Peter, can you post more about this?
R&B
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| Twinkle SIP phone v0.6 byteorder-clean! :) https://powerdeveloper.org/forums/viewtopic.php?f=3&t=511 |
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| Author: | kaltst [ Sat Feb 18, 2006 10:41 am ] |
| Post subject: | Twinkle SIP phone v0.6 byteorder-clean! :) |
Very good VoiceOverIP news for PowerPC-users: The programmer of the IMHO best SIP-phone around, e.g. compared to kphone or linphone, fixed some problems with the byteorder on replay of soundsamples by using libsndfile in Version 0.6. http://www.twinklephone.com 0.6.1 builds and runs great on my Pegasos-II with Debian. You can even build it without KDE (qt-only), although this will disable kaddressbook binding. Excellent freeware. Have fun! |
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| Author: | bbrv [ Sun Feb 19, 2006 9:18 am ] |
| Post subject: | |
Peter, can you post more about this? R&B
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| Author: | kaltst [ Sun Feb 19, 2006 6:47 pm ] |
| Post subject: | featurelist |
Hi R&B! I am using twinkle right now btw. - on the line for more then 1h.  Maybe quoting the featurelist might be of interest: In addition to making basic voice calls Twinkle provides you the following features regardless of the services that your VoIP service provider might offer. 2 call appearances (lines) Multiple active call identities Custom ring tones (new) Call Waiting Call Hold 3-way conference calling Mute Call redirection on demand Call redirection unconditional Call redirection when busy Call redirection no answer Reject call redirection request Blind call transfer Reject call transfer request Call reject Repeat last call Do not disturb Auto answer User defineable scripts to handle incoming calls (new) E.g. to implement selective call reject or distinctive ringing Send DTMF digits (RFC 2833) to navigate IVR systems STUN support for NAT traversal Send NAT keep alive packets when using STUN NAT traversal through static provisioning Missed call indication (new) History of call detail records for incoming, outgoing, successful and missed calls DNS SRV support Automatic failover to an alternate server if a server is unavailable Other programs can originate a SIP call via Twinkle, e.g. call from address book (new) System tray icon (now also on non-KDE builts) System tray menu to quickly originate and answer calls while Twinkle stays hidden |
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| Author: | popper [ Tue Feb 21, 2006 10:03 pm ] |
| Post subject: | easy sip servers for PPC or x86 |
hi kaltst these SIP phones seen like a very nice idea however my limited understanding is you need some form of sip server if your wanting to use them on a private lan/wan such as a wireless network ?. i forget the name now but i seem to remember that the only server i could find was some monsterous thing that didnt do anything to help the average slightly techy people install and configure a basic (less than 10 users) ip phonline. would you happen to know if something exists that would allow me/anyone to setup such a server with webserver for gui frontend an all-in one type package ?. im willing to dedicate an old pc 400mhz or some such to an ip sip server if i can collect enough howto info then just go out and get some usb sip phones or sw sip such as you point out. hell even a small mips server might be useful to enbed into one of the new wireless NOS units. using the sip over the internet would be a bonus for what i have in mind (small community wireless homewatch and such)but not essential. |
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| Author: | kaltst [ Thu Feb 23, 2006 11:10 am ] |
| Post subject: | server needed |
Hi Popper, you seem to refer to Asterisk http://www.asterisk.org/ when talking about "monsterous thing"?  True, that it is very huge and configuring might take a lot of time. (but maybe it´s worth it nevertheless and you gain knowledge for corporate solutions.) SIP is IMHO interesting, because of those many SIP providers/gateways. For LAN-use I tested gnome-meeting - you can connect between your LAN-IPs without having the need for any registration server. I had no need installing SIP-servers on LAN. You might give this one a try: http://freshmeat.net/projects/openser/ Let us know, if it is more easy to set up! Regards, Peter |
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| Author: | kaltst [ Sun Apr 30, 2006 10:54 am ] |
| Post subject: | new Twinklephone V0.7 |
Twinkle SIP-phone V0.7 is available: http://www.twinklephone.com btw, Debian now has V0.62 available as package in testing/unstable: http://packages.debian.org/cgi-bin/sear ... ds=twinkle I just built 0.7 on my Peg/Debian and it runs fine. New in 0.7: 29 apr 2006 - Release 0.7 ========================= - Speex support (narrow, wide and ultra wide band) - Support for dynamic payload numbers for audio codecs in SDP - Inband DTMF (option for DTMF transport in user profile) - UTF-8 support to properly display non-ASCII characters - --cmd command line option to remotely execute CLI commands - --immediate command line option to perform --call and --cmd without user confirmation. - --set-profile command line option to set the active profile. - Support "?subject=" as part of address for --call - The status icon are always displayed: gray -> inactive, full color -> active - Clicking the registration status icon fetches current registration status - Clicking the service icons enables/disables the service - Fancier popup from KDE system tray on incoming call. - Popup from system tray shows as long as the phone is ringing. - Reload button on address form - Remove special phone number symbols from dialed strings. This option can be enabled/disabled via the user profile. - Remove duplicate entries from the dial history drop down box - Specify in the user profile what symbols are special symbols to remove. - Changed default for "use domain to create unique contact header value" to "no" - New SIP protocol option: allow SDP change in INVITE responses - Do not ask username and password when authentication for an automatic re-regsitration fails. The user may not be at his desk, and the authentication dialog stalls Twinkle. - Ask authentication password when user profile contains authentication name, but no password. - Improved handling of socket errors when interface goes down temporarily. Bug fixes --------- - If the far end holds a call and then resumes a call while Twinkle has been put locally on-hold, then Twinkle will start recording sound from the mic and send it to the far-end while indicating that the call is still on-hold. - Crash on no-op SDP in re-INVITE - Twinkle exits when it receives SIGSTOP followed by SIGCONT - call release cause in history is incorrect for incoming calls. Build improvements ------------------ - Break dependency on X11/xpm.h |
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