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 Forum index » Advanced Topics » Additional Software (PETs, n' stuff) » Browsers and Internet
PSip - Puppy SIP
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Evil20071


Joined: 07 Jun 2008
Posts: 425
Location: Piedmont, SC,.United States

PostPosted: 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.
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WB7ODYFred


Joined: 13 Dec 2008
Posts: 104
Location: Oregon & Washington

PostPosted: Mon 16 Feb 2009, 19:51    Post subject: How to test pjsua directly from command line?  

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.

Code:

# pjsua
bash: pjsua: command not found
# ls /usr/local
apps         lib        pburn        pmirror    psip        sbin
autoconnect  libexec    petget       pmusic     pstopwatch  seamonkey
bin          man        pfilesearch  ppg        ptimer      share
etc          net_setup  pfind        pprocess   Pwget       xfprot
jwmconfig2   pbackup    picpuz       pschedule  pwidgets
# ls /usr/local/psip
dialog_addaccount        dialog_setstatus     geany_run_script.sh    psip-gui
dialog_addbuddy          func_buildbuddytree  icons                  resources
dialog_callsipurl        func_crashreport     pjsua.cfg.default      theme
dialog_debugger          func_progressbar     pjsua_custom_03-0.9.0  usr
dialog_disconnectedcall  func_sendchat        pjsua-output-reader
dialog_incomingcall      func_shutdownpjsua   psip
dialog_inputchat         func_startpjsua      psip2
# pjsua_custom_03-0.9.0 --config-file /root/.psip/pjsua.cfg
bash: pjsua_custom_03-0.9.0: command not found
# /usr/local/psip/pjsua_custom_03-0.9.0 --config-file /root/.psip/pjsua.cfg
 15:22:43.887 os_core_unix.c pjlib 0.9.0-release for POSIX initialized
 15:22:43.887 sip_endpoint.c Creating endpoint instance...
 15:22:43.888          pjlib select() I/O Queue created (0xb7b6b098)
 15:22:43.888 sip_endpoint.c Module "mod-msg-print" registered
 15:22:43.888 sip_transport. Transport manager created.
 15:22:43.914    pjsua_app.c Too many arguments specified in cmd line/config file
#



The other problem I am researching with pjsua and psip, is that a call to
my telephone number 971-239-xxxx goes directly to voice email, it does not ring the phone. I will try using the command line version of pjsua to see if something is not set up right with my provider sip7.vitelity.net
Calling out works great.


comments and pointers most welcome. You are welcome to call and
speak with my voice mail. fredfinster@sip7.vitelity.net I would like the testing of my Voip, please.

Fred Finster
WB7ODYFred
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smokey01


Joined: 30 Dec 2006
Posts: 1815
Location: South Australia

PostPosted: Tue 17 Feb 2009, 06:21    Post subject:  

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.

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WB7ODYFred


Joined: 13 Dec 2008
Posts: 104
Location: Oregon & Washington

PostPosted: Fri 20 Feb 2009, 03:15    Post subject: Smokey01, thanks for the test of PSIP  

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
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smokey01


Joined: 30 Dec 2006
Posts: 1815
Location: South Australia

PostPosted: Fri 20 Feb 2009, 04:52    Post subject:  

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

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Aitch


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

PostPosted: Wed 25 Feb 2009, 12:00    Post subject:  

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
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x_XP

Joined: 01 Jan 2009
Posts: 31

PostPosted: Sat 07 Mar 2009, 15:42    Post subject:  

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

Last edited by x_XP on Fri 24 Apr 2009, 15:22; edited 4 times in total
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Dromeno

Joined: 12 Sep 2008
Posts: 536

PostPosted: Mon 23 Mar 2009, 07:04    Post subject: VoipCheap in PSIP
Subject description: Noob needs manual - who can help?
 

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!
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Caneri

Joined: 04 Sep 2007
Posts: 1580
Location: Canada

PostPosted: Sun 19 Apr 2009, 19:33    Post subject:  

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

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kpfuser

Joined: 19 Mar 2006
Posts: 150
Location: Mt Pelion, Greece

PostPosted: Fri 08 May 2009, 03:51    Post subject:  

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?
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WB7ODYFred


Joined: 13 Dec 2008
Posts: 104
Location: Oregon & Washington

PostPosted: Fri 08 May 2009, 20:03    Post subject: Edit PSIP configuration file /root/.psip/pjsua.cfg file
Subject description: Setting up PSIP to make Viop SIP calls to regular PSTN lines.
 


    # from file /root/.psip/pjsua.cfg
    #
    # All lines preceded with a "--" below is a valid config file command
    #
    # Logging options:
    #
    --log-file /tmp/psip/app-log
    #
    #
    # Network settings:
    #
    --local-port 5060
    # I think ice is important for maintaining routing information for those with dynamic internet ip addresses
    # ice seems to cause problems for may people
    # --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 6
    # using default --ec-tail 200
    # using default --ilbc-mode 20
    --rtp-port 4000
    #
    # Some people have experienced problems with the speex codec
    # Uncomment the next three lines to fix
    # --dis-codec=speex/32000
    # --dis-codec=speex/16000
    # --dis-codec=speex/8000
    # --add-codec pcmu
    #
    # User agent:
    #
    # --max-calls 4
    --max-calls 2
    #
    #
    # Buddies: You need to establish an account with a sip provider before you can use this
    # --add-buddy=sip:yourname@proxy01.sipphone.com
    # --add-buddy=sip:yourname@realsip.com
    #
    # Account 0:
    # --id sip:yourname@proxy01.sipphone.com
    # --registrar=sip:proxy01.sipphone.com
    # --realm *
    # --username=yourname
    # --password=password
    # --reg-timeout=55
    #
    # Account 0:
    --id sip:fredfinster@sip7.vitelity.net
    --registrar=sip:sip7.vitelity.net
    --realm *
    --username=fredfinster
    --password=xxxxxx # get the password from your SIP line provider
    --reg-timeout=55
    #
    # Account 1: The "--next-account" line is required for each additional account
    # some where in the front or tail of the file you can hand edit add a buddy. or use PSIP button 'add buddy'
    --add-buddy sip:86@sip7.vitelity.net
    --add-buddy sip:13604031234@sip7.vitelity.net
    --add-buddy sip:15038511234@sip7.vitelity.net
    --add-buddy sip:15035917890@sip7.vitelity.net
    --add-buddy sip:15416725555@sip7.vitelity.net
    --add-buddy sip:fredfinster@sip7.vitelity.net
    --add-buddy sip:19712390140@sip7.vitelity.net

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

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.

Code:

./pjsua_custom_03-0.9.0 

or

# /usr/local/psip/pjsua_custom_03-0.9.0 --config-file /root/.psip/pjsua.cfg
 15:22:43.887 os_core_unix.c pjlib 0.9.0-release for POSIX initialized
 15:22:43.887 sip_endpoint.c Creating endpoint instance...
 15:22:43.888          pjlib select() I/O Queue created (0xb7b6b098)
 15:22:43.888 sip_endpoint.c Module "mod-msg-print" registered
 15:22:43.888 sip_transport. Transport manager created.
 15:22:43.914    pjsua_app.c Too many arguments specified in cmd line/config file
#
#  Note, can not have more than 20 (twenty) --add-buddy lines in the pjsua.cfg file.
#


Hope this helps Kpfusr.

Testing from the command line, will help to see the hidden error messages
that get lost behind the GUI interface.

I did not setup a proxy or use STUN. I can dial out to regular telephones and when I have PSIP up and running people can call me, because I have a $35 account setup
with Vitelity, a SIP to PSTN provider.
http://www.vitelity.net


Give my SIP address a try for testing your SIP phone Kfpusr.
the telephone number will have to be something like sip:16785551212@xxx,sipphone.com Or whatever your SIP provider gives you.
Using the upper left hand call button.
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kpfuser

Joined: 19 Mar 2006
Posts: 150
Location: Mt Pelion, Greece

PostPosted: Sat 09 May 2009, 12:49    Post subject:  

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.
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zozu

Joined: 29 Jul 2011
Posts: 3

PostPosted: Fri 29 Jul 2011, 08:20    Post subject:  

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?
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smokey01


Joined: 30 Dec 2006
Posts: 1815
Location: South Australia

PostPosted: Fri 29 Jul 2011, 08:44    Post subject:  

zozu wrote:
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?


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.

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zozu

Joined: 29 Jul 2011
Posts: 3

PostPosted: Fri 29 Jul 2011, 10:50    Post subject:  

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.'
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