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Posted: Sun 28 Sep 2008, 14:35
by Caneri
Ok...using the new 4.1rc and psip 0.12 August 8 2008 version with call quality set to 10 and max calls set to 4 works for me..hooray!! I don't see anyone online however but maybe none are online..dunno.
I called Grant and got him on his cell phone..ooops...I wondered why the rings sounded like the British double ring...lol
Overall my setup works well and vm messages are getting forwarded to email..hooray again.
I have posted a few vm messages to Lobster,Grant, Will and myself for testing so if you don't gt them let me know
Best,
Eric
Posted: Sun 28 Sep 2008, 15:26
by HairyWill
eric,
I got two messages from you
the first was sent wiith 0.12 but had very poor audio quality
the second with 0.11 which sounded much better, OK
edit just received one today from 0.12, exellent call quality, cheers
Posted: Sun 28 Sep 2008, 23:45
by Béèm
I have now a telephone facility via internet with my provider.
This is done through my rooter/adsl modem which has voip support.
Could I use that SIP domain and userid with this program?
Posted: Mon 29 Sep 2008, 08:03
by Lobster
I am glad Eric has PSIP working
- it is not working for me
No doubt it is some setting
I have forgotten to set in the config file
I am having to go back to pjsua from command line . . .
Code: Select all
# ./psip2
/usr/local/psip/func_startpjsua: line 43: 13482 Broken pipe tail -f $PJSIGNAL >$PJTMP/pjsua-input-pipe
/usr/local/psip/func_progressbar: line 3: 13957 Broken pipe cat "$PJTMP/status-message"
/usr/local/psip/func_progressbar: line 3: 13974 Broken pipe cat "$PJTMP/status-message"
/usr/local/psip/func_progressbar: line 3: 13988 Broken pipe cat "$PJTMP/status-message"
/usr/local/psip/func_progressbar: line 3: 14019 Broken pipe cat "$PJTMP/status-message"
/usr/local/psip/func_progressbar: line 3: 14033 Broken pipe cat "$PJTMP/status-message"
/usr/local/psip/func_progressbar: line 3: 14050 Broken pipe cat "$PJTMP/status-message"
/usr/local/psip/func_progressbar: line 3: 14064 Broken pipe cat "$PJTMP/status-message"
Posted: Mon 29 Sep 2008, 12:53
by Caneri
Hi Lobster,
This below is what I'm using along with speex...some people can't use speex so maybe look there. Also at one time I had to delete ALL psip files including the tmp/psip directory to start anew...this worked.
Also remember to leave one line empty at the bottom of the cfg file.
My partial settings are as follows
#
# Network settings:
#
--local-port 5060
#I think ice is important for maintaining routing information for those with dynamic internet ip addresses
#--use-ice
#I think this might only work if you have a gizmo account
#--stun-srv=stun01.sipphone.com
#
# Media settings:
#
# using default --clock-rate 12000
--quality 10
# using default --ec-tail 200
# using default --ilbc-mode 20
--rtp-port 4000
#
# User agent:
#
--max-calls 4
Best,
Eric
Posted: Mon 29 Sep 2008, 21:53
by smokey01
Béèm wrote:I have now a telephone facility via internet with my provider.
This is done through my rooter/adsl modem which has voip support.
Could I use that SIP domain and userid with this program?
Beem, that would be a Yes.
gazb uses this type of setup all the time.
Gaz if you see this maybe you can explain your setup to beem.
Smokey
Posted: Mon 29 Sep 2008, 21:57
by Béèm
Smokey, thanks for the confirmation.
Hope Gaz can put me on track.
call out to pstn
Posted: Tue 30 Sep 2008, 00:13
by gazb
Béèm wrote:I have now a telephone facility via internet with my provider.
This is done through my rooter/adsl modem which has voip support.
Could I use that SIP domain and userid with this program?
There is some info here
http://www.murga-linux.com/puppy/viewto ... &start=120
You have to add county codes etc..the help section at gizmo5 would help.
Posted: Tue 30 Sep 2008, 20:14
by Béèm
Gazb
I am very new to this stuff.
I did come that far that the whitebook did found my provider *8260 voip.belgacom.be, but as there is a red sign with a little cross, I understand Belgacom is refusing incoming calls.
I don't know if it is the same for outgoing calls.
To have the gizmo help you talked about, do I have to install another program then psip?
Posted: Tue 07 Oct 2008, 23:20
by Trobin
In Puppy 4.2, is there a way to access psjua from the command line?
Posted: Wed 08 Oct 2008, 04:17
by Lobster
In Puppy 4.2, is there a way to access psjua from the command line?
Trobin = time traveler
You are using 4.1? In which case . . .
Phone / restart in CLI mode from the pull down menus
Posted: Tue 23 Dec 2008, 19:16
by Caneri
Hi all you Voip/Psip users.
We (Prit and myself) are running a test of Gizmo from a web page.
Look here
http://www.puppylinux.asia/
So far it does not work for me but maybe a few testers will help.
To all the psip/voip users..sorry I have been away. After Christmas I would like to get back into the voice contacts again....it was a great time to talk to all the people using Psip, Skype and Gizmo.
Merry Christmas and Happy New Year.
Eric
Posted: Tue 23 Dec 2008, 20:47
by smokey01
Eric, it doesn't work for me either.
I even registered another Gizmo logon and that didn't work either.
Posted: Tue 23 Dec 2008, 21:00
by Caneri
Hiya Grant,
Long time no psee (psip see)
Well I'll let it go for a bit then put it in the biz bag (biz bag was a commercial on TV years back over here).
Thanks smokey....we'll chat soon.
Eric
Posted: Tue 30 Dec 2008, 03:41
by peppyy
Caneri wrote:Hi all you Voip/Psip users.
We (Prit and myself) are running a test of Gizmo from a web page.
Look here
http://www.puppylinux.asia/
So far it does not work for me but maybe a few testers will help.
To all the psip/voip users..sorry I have been away. After Christmas I would like to get back into the voice contacts again....it was a great time to talk to all the people using Psip, Skype and Gizmo.
Merry Christmas and Happy New Year.
Eric
And the same to you
I was able to get the echo test voice to play at half speed and the test came back but Very ch-o-py . Interesting project
Posted: Tue 30 Dec 2008, 13:47
by Caneri
Hey Peppyy,
This gizmo thingy is still early beta so it may take awhile for it to come online fully.
I still can't login...it just sits there trying to connect...hmmm
Happy New Year..have a beer (replace with beverage of your choice here ) for me...
Eric
caneri@proxy01.sipphone.com
Posted: Tue 30 Dec 2008, 19:58
by Aitch