Puppy Linux Discussion Forum

Evil20071 · Posted: Wed 31 Dec 2008, 00:22 Post subject:

Actually, was planning on putting a note in for someone to re-package it with my sip url, or to add in the presets for like, conference calling/voicemail that are in the wiki to the config file so that's no longer an issue.

WB7ODYFred · Joined: 13 Dec 2008 Posts: 90 Location: Oregon & Washington

Questoins on puppylinux 4.1.1 how do you directly call pjsua with out using psip.

Here is how I do it. I notice that I cannot have more than 20 --addbuddy lines in the /root/.psip/pjsua.cfg file.

smokey01 · Joined: 30 Dec 2006 Posts: 1605 Location: South Australia

Fred, I tried to call you on fredfinster@sip7.vitelity.net but your address could not be found. It did not give me a chance to leave you a voicemail.

I did receive your voicemail the other day but I haven't had time to reply, sorry.

If you create an account on proxy01.sipphone.com we might have more luck.

WB7ODYFred · Joined: 13 Dec 2008 Posts: 90 Location: Oregon & Washington

I am away from my puppy computer for a few days.

Do you know how to run pjusa from the command line? I want to see
what others are using.

is
/usr/local/psip/pjusa-0xxx --config-file=/root/.psip/pjsua.cfg

Is it something like that, Smokey01? I could find no "pjsua" command in
stock 4.1.1 install. I have not figured out / tested if I can answer the phone from PSIP, when my landline number 971-239-xxxx is called.

Thanks again for everybodies testing. I was using www.vitellity.net for
playing with www.PBXinaflash.net a Asterisk type PBX. Then found puppy 4 and PSIP, so made somechanges to my Vitelity account and low and behold PSIP worked to call out from inside my 18 wheeler truck to the truck stop WiFI. I was calling back home to Seattle Washington from Salt lake City Utah. That was neat. I thought testing with something other than sipphone might uncover little problems. (like not being able to answer the telephone)

Fred

smokey01 · Joined: 30 Dec 2006 Posts: 1605 Location: South Australia

Fred,

Try this:

# ./pjsua_custom_03-0.9.0

From the console in the /usr/local/psip directory.

Also the help menu will direct you to many of the command line options.

http://www.pjsip.org/pjsua.htm

Hope this helps

Aitch · Joined: 04 Apr 2007 Posts: 6825 Location: Chatham, Kent, UK

HairyWill/Smokey/Lobster

Would this service be of any use to the PSip team?

http://www.freeworlddialup.com/learnmore/?p=features&s=accessnumbers#peering

tip from here

http://bobsbasement.co.uk/Asterisk_box

Aitch Smile

x_XP · Joined: 01 Jan 2009 Posts: 31

can someone help me here.
I am using puppy 4.1. and I downloaded latest PSIP, installed and configured but it will not register to my SIP provider. There is no activity in the PSIP window I do not have a clue why doesn't work for me.
I did use successfully zioper and x-lite in puppylinux but they are non recording softphone.
my PSIP debug crash report
http://pastebin.ca/1400624
and my config file
http://pastebin.ca/1400622
perhaps someone can give me an idea what am I doing wrong ?
btw most important function for me in VoIP is call recording, can I get help how to set this up as well?

thanks mike

Dromeno · Joined: 12 Sep 2008 Posts: 469

A housemate asked me to figure out if/how PSIP can work with http://www.voipcheap.com

but I am totally lost.

I can not even get PSIP to work when I try to follow the wiki. That document suggests to try a Gizmo account first.

What I have is a Gizmo Project name, a SIP number plus Proxy server, proxy server Port and of course an email adress

But now PSIP asks me for a SIP url, Registrar URL, Username, password, Auth Realm.

I am too much a noob here, i do not get it. Can somebody please refer me to the right manual document?

Gizmo is nice (the it at least should work) but the goal in our house is to make it work for VoipCheap - if anyone can help it would be greatly appreciated!

Caneri · Joined: 04 Sep 2007 Posts: 1565 Location: Canada

Hi psip fans,

Well Irihapeti and myself are trying a phone call and there seems to be no connection via voice or chat although the sip server does complete the presets voice lady.(sip server?)

I'm on Aragons 4.2smp and tried with an old 4.09 install using 0.12 and even went back to ps7 with no joy.

How about we start a new test run for psip...hey, it's always good to chat live anyway so let's get some voice going again to say hello and meet face to face so to speak.

Any takers?
Eric

kpfuser · Joined: 19 Mar 2006 Posts: 104 Location: Mt Pelion, Greece

Since this is supposed to be the "official" PSIP thread, I have to give it a try in the hope of solving my mounting problems with trying to make VOIP calls using PSIP.

I am a beginner with PSIP (and VOIP calls as well) and my present situation is as follows:

I have opened an account with sipphone.com. When I asked for info on what to enter to PSIP I was given the following:

* Proxy Server (Host): proxy01.sipphone.com
* Proxy Server (IP): 198.65.166.131
* Registration Server: proxy01.sipphone.com
* SIP Port (UDP): 5060
* STUN Server: stun01.sipphone.com
* STUN Port: 3478
* User ID: 17471115555 (replacing with your 1747)
* Auth ID: 17471115555 (replacing with your 1747)
* Password: (your gizmo5 password)

Going to PSIP --> Configure --> Edit Account I filled the available fields as follows:

Your SIP URL: sip: kpfuser@proxy01.sipphone.com
Registrar URL: sip: proxy01.sipphone.com
Auth Realm:(*) kpfuser
Username: kpfuser
Password: ***********(my password)

Subsequently i added one buddy (corresponding to a landline) as follows:

sip: 00(country code)(area code)#####

Then with Pjsua logged in I tried to place my first call by selecting my buddy and clicking on "Call." A small popup indicating "0, CALLING" came up but the speakerphones remained silent. Finally a new popup saying "Call has been disconnected: Request Timeout" came up and that was that. Following this I could get no sound from the sound card unless I quit PSIP from the "Phone" menu and restart any open application that uses the sound card, e.g., Xine Player.

Can anyone help me get around this?

WB7ODYFred · Joined: 13 Dec 2008 Posts: 90 Location: Oregon & Washington

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

2nd look about you can start the PSIP application from the command line in an open RSVT terminal. start ---> Utiltity ----> RXVT terminal

pwd /root
cd /usr/local/psip
pwd /usr/local/psip
ls -l pjs*
# execute the PSIP command manual to see what errors show up.

kpfuser · Joined: 19 Mar 2006 Posts: 104 Location: Mt Pelion, Greece

WB7ODYFred,

Thank you very much indeed for your reply! Please correct me if I am wrong, but I do not recall seeing the sort of info you posted in any PSIP or similar manual. Does this mean that the info is self-evident to the majority of newbies that try making calls using PSIP or that the manuals etc. leave a bit to be desired?

Anyway, before implementing your configuration, I would like to clarify something first. It follows from your info (as well as from other posts) that PSIP will probably behave better for a static ip address. But what kind of static ip address does this imply, a static ip address given to you by your ISP or an internal static ip address one can assign to his/her pc behind a router via the Puppy network setup wizard? If the former is the case, then I will have to live with the limitations of a dynamic ip address. If, however, the latter applies, it would be a simple matter to assign a static ip address to my computer each time I boot up.

zozu · Joined: 29 Jul 2011 Posts: 3

I am new to Puppy, but I am getting amazed. I tried to configure PSIP 0.12 in lupu525 but I cannot get it work. I have a SIP account with a SIP provider, called jumblo. This is their SIP settings
http://www.jumblo.com/en/sipp.html
If I click to Configure/Edit Account to enter the detail for my SIP account, it does not work and I tried all combinations.r
The numpad stays greyed out and I cannot enter any number to call.
What do I do wrong?

smokey01 · Joined: 30 Dec 2006 Posts: 1605 Location: South Australia

zozu · Joined: 29 Jul 2011 Posts: 3

Try this:
--id sip:username@sip.jumblo.com
--registrar sip:sip.jumblo.com
--realm *
--username username
--password password
--reg-timeout 55

What is your sip address? (zozu@sip.jumblo.com)

Add me to your buddy list at smokey01@realsip.com

If you see me online, give me a call to test.[/quote]

I tried but still the same. I added you to my buddy list, tried to call you and it said. 'Call has been disconnected. Temporarily unavailable.'