Puppy Linux Discussion Forum

[HairyWill]

[HairyWill]

Here is psip-0.10 a version with the interface modded to fit on 480 pixels high (actually it is 481 with my jwm theme but I'm sure you will cope)

Buttons designed by rastapax. No significant functional changes, I've just updated the preset buttons to show you as On the Phone when you use them.

[smokey01]

Another SIP server that works quite well

http://www.realtunnel.com/index.jsp?page=Create

I am finding proxy01.sipphone.com sometimes has problems. I have noticed lately you can listen to voice mail but you can't leave it, well you can leave it but it doesn't play the greeting message.

I also had a very jittery call with Will earlier this evening but now it's fine. The call might have been made at the same time the server was either being backed up or experiencing heavy traffic, not sure.

Anyway I have made successful calls from me on realsip.com to Will on proxy01.sipphone.com.

The new release 0.10 has been designed to work on the eeepc. I'm yet to try it on my eeepc but it works fine on my desktop PC.

[smokey01]

I just tested psip 0,10 on the eeepc.

It looks good and just fits on the screen.

The only thing that didn't seem to work was the sound. Although I had everything set at maximum in alsamixer I still couldn't hear any sound either through the speakers or headphones.

Will, I think psip is using oss and not alsa, and I think the eeepc uses alsa, could be the problem? What do you think?

Have a good break. You have earned it. I know I could do with some sleep.

[Evil20071]

Could be that I entered it in the wrong location. My review on layout only:

+ Added button labels
+ Online Indicator
+ Buddy List Integrated
- Possibly integrate the chat typing box with the log window would be nice
- Having a seperate chat window for each person would also be nice
- Clearing the chat log without having to delete the log file possible in the future?

Just got off the psip with Hairywill and can I just say WOW. Audio on my end was great, and I'm assuming that it worked well on the other end as well. Great work. Go feed the kids now before they start to really attack. lol

[Evil20071]

[HairyWill]

smokey
if the portaudio version is the problem maybe 0.9.11 will work on your eee, if so I can supply pjsua in a version that will work on your eee and drive the gui for psip-0.10

evil
it was good to talk
thanks for the +s
as far as the chatlog goes it is really a secondary feature and not intended as a full im client. I would like to integrate the windows if I can work out how but am pushing the limits of gtkdialog. As I said on the phone I'm experimenting with using gtkmoz as the chatlog viewer, which should be able to do some nice presentation if I can get past the problem of filtering out javascript.
If you want to delete bits of (or edit your log) you can just open the file in a text editor, I suppose their could be a menu entry to do this.

[smokey01]

[quote="Evil20071"]

[Evil20071]

Ok, is there a listing of everyone's SIP id? I don't have everyone.

[Lobster]

http://tmxxine.com/wik/wikka.php?wakka=PSIP

[magerlab]

i had my microfone working after recording some seconds thru it in audacity
i do not know how it is connected but now it works

[Evil20071]

[Lobster]

[smokey01]

I had a good chat with Gaz this morning. He has fixed his jitter problem. He changed his codec and now all works fine. Basically he disabled the speex codec as it was causing problems with his computer. I will let him explain the finer detail.

On another matter: The Gizmo conferencing function is now working again. Lately if you had tried it, it was refusing connections. Gaz and I had a 10 minute conference call which worked well. The quality was not quite as good as a 1 to 1 conversation but very acceptable.

I was hoping to be able to make a couple of default conference rooms/numbers but I think the numbers are dynamic. Anyway it's quite simple to setup.

Here is what you do:

The originator of the conference clicks on Presets, select Conference.
You will be asked to enter a conference number. If you don't have one click on 0 and one will be given to you. Write it down or memorize it.

Now look to see who is online. Send them a Text Chat message and tell them the conference number. They select presets > conference and type in the number and shortly after they will be connected to the conference. While you are waiting for people to join you will hear soothing music, pretty cool but try not to fall asleep

To leave the conference simply hangup. If the conference organiser hangs up I thing everyone will be disconnected.

I'm not sure how many people can be in the conference at the same time. Using Gizmo VOIP software it appeared to be limited to 4.

Smokey

[gazb]

I have had lots of problems with audio on my side(some words heard then blip blip blip) I thought it may be a codec problem or router problem as when I go to voicemail I could hear well ...the codec used was ilbc, but when talking live to people the codec used was speex, Hairwill suggested I disable speex by putting the below into my config file

--dis-codec=speex/32000
--dis-codec=speex/16000
--dis-codec=speex/8000

Audio on my side worked!
It defaulted to ilbc codec.

Further I decided to enable pcmu as the first preferred codec by inserting --add-codec pcmu as below
#
# Network settings:
#
--local-port 5060
--use-ice
--dis-codec=speex/32000
--dis-codec=speex/16000
--dis-codec=speex/8000
--add-codec pcmu
#
# Media settings:
#
--clock-rate 16000
# using default
--quality 10
# using default --ec-tail 200
# using default --ilbc-mode 20
--rtp-port 4000

I read its worth changing the clock rate to the preferred codec as well
so maybe the clock rate could be 8000 as well but have not tested.

Read more info at http://trac.pjsip.org/repos/wiki/sound-problems

Be aware that I think the codecs have to be the same both ends, if I spoke with a dialup or high latency connection , the voice would fail if it was pcmu . I think to talk to dialup you would need iLBC, or GSM both ends.