the asterisk end here. In fact echocancelwhenbridged=no should be fine as well. If you are connecting sip calls to telco or pbx it may be beneficial to uncomment KFLAGS+=-DAGGRESSIVE_SUPPRESSOR in the zaptel Makefile then make install. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Softphone Echo!!
I am using a plantronics headset.. so as far as I know the echo is not being caused by the mic picking up what is coming out of the headphones. I have looked at all the settings in the software and I can't find any echo cancellation features.. Oh well, guess its time to look for another SoftPhone... :) - Original Message - From: John Harragin [EMAIL PROTECTED] Date: Fri, 21 Mar 2003 22:49:17 -0500 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP Softphone Echo!! It is the responsibility of your device (SJpnone, mic/speaker pc) to handle it's half of the echo problem (prevent what is playing in the speaker from being picked up in the microphone. I played with sjphone months ago with mic speakers and experienced the same trouble. I would expect it to work much much better with headphones instead of pc speakers (the test licence expired before I could check this out). Are there any parameters I can use in the sip.conf or anywhere else to reduce the echo?? When making a PSTN call from the phone on the S100U there is no echo at all.. ( I have echocancel=yes and echocancelwhenbridged=yes in zapata.conf ) You are handling the asterisk end here. In fact echocancelwhenbridged=no should be fine as well. If you are connecting sip calls to telco or pbx it may be beneficial to uncomment KFLAGS+=-DAGGRESSIVE_SUPPRESSOR in the zaptel Makefile then make install. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Softphone for RedHat 8.0
Hi, I am looking for a SIP (or IAX) softphone for Gnome on RedHat 8.0.. I have tried the Gnophone and Linphone RPM's and both don't seem to work so I was wondering if there were any others out there.. I don't know how to compile software if it requires more than running 'make install'.. Gnophone seems to require something more than this simple command to get it to compile using the CVS source.. Thanks.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
