PSip - Puppy SIP

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Aitch
Posts: 6518
Joined: Wed 04 Apr 2007, 15:57
Location: Chatham, Kent, UK

#201 Post by Aitch »

HairyWill/Smokey/Lobster

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

http://www.freeworlddialup.com/learnmor ... rs#peering

tip from here

http://bobsbasement.co.uk/Asterisk_box

Aitch :)

x_XP
Posts: 31
Joined: Thu 01 Jan 2009, 04:57

#202 Post by x_XP »

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, 19:22, edited 4 times in total.

Dromeno
Posts: 534
Joined: Fri 12 Sep 2008, 07:01

VoipCheap in PSIP

#203 Post by Dromeno »

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
Posts: 1513
Joined: Tue 04 Sep 2007, 13:23
Location: Canada

#204 Post by Caneri »

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
[color=darkred][i]Be not afraid to grow slowly, only be afraid of standing still.[/i]
Chinese Proverb[/color]

kpfuser
Posts: 207
Joined: Sun 19 Mar 2006, 15:02
Location: Mt Pelion, Greece

#205 Post by kpfuser »

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
Posts: 169
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Location: Oregon & Washington

Edit PSIP configuration file /root/.psip/pjsua.cfg file

#206 Post by WB7ODYFred »

  • # 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: Select all

./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.

kpfuser
Posts: 207
Joined: Sun 19 Mar 2006, 15:02
Location: Mt Pelion, Greece

#207 Post by kpfuser »

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
Posts: 3
Joined: Fri 29 Jul 2011, 12:00

#208 Post by zozu »

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
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#209 Post by smokey01 »

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.

zozu
Posts: 3
Joined: Fri 29 Jul 2011, 12:00

#210 Post by zozu »

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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smokey01
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Location: South Australia :-(
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#211 Post by smokey01 »

zozu wrote: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.

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.'
it said Call has been disconnected. Temporarily unavailable. That's because I had gone offline. I need to be logged into my sip server for you to see me.

The good thing about psip is you don't have to be logged into a server to use it. For example: if you have Psip running and I know your IP address I could add it as a buddy and call you direct.

The buddy address entry would be something like sip:192.168.0.6

This is really good if you have a static IP address, not dynamic as it continually changes each time you log on.

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Flash
Official Dog Handler
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Location: Arizona USA

#212 Post by Flash »

smokey01 wrote:...The good thing about psip is you don't have to be logged into a server to use it. For example: if you have Psip running and I know your IP address I could add it as a buddy and call you direct.

The buddy address entry would be something like sip:192.168.0.6

This is really good if you have a static IP address, not dynamic as it continually changes each time you log on.
I've wanted to try that, emailing my DHCP-assigned IP address to my brother after I've booted my computer for the day.

It is my understanding that a "client" program running in Puppy, such as Psip, won't accept a connection it didn't initiate. If I understand correctly, that's what makes it necessary to log into the intermediate server. Will Psip, running in my computer, somehow get around that if I try to make a direct connection with my brother? Do we have to attempt to connect with each other at pretty much the exact same time (so that each one thinks it initiated the connection?)

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smokey01
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#213 Post by smokey01 »

Flash, it suppose to be able to do it. I've been thinking about making a very simple P2P version but I am having a similar problem.

Here is the manual http://www.pjsip.org/pjsua.htm

In particular here http://www.pjsip.org/pjsua.htm#basic

I can make it work within my own network eg: 192.168.0.3 etc but how to make it work on the internet is the issue.

I know what my dynamic IP address is but how do I make it translate to my internal network address? This is the problem I'm having.

I guess NAT should be able to sort it but I'm not sure how to define it in the config file.

Maybe we can work together to sort it out.

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technosaurus
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#214 Post by technosaurus »

This is kinda an old thread, is psip still being maintained?
Check out my [url=https://github.com/technosaurus]github repositories[/url]. I may eventually get around to updating my [url=http://bashismal.blogspot.com]blogspot[/url].

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smokey01
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#215 Post by smokey01 »

technosaurus wrote:This is kinda an old thread, is psip still being maintained?
It hasn't been for some time but I am considering doing some work on it.

I would like to get it working as a proper peer 2 peer VOIP application without using a sip server. The engine, pjsua, says it can be done as described in the docs in my last post.

I might even rework the GUI but I might need some help.

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Flash
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#216 Post by Flash »

smokey01 wrote:...I can make it work within my own network eg: 192.168.0.3 etc but how to make it work on the internet is the issue.

I know what my dynamic IP address is but how do I make it translate to my internal network address? This is the problem I'm having.

I guess NAT should be able to sort it but I'm not sure how to define it in the config file...
All I can contribute are some general thoughts based on my understanding of how the internet transmission protocol works. :lol:
When you enter an IP address in a browser window and click "go", an IP packet is sent from your computer through your router to the server at the internet address you entered. Your router has to somehow keep track of where that packet originated in your network so it can forward the reply back to the originating program (the browser.) Ports come into it somewhere, but I think at the very end, after your computer has received the packet from the router. The same general description must apply for VOIP applications such as Psip.

I think VOIP uses a different protocol from TCP, called UDP. UDP is the same protocol used for streaming internet video. UDP is used because it does not require the destination to acknowledge receipt of each packet, saving time and wear and tear on the network. :) Some packets might never make it, but for many applications such as VOIP, that's okay; you just hear a bit of noise.

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Aitch
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#217 Post by Aitch »

Smokey

Try dynamicDNS - it allows a fixed IP Address of your dynamically assigned one

http://www.dyndns.com/services/dns/dyndns/

I found PSIP to be unreliable, from memory, I only ever made brief connections to Eric, HairyWill and Lobster, I think

I now use skype, but it uses huge resources for what it gets used for.... :(

Aitch :)

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smokey01
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#218 Post by smokey01 »

Aitch I've just heard from Benny, the author, who has given me a couple of avenues to explore.

I'm surprised that Psip was unreliable for you as Will and myself spent many hours communicating with it while we were in development.

Anyway I will also investigate your suggestion.

Thanks

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smokey01
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#219 Post by smokey01 »

Aitch maybe you should try a free sip account from https://www.ekiga.net/

Your account in Psip would look something like:

--id sip:Aitch@ekiga.net
--registrar sip:ekiga.net
--realm *
--username Aitch
--password xxxxxx
--reg-timeout 55

Let me know a suitable time and we could test.

I'm GMT +9.5 hours.

tide
Posts: 56
Joined: Fri 13 Aug 2010, 04:33

#220 Post by tide »

Could some kind soul please please tell me how to configure PSIP?
I signed up with sipgate.co.uk but after trying for hours, google, forumsearch etc. just can't get it to work!

Sipgate tells me that:

Code: Select all

SIP-ID: 	               134xxxx 
SIP password: 	9xxxx 
Status: 	                offline 
Nickname:  No name was set 	Edit 

Registry: 	sipgate.co.uk (Port: 5060) 
Proxy: 	sipgate.co.uk (Port: 5060) 
NTP: 	ntp.sipgate.net 

SIP password: 	9xxxx
:?

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