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 Forum index » Advanced Topics » Cutting edge
VOIP
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Caneri

Joined: 04 Sep 2007
Posts: 1580
Location: Canada

PostPosted: Sun 28 Sep 2008, 10:35    Post subject:  

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

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HairyWill


Joined: 26 May 2006
Posts: 2949
Location: Southampton, UK

PostPosted: Sun 28 Sep 2008, 11:26    Post subject:  

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

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Béèm


Joined: 21 Nov 2006
Posts: 11782
Location: Brussels IBM Thinkpad R40, 256MB, 20GB, WiFi ipw2100. Frugal Lin'N'Win

PostPosted: Sun 28 Sep 2008, 19:45    Post subject:  

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?

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Lobster
Official Crustacean


Joined: 04 May 2005
Posts: 15117
Location: Paradox Realm

PostPosted: Mon 29 Sep 2008, 04:03    Post subject:  

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:
# ./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"

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Caneri

Joined: 04 Sep 2007
Posts: 1580
Location: Canada

PostPosted: Mon 29 Sep 2008, 08:53    Post subject:  

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

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smokey01


Joined: 30 Dec 2006
Posts: 1795
Location: South Australia

PostPosted: Mon 29 Sep 2008, 17:53    Post subject:  

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

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Béèm


Joined: 21 Nov 2006
Posts: 11782
Location: Brussels IBM Thinkpad R40, 256MB, 20GB, WiFi ipw2100. Frugal Lin'N'Win

PostPosted: Mon 29 Sep 2008, 17:57    Post subject:  

Smokey, thanks for the confirmation.

Hope Gaz can put me on track.

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gazb

Joined: 09 Jul 2008
Posts: 25
Location: Australia Sydney

PostPosted: Mon 29 Sep 2008, 20:13    Post subject: call out to pstn  

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/viewtopic.php?t=31061&start=120

You have to add county codes etc..the help section at gizmo5 would help.
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Béèm


Joined: 21 Nov 2006
Posts: 11782
Location: Brussels IBM Thinkpad R40, 256MB, 20GB, WiFi ipw2100. Frugal Lin'N'Win

PostPosted: Tue 30 Sep 2008, 16:14    Post subject:  

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?

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Trobin

Joined: 18 Aug 2005
Posts: 907
Location: BC Canada

PostPosted: Tue 07 Oct 2008, 19:20    Post subject:  

In Puppy 4.2, is there a way to access psjua from the command line?
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Lobster
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Joined: 04 May 2005
Posts: 15117
Location: Paradox Realm

PostPosted: Wed 08 Oct 2008, 00:17    Post subject:  

Quote:
In Puppy 4.2, is there a way to access psjua from the command line?


Trobin = time traveler Smile
You are using 4.1? In which case . . .

Phone / restart in CLI mode from the pull down menus

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Caneri

Joined: 04 Sep 2007
Posts: 1580
Location: Canada

PostPosted: Tue 23 Dec 2008, 15:16    Post subject:  

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

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smokey01


Joined: 30 Dec 2006
Posts: 1795
Location: South Australia

PostPosted: Tue 23 Dec 2008, 16:47    Post subject:  

Eric, it doesn't work for me either.

I even registered another Gizmo logon and that didn't work either.

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Caneri

Joined: 04 Sep 2007
Posts: 1580
Location: Canada

PostPosted: Tue 23 Dec 2008, 17:00    Post subject:  

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

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peppyy


Joined: 27 Jun 2005
Posts: 429
Location: VT USA

PostPosted: Mon 29 Dec 2008, 23:41    Post subject:  

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 Wink

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 Wink

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