[Ekiga-list] Feature Request: Support E164.org DynDNS API or Remote Management Protocol
Hi, we all know we can map E164 numbers to SIP addresses by using ENUM. But what about road warriors or people with dynamic IPs? Currently they have to use a Dynamic DNS service like DynDNS.org and add NAPTR records to their ENUM number in E164.org. But there are much easier ways - both, the DynDNS API http://www.e164.org/wiki/DynDNS?highlight=(dyndns) and the Remote Management Protocol http://www.e164.org/wiki/RemoteManagementProtocol; allow to update NAPTR records automatically. So I suggest to implement functions in Ekiga to automatically add/update a NAPTR record in E164.org's DNS system to the current IP address and SIP port of the Ekiga client. Have phun with Ekiga Renne ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
Re: [Ekiga-list] how to bypass SIP server and call a computer directly?
Rene Bartsch [EMAIL PROTECTED] writes: What does Ekiga do with a user name on an incoming call? Just ignore it? As Ekiga doesn't have a user management, I assume so. What happens in general when the user name is left out? Calling Ekiga by IP/domain worked fine for me (haven't tested 3.x series, yet). But there's one special case in Ekiga. Alphanumeric addresses are interpreted as sip: while numeric-only addresses are interpreted as ENUM addresses. Thanks for this info! Does this mean that 1234 is equivalent to enum:1234? Is there some standard prefix like âenum:â that is used for ENUM addresses? http://wiki.ekiga.org/index.php/Enum ;) Renne ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
Re: [Ekiga-list] how to bypass SIP server and call a computer directly?
michel memeteau [EMAIL PROTECTED] writes: I think putting [EMAIL PROTECTED] will work. Thanks for this information! Testing seems to indicate that [EMAIL PROTECTED], IPADDRESS, sip:[EMAIL PROTECTED], and sip:IPADDRESS all seem to work similarly. What is the rule here? It seems that the âsip:â prefix is the default and can be omitted. What is used for the part to the left of the [EMAIL PROTECTED] when it is omitted? Depends on the receiver. The syntax is user@host:port. When using standard ports, you don't need port (defaults to 5060). host is either domain or IP. Some soft- and especially hardphones demand for a user while others don't. The AVM Fritzbox is much more special as it only calls callees with a SRV record. But often ip only works. But there's one special case in Ekiga. Alphanumeric addresses are interpreted as sip: while numeric-only addresses are interpreted as ENUM addresses. Renne ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
Re: [Ekiga-list] Setting up a videophone (from scratch)
I have an application; a person walks up to the Linux box, presses a button and (after a sec, with some pretty graphics) a person's face comes up and we're talking to someone we're also seeing. This shouldn't be hard, the idea is to make this manual free at all times. With this background, here's the question: How hard is it to open a communication with a remote person (probably on an Asterix PBX built for this purpose) to make the call _from_an_icon_ or other no-brainer construct? Ekiga is a great tool, even better than it was as GnomeMeeting, but I need something less flexible (to the user) and more simple, too. Is that something that's done without source code? It's some kind of overkill, but Asterisk has video support for the oss console channel. I don't know which version you need, or even CVS version, but I've read about a patch added some month ago. Renne ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
Re: [Ekiga-list] For or against ?
Hello, Would you be for or against such a scheme ? http://www.voip-info.org/wiki/view/Fee+Announcement+from+Free+World +Dialup I've received that mail, too. With the result I won't use FWD anymore. Drawbacks: 1.) Having fees forces the use of payment systems which often need additional personal information of the user. NOT Good! :( 2.) There is no global numbering plan and I won't register with 20 registrars to pay 600 $ per year. So I'd propose a donation system with a donation meter with red, yellow and green area showing how many days the system can be kept running. When reaching yellow, comfort functions will be disabled, and red means the service will be suspended until enough donations have been made. By the way, you don't need a SIP registrar when using a ENUM-Domain with NAPTR-,A- and SRV-Records. So I would only pay for a ENUM-Domain with some kind of DynDNS or SOAP interface to update the records. Cheers, Renne ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
[Ekiga-list] Roadmap for Ekiga 3.00?
Hi, how's the current state of upcoming version 3.00? About when will a stable version be available? Will it have full IAX2 support? Thanks Renne ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
Re: [Ekiga-list] Access Numbers Peering
Does ekiga.net have ENUM registrations? FWD uses a [ENUM-Prefix][numeric FWD user ID] for the mapping. That way any ENUM-capable UA can call FWD-users. Regards, Renne Le samedi 10 mai 2008 à 12:27 -0300, J. Paul Bissonnette a écrit : 1 last question how does some from another net reach me at ekiga.net? Your SIP address should be enough: sip:[EMAIL PROTECTED] The Ekiga.net service accept calls to its registered users without being registered to Ekiga.net. Just call the sip:[EMAIL PROTECTED] address directly. With this solution you'll need an alphanumeric keypad. Service numbers like the Ekiga.net echo test (sip:[EMAIL PROTECTED]), or the conference rooms are reserved to registered users of the Ekiga.net service. http://wiki.ekiga.org/index.php/Peering#Using_the_Ekiga.net_SIP_address yannick wrote: Le samedi 10 mai 2008 à 10:06 -0300, J. Paul Bissonnette a écrit : I am using Ekiga.net how do I call a user on another net eg. FWD or sipgate.de Hi, You can use peering: to reach FWD: [EMAIL PROTECTED] to reach sipgate.de: [EMAIL PROTECTED] (but this not seems to work atm) More infos: http://wiki.ekiga.org/index.php/Peering Regards, Yannick Thanks Paul ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list -- Me joindre en téléphonie IP / vidéoconférence ? sip:[EMAIL PROTECTED] Logiciel de VoIP Ekiga : http://www.ekiga.org http://wiki.ekiga.org/index.php/Which_programs_work_with_Ekiga_%3F ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
[Ekiga-list] IAX2 support (was (Re: [Ekiga-devel-list] Ekiga 3.00 available for WIN32 *only*)
and we can bury SIP finally. Joking ? Sip support is really important. Most VideOIP software support it. Keep cool! I think all he wanted to say was: Nobody who does not *want* to use SIP but would rather prefer IAX2 (for example, because he's behind a NAT firewall and has problems to make STUN work properly) will not be forced into SIP anymore. Yes, I just want to be able to connect to anyone with SIP. Mmh, I never tried whether IAX clients can work work peer-to-peer (with some DNS tricks it's possible with any SIP client). If IAX2 will marginalize SIP (as SIP did with H.323) is to be seen. If it's standardized it has the potential to do this within a few years. - VSPs: Less horsepowers on the VoIP servers - Hardware vendors: Much cheaper hardware - Users:Much more reliable, only one NAT-port By the way, about three years ago I did some testing with IAX and SIP. On a Server (Intel P4, 1024 MB RAM, 100 MBit/s inernet connection) I had set up an Asterisk installation with facsimile support. The server was connected to a PSTN gateway of a VSP with SIP and IAX. Then I tried to send 10 facsimilies from a facsimile machine (FAX (analogue) - ISDN - PSTN - PSTN Gateway - Internet - Asterisk server). The latencies between PSTN gateway and Asterisk server were about 5 -10 msecs. With IAX eight facsimilies were successful, SIP failed completely! Just wondering, and yes, this is entirely off-topic: Does IAX2 support video? http://iaxclient.wiki.sourceforge.net/ Regards, Rene ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
Re: [Ekiga-list] IAX2 support (was (Re: [Ekiga-devel-list] Ekiga 3.00 available for WIN32 *only*)
Yes, I just want to be able to connect to anyone with SIP. Oops, meant IAX2, of course !!! Regards, Rene ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
[Ekiga-list] State of ZRTP integration?
Hi, how is the current state of ZRTP integration into Ekiga? Thanx Renne ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
Re: [Ekiga-list] Why H.263 is not used in ekiga?
Hi all, Eventhough H.263 codec in opal supports high resolutions, why it is not used in ekiga? Please be gentle. I am quite new using ekiga. Please help me by giving a reply. That's a quite interesting question. When testing Ekiga with 1 MBit/s upstream in the LAN resolution was something about 172 dpi although my Logitech QuickCam for Notebooks Pro provides a resolution of 640x480 (V4L2). It would be nice to have more efficency (higher resolution and framerate) in Ekiga. Thanx for any hint Renne ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
[Ekiga-list] TOS settings?
Hi, which Type of Service settings does Ekiga set for the network connection? Renne ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
Re: [Ekiga-list] Feature request: ZRTP support
On Mon, 26 Feb 2007 18:50:26 +0100 (CET) Rene Bartsch [EMAIL PROTECTED] wrote: Hi, could you please add ZRTP support to Ekiga? I'm working on ZRTP integration into OPAL. I'm hoping to have it finished in the next few weeks. Great! ;-) That way VoIP gets more and more attraction even from people who didn't consider using VoIP before! ;-) Best regards Renne ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
[Ekiga-list] Feature request: ZRTP support
Hi, could you please add ZRTP support to Ekiga? There is a SDK at http://zfoneproject.com/prod_sdk.html. But I don't know which license will be available for OSS projects. Renne ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
Re: [Ekiga-list] Feature request: ZRTP support
On Mon, 26 Feb 2007 18:50:26 +0100 (CET) Rene Bartsch [EMAIL PROTECTED] wrote: Hi, could you please add ZRTP support to Ekiga? There is a SDK at http://zfoneproject.com/prod_sdk.html. But I don't know which license will be available for OSS projects. Renne http://bugzilla.gnome.org/show_bug.cgi?id=335594 I guess the main problem is time as usual. Regards, Jan Does Ekiga use GNU ccRTP or is RTP function included in OPAL or PWLIB? Regards, Renne ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
Re: [Ekiga-list] Click-to-Dial from Firefox
Hi, Le dimanche 04 février 2007 à 01:46 +0100, Rene Bartsch a écrit : Hi, is there any way to do click-to-dial (tel, sip and h323 uris) in Firefox? Running ekiga -c sip:xxx command only works in case Ekiga hasn't been started, yet. Doing this while running an Ekiga instance causes an error message. It should work in both cases. I just tried again here in a gnome-terminal. I've just tried again. Ekiga 2.0.4 ist automatically started by the autostart from KDE. When I run ekiga -c sip:[EMAIL PROTECTED] I get two windows with error messages, one complaining about the listener port for SIP being in use and the other complaining about the listener port for H.323 being in use. When I press OK in both windows, a new Ekiga instance is started dialing sip:[EMAIL PROTECTED]. But every time I get the two error messages and two instances of Ekiga trying to use the same resources. Is there any voodoo in Gnome which is not available in KDE? Is it possible to use some dbus-send xxx command for dialing (I have no plan of dbus ...)? Thanx for any hint Renne ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
[Ekiga-list] Click-to-Dial from Firefox
Hi, is there any way to do click-to-dial (tel, sip and h323 uris) in Firefox? Running ekiga -c sip:xxx command only works in case Ekiga hasn't been started, yet. Doing this while running an Ekiga instance causes an error message. Is there any other way to send a remote command to Ekiga? Thanx for any hint ... Renne ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
Re: [Ekiga-list] Ekiga 2.04 and ALSA
Since I have upgraded to Ekiga 2.04, I can no longer use the default device for input or output. I have to use my sound card directly. I was wondering if anyone else was experiencing this problem? I had that problem with Ekiga-2.0.3. With 2.0.4 it's gone :-) Maybe you have to edit you asound.conf ... Renne ___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
