Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer
You can find it here: http://www.zoiper.com/downloads/free/linux/zoiper201-linux.tar.gz Note that the linux version does not support TLS and SRTP yet. * Instructions: * 1) Download zoiper201-linux.tar.gz 2) Extract Zoiper. If you don't use a GUI application for archive processing, here is the command line: tar zxf zoiper201-linux.tar.gz ./zoiper 3) Start Zoiper. *ZoIPer depends on ALSA library, so it* **must** *be installed! * Zoa Robert Moskowitz wrote: zoa wrote: Have you tried our Zoiper softphone yet (www.zoiper.com) - new version scheduled for in a couple of days ? If so, can you send me any remarks of list so that we can keep those things in mind for future versions ? Do you know where I can get it as an rpm to install on Centos 5 with Gnome? I do not have the time resources to do compiles. I am really a security protocol researcher and would be very interested in seeing what you have done for SIP TLS and SRTP. But for the later, I am all Linux. The one XP system is a corp box that I cannot add any software too. ___ -- 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
Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer
Zoiper is pretty impressive, it's a simple, neat little client. The one problem I have with it is the keyboard. I've had problems trying to use the keyboard to send DTMF on the current call. The left hand popout keypad is also a little small for my users' taste. It would be nice to have a keyboard hang-up, something like ESC, ditto for things like cancel buttons around the app. I really like the fact it does both SIP and IAX. Onto sillier issues: the icon is nice, but it would be great to have proper gamma anti-aliasing on the mac one. Just my .02 on the free mac os version, I might have to check out the biz edition too. It's all looking good. Good luck with the next release! Simon Simon Elliston Ball [EMAIL PROTECTED] On 23 Jan 2008, at 08:35, Zoa wrote: You can find it here: http://www.zoiper.com/downloads/free/linux/zoiper201-linux.tar.gz Note that the linux version does not support TLS and SRTP yet. * Instructions: * 1) Download zoiper201-linux.tar.gz 2) Extract Zoiper. If you don't use a GUI application for archive processing, here is the command line: tar zxf zoiper201-linux.tar.gz ./zoiper 3) Start Zoiper. *ZoIPer depends on ALSA library, so it* **must** *be installed! * Zoa Robert Moskowitz wrote: zoa wrote: Have you tried our Zoiper softphone yet (www.zoiper.com) - new version scheduled for in a couple of days ? If so, can you send me any remarks of list so that we can keep those things in mind for future versions ? Do you know where I can get it as an rpm to install on Centos 5 with Gnome? I do not have the time resources to do compiles. I am really a security protocol researcher and would be very interested in seeing what you have done for SIP TLS and SRTP. But for the later, I am all Linux. The one XP system is a corp box that I cannot add any software too. ___ -- 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 ___ -- 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
Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer
On Wed, 23 Jan 2008, Zoa wrote: You can find it here: http://www.zoiper.com/downloads/free/linux/zoiper201-linux.tar.gz Note that the linux version does not support TLS and SRTP yet. * Instructions: * 1) Download zoiper201-linux.tar.gz 2) Extract Zoiper. If you don't use a GUI application for archive processing, here is the command line: tar zxf zoiper201-linux.tar.gz ./zoiper 3) Start Zoiper. I liked Zoiper when it was idefisk however I'm very irritated that they changed the account limit to 2 in Zoiper after it was seemingly unlimited in idefisk, so guess what I stick with... Gordon ___ -- 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
Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer
Ok good piece software easy on the eyes as they say and I have to say this before I start listing a lot of things that I would love to see, for it to be usable as a good high performance phone. Working with industrial pc switchboards and soft phones of various vendors for some years now, and it all boils down to. How much functionality you can boil into the keyboard. No mouse action should be needed to search a number add an F-key for it. No mouse action should be needed to dial or transfer a number. No mouse action should be needed unless absolutely unavoidable. A_PARTY = caller B_PARTY = operator / called person C_PARTY = number to transferred to STATES: Example to keep it within the numeric key-pad when you receive a call and transfer it. STEP 1 A call is presented. LINE_STATE: Ringing TRANSFER_STATE: inactive TALKING_TO_STATE: inactive STEP 2 Press numeric enter to pick up call. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: inactive TALKING_TO_STATE: A_PARTY STEP 3 Transfer the call Scenario 1: Search out the number in the phonenbook by pressing ex: F10, while talking to the caller, the phone book appears search by name, number or whatever is available and mark the number with arrow keys and dial with NUM-enter. Scenario 2 Press enter a new dial box appears. Type in the number to call. Press enter. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CALLING_C_PARTY TALKING_TO_STATE: DIALBACKTONE STEP 4 The person transferring the call can now make a choice either to do a attended transfer or a blind transfer. Scenario Blind transfer: Simply pressing NUM-enter should do a blind transfer, and the call handling is done and all states are reset, C_PARTY becomes the B_PARTY and so on. The phone is ready for a new call. LINE_STATE: inactive TRANSFER_STATE: inactive TALKING_TO_STATE: inactive Scenario: Attended transfer: The person transferring the call can talk to the C_PARTY LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: C_PARTY Should the operator wish for switching back do the previous call that currently placed on hold it could be done by pressing the NUM+ key placing the C_PARTY on hold and reconnecting the A_PARTY LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: A_PARTY Switch back by NUM+ LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: C_PARTY Connect the call by NUM-enter at any point talking to either the A_PARTY or C_PARTY. The call handling is done and all states are reset, C_PARTY becomes the B_PARTY and so on. The phone is ready for a new call. LINE_STATE: inactive TRANSFER_STATE: inactive TALKING_TO_STATE: inactive Scenario: disconnect the party you are talking to Press NUM- If the states are as follows. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: CONNECTED_C_PARTY TALKING_TO_STATE: C_PARTY The C_PARTY would be disconnected and the states would go to. LINE_STATE: CONNECTED_A_PARTY TRANSFER_STATE: inactive TALKING_TO_STATE: A_PARTY And the here we go again with a new transfer or a goodbye and hang up with NUM-. Some side notes: The calling transfer functions are already in the phone alle that needs to be done is associate the functions to the states and numeric keys. The features could be activated by putting the phone in operator mode, if this was the case you could turn of the DTMF and just start typing the new number and hit NUM-enter twice to transfer the call fast. 1 enter to dial number the other to transfer. DTMF could be turned of since the operator rarely calls any ivr, that needs a DTMF response, if so you could leave dtmf open on the QWERTY number keys HEX 30 31 33 34 so on. A tcp port on the phone that allowed for picking up calls and hanging up calls, and perhaps being able to read the number status would make is possible for people write some very nice callcenter agent software for this phone, without having to worry about the functionality of a phone in their agent software. These things might be on the table already if so happy days and then I can't wait to see the product then. Shw that was a little longer than expected. Just my way to keep it simple :), but I hope this could the first really good sip phone with switchboard properties out there. Regards Christian Ejlertsen -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Simon Elliston Ball Sent: 23. januar 2008 13:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer Zoiper is pretty impressive, it's a simple, neat little client. The one problem I have with it is the keyboard. I've had problems trying to use the keyboard to send DTMF on the current call. The left hand popout keypad is also a little small for my users' taste
Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer
the functionality of a phone in their agent software. These things might be on the table already if so happy days and then I can't wait to see the product then. Shw that was a little longer than expected. Just my way to keep it simple :), but I hope this could the first really good sip phone with switchboard properties out there. Regards Christian Ejlertsen -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Simon Elliston Ball Sent: 23. januar 2008 13:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer Zoiper is pretty impressive, it's a simple, neat little client. The one problem I have with it is the keyboard. I've had problems trying to use the keyboard to send DTMF on the current call. The left hand popout keypad is also a little small for my users' taste. It would be nice to have a keyboard hang-up, something like ESC, ditto for things like cancel buttons around the app. I really like the fact it does both SIP and IAX. Onto sillier issues: the icon is nice, but it would be great to have proper gamma anti-aliasing on the mac one. Just my .02 on the free mac os version, I might have to check out the biz edition too. It's all looking good. Good luck with the next release! Simon Simon Elliston Ball [EMAIL PROTECTED] On 23 Jan 2008, at 08:35, Zoa wrote: You can find it here: http://www.zoiper.com/downloads/free/linux/zoiper201-linux.tar.gz Note that the linux version does not support TLS and SRTP yet. * Instructions: * 1) Download zoiper201-linux.tar.gz 2) Extract Zoiper. If you don't use a GUI application for archive processing, here is the command line: tar zxf zoiper201-linux.tar.gz ./zoiper 3) Start Zoiper. *ZoIPer depends on ALSA library, so it* **must** *be installed! * Zoa Robert Moskowitz wrote: zoa wrote: Have you tried our Zoiper softphone yet (www.zoiper.com) - new version scheduled for in a couple of days ? If so, can you send me any remarks of list so that we can keep those things in mind for future versions ? Do you know where I can get it as an rpm to install on Centos 5 with Gnome? I do not have the time resources to do compiles. I am really a security protocol researcher and would be very interested in seeing what you have done for SIP TLS and SRTP. But for the later, I am all Linux. The one XP system is a corp box that I cannot add any software too. ___ -- 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 ___ -- 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 ___ -- 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 ___ -- 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] Free IAX / SIP Softphone with attended transfer
Hello, any one advise a good, strong and free softphone that can work with SIP or/and IAX lines and supports attended transfer ? Thanks for help. Mit freundlichen Grüßen / best regards André Herrlich IT-Operator / Developer LetMeRepair LMR Service and Consulting GmbH Fichtestr. 1A 02625 Bautzen Tel.: + 49 - (0)3591 - 2722 - 1451 Fax: + 49 - (0)3591 - 2722 - 1459 E-Mail: [EMAIL PROTECTED] Internet: www.letmerepair.com ___ -- 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
Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer
Andre Herrlich wrote: any one advise a good, strong and free softphone that can work with SIP or/and IAX lines and supports attended transfer ? IMHO there are no good softphones - at least not for Mac OS X and I think that is true for Linux as well. They're either not stable or their interface is unusable. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- 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
Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer
Hello, Have you tried our Zoiper softphone yet (www.zoiper.com) - new version scheduled for in a couple of days ? If so, can you send me any remarks of list so that we can keep those things in mind for future versions ? Greetings, Joachim Philipp Kempgen wrote: Andre Herrlich wrote: any one advise a good, strong and free softphone that can work with SIP or/and IAX lines and supports attended transfer ? IMHO there are no good softphones - at least not for Mac OS X and I think that is true for Linux as well. They're either not stable or their interface is unusable. Regards, Philipp Kempgen ___ -- 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
Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer
On Jan 22, 2008, at 10:23 AM, Philipp Kempgen wrote: Andre Herrlich wrote: any one advise a good, strong and free softphone that can work with SIP or/and IAX lines and supports attended transfer ? IMHO there are no good softphones - at least not for Mac OS X and I think that is true for Linux as well. They're either not stable or their interface is unusable. I've been happy with both, zoiper and x-lite at a osx, (...may be I have a low threshold for hapiness?) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- 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 ___ -- 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
Re: [asterisk-users] Free IAX / SIP Softphone with attended transfer
--- zoa [EMAIL PROTECTED] wrote: Hello, Have you tried our Zoiper softphone yet (www.zoiper.com) - new version scheduled for in a couple of days ? If so, can you send me any remarks of list so that we can keep those things in mind for future versions ? Attended transfer is not available in the free version and that's what the OP is looking for. I've been using SJPhone (v. 1.60.289a , not the newer one) and it has been working just fine (supports attended transfers). It's free software but it's development seems to have come to a stop. I was hoping to move to Ekiga but haven't had time to test it yet. Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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
