Has any one got ekiga to work?

English Network/Internet
1

I made an account. I’m trying it.

In the terminal I start it I see “XCAP error: unauthorized”, and in the
account list it says: “Ekiga.net Registered”.

I have seen a post here with the same problem
(http://forums.opensuse.org/english/get-technical-help-here/applications/436659-no-echo-ekiga.html)
saying to enable certain ports in the router:


Protocols 	Ports 	Types 	Descriptions
SIP 	5000 to 5100 	UDP 	SIP signalling, listen port: 5060
STUN 	3478 to 3479 	UDP 	Outgoing traffic to the STUN server
H323 	1720 	TCP 	H323 listen port

I did that. No difference, same xcap error.

I see no entries in the firewall log, but dropping it makes no difference.

I tried the test call, it completes and I see a stream of data going to
internet, about 5 KB/s constant, but no sound comes back, there is no
“echo”. I tried changing sound settings, no change, I hear nothing (it is
using alsa, no pulse support, apparently)

Sound otherwise in this laptop works fine. Ring test in ekiga sounds.

I tried also the call back test, but I get no call back.

Is there any other phone application that is KNOWN to work?

For example, I looked at linphone, but I see no test phone to see if it
works or not.

–
Cheers / Saludos,

Carlos E. R.
(from 11.4 x86_64 “Celadon” at Telcontar)

2

On 2012-02-27 00:38, Carlos E. R. wrote:

> In the terminal I start it I see “XCAP error: unauthorized”, and in the
> account list it says: “Ekiga.net Registered”.

This problem I haven’t located.

…

> I tried the test call, it completes and I see a stream of data going to
> internet, about 5 KB/s constant, but no sound comes back, there is no
> “echo”. I tried changing sound settings, no change, I hear nothing (it is
> using alsa, no pulse support, apparently)

This one is final: no pulse support means no sound, it simply does not work
(microphone for sure doesn’t work). All SIP phones I found for Linux want
to use ALSA, and thus, all of them fail to work.

I have only located one that uses pulse, that is SFLphone. The problem with
this one is finding a sip server to register with! Because the one they
have gives incomplete registration data, there is no password.

And using the registration data of, for example, ekiga, it doesn’t work.
Apparently SIP support is not standardized. And I’m not the only one
finding this problematic, there is a non answered question on their mail
list asking about this.

So it seems that the only VoIp that works in Linux is Skype or Google talk :-/

–
Cheers / Saludos,

Carlos E. R.
(from 11.4 x86_64 “Celadon” at Telcontar)

3

I use Linphone
Works perfectly

4

On 2012-02-27 05:36, caf4926 wrote:
>
> I use Linphone
> Works perfectly

In gnome with pulse enabled?

Do you know of a test address that can be used with linphone?

–
Cheers / Saludos,

Carlos E. R.
(from 11.4 x86_64 “Celadon” at Telcontar)

5

I use kde

I just call home to test

6

On 2012-02-27 15:16, caf4926 wrote:

> I use kde
>
> I just call home to test

I can’t call myself.

I tried “echo@iptel.org” and got “no common codecs”. Then I tried
“500@ekiga.net” and got no echo, the microphone doesn’t work. I see a
stream of about 10 Kb/s going out the network, so it is working, but sound
is not.

I can hear the test ring sound, though.

I can make a recording using “gmerlin”, both video and sound working, via
pulse, and can control both microphone and speakers. My hardware is fine.

But linphone insists on using the obsolete alsa libraries, which do not
work under gnome any more.

Yes, I have tried disabling pulse. I got nowhere, phones apps
did not work, and sound stopped working for any other
application that worked before. I had to restore pulse
and fight the configuration, and reboot, and fight the
configuration again to get sound working again in apps.

I then tried SFLphone (telephony repo), configured with the linphone
account (because the sflphone account doesn’t have a password and there is
no way to get one. They have a mail list but do not understand my question,
they say that the password is stored in the configuration file).

So I tried sflphone with the linphone account. I called the ekiga echo
adress - no go. I tried the echo@iptel.org - instant success.

So, the only phone that works in gnome with the default pulse sound is
sflphone. I have tried ekiga, linphone, kiax, kphone, twinkle - if I
remember correctly.

The problem with this one is that the phone book doesn’t work, the history
doesn’t work, to choose an account of a list you have to set it up as the
default one… I don’t understand these problems, as the wikipedia reports
this as one of the best enterprise level phones! Maybe it has to do with
the repo version being 0.9.12, and the web saying the latest is 1.0.2
released on February. I’ll have to build it myself :frowning:

…]

After several cycles of configure, install whatever it needs (commoncpp2,
ccrtp, zrtp…) I’m stuck at this one:


No package 'dbus-c++-1' found
configure:17292: error: You need the DBus-c++ libraries (version 0.6.0-pre1
or better)

I don’t know which package is this one. I tried libdconf-dbus-devel and
dbus-1-devel-32bit, no go.

I’ll ask in the programming forum.

–
Cheers / Saludos,

Carlos E. R.
(from 11.4 x86_64 “Celadon” at Telcontar)

7

Question:
Is your machine (or the machine you’re connecting to) behind and crossing NAT?

The problems you describe sound a lot like that situation which <does not work> without some special configuration.

A “feature” of SIP is that it does not cross NAT by itself… You need to either configure your machine with a public IP address or place a SIP gateway on or side by side with the NAT device.

Alternative are to use a non-SIP protocol although most of those will also require gateways (Skype is a notable exception).

TS

TS

8

On 2012-03-01 02:16, tsu2 wrote:
>
> Question:
> Is your machine (or the machine you’re connecting to) behind and
> crossing NAT?

At this moment, the other machines are the echo services some sip providers
offer. And mine is, of course, behind nat, but stun is designed for the
purpose. I don’t have a problem there, as “sflphone” managed to work fine
and I could hear myself.

No, the problem is that all phones use plain alsa instead of pulse and the
microphone is thus unusable.

sflphone uses pulse instead and works fine.

–
Cheers / Saludos,

Carlos E. R.
(from 11.4 x86_64 “Celadon” at Telcontar)

9

On 2012-03-01 03:33, Carlos E. R. wrote:
> sflphone uses pulse instead and works fine.

I noticed another client, this time using java: Jitsi (http://jitsi.org).
There are working rpms. It is not only a sip client, but a instant
multimessage client. I did not try it fully, it is not my type.

–
Cheers / Saludos,

Carlos E. R.
(from 11.4 x86_64 “Celadon” at Telcontar)

10

There was an article about this in the recent Linux Format Magazine

11

On 2012-03-02 04:36, caf4926 wrote:

(jitsi)

> There was an article about this in the recent Linux Format Magazine

It is recommended by
http://www.iptel.org, that’s why I
installed it. But I do not want a client for multiple instant messaging
systems as I don’t use them, I only want a computer phone.

–
Cheers / Saludos,

Carlos E. R.
(from 11.4 x86_64 “Celadon” at Telcontar)

12

I use kde I just call home to test I can’t call myself.](http://www.utalkmarketing.com/Hubs/HubHome.aspx?HubID=100366)the problem is that all phones use plain alsa instead of pulse and the microphone is thus unusable.