Yes, it's the KDE client, installed from the Launchpad PPA for stable releases
(https://launchpad.net/~savoirfairelinux/+archive/ppa).
Curiously, the About dialog indicates 1.2.1. But the installed package is 1.2.2:
steve@t520:~$ apt-cache policy sflphone-client-kde
sflphone-client-kde:
Installed: 1.2.2~ppa1~quantal
Candidate: 1.2.2~ppa1~quantal
Version table:
*** 1.2.2~ppa1~quantal 0
500 http://ppa.launchpad.net/savoirfairelinux/ppa/ubuntu/ quantal/main
amd64 Packages
100 /var/lib/dpkg/status
I will try the GNOME client a bit later and report back.
My employer uses an Avaya Communication Server of some kind. I don't know the
details, I'll need to research that a bit more. Configuration is fairly simple;
I provide only the server's IP, my username (which is my extension), and the
password on the "Basic" page. No STUN is required. No realm/auth-name/password.
Does selection of codec have anything to do with it, perhaps?
On 2013-03-11 16:12:53 Emmanuel Lepage <[email protected]>
wrote:
>
> Hi Steve,
>
> Are you using the KDE client? If yes, from where have you downloaded it? Are
> you sure it say version 1.2.2 in Help->About SFLPhone KDE? 1.2.0 had a bug
> that prevented DTMF from being sent, but it have been fixed months ago. I
> just tried many different scenarios and most seem to work, but you are right,
> you can't hear the sound in analog as it seem to stay digital from point to
> point. This may the problem. If you have access to your SIP provider
> configuration, you could check how it is configured. Here are some links:
>
> http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode
> http://www.voip-info.org/wiki/view/SIP+Info+DTMF
>
> I am not really the one who can help you debugging the issue further, but if
> you could check with both KDE and Gnome clients, it would be appreciated. If
> you still can't get it to work, can you give us more information about your
> SIP provider/registrar?
>
> Thanks for reporting this
>
> ----- Original Message -----
> From: "Steve Riley" <[email protected]>
> To: [email protected]
> Sent: Monday, March 11, 2013 3:01:34 PM
> Subject: [SFLphone] DMTF not received on other side
>
> Good day. Similar to another issue I saw in the mailing list archive, I'm
> unable to get SFLphone to send tones to the other side. This makes it
> impossible to work with conference call systems and other VRUs.
>
> Details:
>
> * Kubuntu 12.10
> * SFLphone 1.2.2~ppa1~quantal
> * protocol: SIP
> * server: Avaya
> * SRTP: disabled
>
> I've tried both settings for DTMF: Over RTP and Over SIP.
>
>
> When configured with over SIP, the log shows this each time I press a digit:
>
> sipvoiplink.cpp:1367:tid 63360: Unknown DTMF type oversip, defaulting to
> sipinfo instead
>
>
> When configured with over RTP, the log shows this:
>
> audio_rtp_session.cpp:114:tid 63232: Send RTP Dtmf (5)
>
>
> However, the tone is not actually heard by the other end. As a test, I
> installed Ekiga, and the tones it generates are heard fine.
>
> Is there any additional debugging or configuration information I can supply
> to help find the cause, and a solution? Thanks.
>
> ...Steve
>
> _______________________________________________
> SFLphone mailing list
> [email protected]
> http://lists.savoirfairelinux.net/mailman/listinfo/sflphone
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