Puppy Linux Discussion Forum

Evil20071 · Posted: Mon 08 Dec 2008, 01:52 Post subject:

Had to re-burn it... I think I had an issue downloading it, not sure though... trying it again tonight if I am able to get the Live CD to boot. New version, of course.

Evil20071 · Posted: Tue 09 Dec 2008, 22:27 Post subject:

I need a hand with the in-built PSIP of 4.1.2. It's saying it's loading and everything, and in CLI mode, it returns no errors, but it's not connecting, from what I can tell.

Suggestions?

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

Evil20071,

I'm not quite sure what you are asking but I will attempt to answer your question. I am currently using puppy 4.1.1.

If you want Psip to login you need to specify a server location in the config file. I am assuming you have already created an account on proxy01.sipphone.com. This is the server most of us use.

Assuming you used abcde as your username and 12345 as your password the below part of the config should look like this.

# Account 0:
-id sip:abcde@proxy01.sipphone.com
--registrar=sip:proxy01.sipphone.com
--realm *
--username=abcde
--password=12345
--reg-timeout=55

The above lines need to be in your config file which is called pjsua.cfg. The file can be found in ./psip

This will register you on the proxy01.sipphone.com server. If another person has your address (abcde@proxy01.sipphone.com) added to their buddies list they will see when you are online and they can call you.

You don't actually have to be logged into a server to make calls. Providing you know the IP address of the person you want to call, and they are online, you can call them. Very much peer to peer.

If you haven't already done so select Help > Help from the menu within Psip. This will give you most of the information you need to know. If not, repost here and I and others will offer more advice.

Evil20071 · Posted: Fri 12 Dec 2008, 01:32 Post subject:

Um, yeah. I was kinda the one that originally posted the registration link. Razz

Also, I already have all of that flled in but it just freezes on "Starting PJSUA" and never loads past that point. This sat is my b-day, and my sister is having a baby tomorrow, so I'll be busy. Monday or Tuesday when I get the chance to take my puppy out for a walk again, I'll screenshot PSIP when I try and load it...

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

I realised who you were within seconds after posting. Embarassed

Evil20071 · Posted: Sat 13 Dec 2008, 01:02 Post subject:

That's fine. Here's what it gets stuck at:

If you would like to see the debugging report, I will post it, but idk if it'll help any. I looked it over and can't find anything. I double checked my settings as well and they match.

EDIT: I just noticed my Config file doesn't have all of the contacts on it now. (ODD Much?) That may well be the issue.

Evil20071 · Posted: Sat 13 Dec 2008, 01:26 Post subject:

Ok. Figured it out. Was a config file issue. lost my buddies for some reason. I added them in again and it's fine now.

prettyreplica · Joined: 14 Dec 2008 Posts: 1

It seems very good

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

Hello PSIP Friends,

I am detailing what worked for me with my SIP provider vitelity.net to make outbound PSTN Calls in the USA.

Notes #1)

Dialing a SIP phone number for somone using a different SIP provider Page 9 top by Author Gazb
http://www.murga-linux.com/puppy/viewtopic.php?t=31061&start=120

http://sipbroker.com/sipbroker/action/providerWhitePages

lookup the code for us.voxalot.com which is *010
make sure the green tick is on, this means the service is available.

then in psip open Phone>CallSIP URL

in the box put the code(*010) then friends number(eg 123456789) then @sipbroker.com

eg sip:*010123456789@sipbroker.com

click OK and the phone will ring the number.

Notes #2)

Dialing PSTN numbers Page 8 by Author CEL question by Dogone
http://www.murga-linux.com/puppy/viewtopic.php?t=31061&start=111

CEL wrote
sip:###########@proxy01.sipphone.com

where the #s are the phone number to call, including country code.

This is assuming your SIP account is from Gizmo/SIPphone… With other SIP providers, you would use their registrar as the domain instead of proxy01.sipphone.com.

Note that I clicked on menu Configure --> Add Buddy to open a window, enter the 10 digit PSTN number followed by the sip registrar SIP name hit enter. Click on Refresh Buddys list and there it is in the Buddys list. Click on the Buddy entry to Highlight, Then click the green "CALL" button, phone should start riniging. Yahooo!! its working now.

Below is pjsua.cfg file.

--add-buddy sip:14253460202@sip7.vitelity.net
--add-buddy sip:15035911234@sip7.vitelity.net
#
# 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
--quality 4
# 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
#
# --id sip:yourname@proxy01.sipphone.com
# --registrar=sip:proxy01.sipphone.com
# --realm *
# --username=yourname
# --password=password
#
# Account 1: The "--next-account" line is required for each additional account
# --next-account
# --id=sip:yourname@realsip.com
# --registrar=sip:realsip.com
# --realm *
# --username=yourname
# --password=password
# Account 0:
--id sip:fredfinster@sip7.vitelity.net
--registrar sip:sip7.vitelity.net
--realm *
--username fredfinster
--password xxxxxxx
--reg-timeout 55

# End of pjsua.cfg file.

~~~~~~~~~~~~~ Comments Below ~~~~~~

I would start PSIP 0.12 without any Buddys in my Buddy list.
That Black message at the Bottom, would say "PJSUA is starting application" Nothing would change. I was looking for a message saying that PSIP had register itself with my SIP provider. So that I would know that the PJSUA.cfg file was setup correctly and working. But nothing in the user interface gave me indication that PSIP had registered correctly.

After I added my first Buddy, it change to read and said "PJUSA Logged in, Hti Refresh, to see your Buddyst"
After clicking and refreshing Buddy list, I could see my Buddys and that
"PJSUA Logged in" in red was telling me I was operatoinal.

Other problem I will work on next. Is I can call out fine, but a call back to
my SIP DID number, gets a busy signal. So nobody can call in, just yet.
Will check through my vitelity account settings to see if something else is a miss. Great Audio, Thanks GUYS for this wonderful PSIP software.
I am sitting in my 18 Wheeler Semi-truck, waiting for a load Monday with a desktop computer in the cab, and you have made me VERY HAPPY to be able to call out to PSTN phones!

Fred Finster fredf at KENDRA dot COM

ps. How could I create a HOWTO document to help others setup PSIP with a provider other than sipphone.com. Hopefully, the above is a good start.
ps. which page in this List has that default buddys list for testing. OR Is it in the WIKI?

pss. I can seem to highight a section of text and hit the "CODE" button to make it change from standard text. So I was unable to highlight and organize this post better. Sorry :<(

Lobster · Posted: Sun 14 Dec 2008, 18:31 Post subject:

Glad to hear of your success Smile

You can add a test to the HOWTO section of the forum

Add a Howto here or modify/improve here
http://www.puppylinux.org/wiki/applications/dot-pets/psip

dogle · Joined: 11 Oct 2007 Posts: 302

Fred, welcome and thanks for your excellent post. I'm so glad to have a full specimen working config for checking purposes.

Lobster is right, we do need better documentation if PSip is to be utilised as it deserves to be. Please count me in if I can help with this, even just as a copyreader/guinea pig.

I've struggled for ages trying to get PSip working in 3.01; at last I'm nearly there in 4.1.2, but still trying to figure out what was going wrong.
I hope to be able to post some info tomorrow, and should anyone want to try calling my [latest] account identity is dogle3.

[edit: corrected version number]

Evil20071 · Posted: Sun 14 Dec 2008, 22:52 Post subject:

Might I suggeest the next version of PSIP be packaged as the original was, which included the friends list for currently known puppy/PSIP user SIP id's. Having 0.12 including no friend list built in causes that error, and including it would make it an easy fix. I believe that a prompt for user account details at the first run of PSIP should be able to be coded in? Just a thought.

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

Guys the problem is that PJSUA does not handle error checking very well regarding the configuration file.

Any typo's will cause a problem. This seems to be more sensitive in the later versions of puppy. Maybe it's just my imagination.

There must be a blank line at the bottom of the configuration file. All of these issues were discovered and fixed when we built this application.

It's probably best to make the configuration file as small as possible. In other words get rid of all the commented lines. These are the lines that start with #. Most of the commented lines in the default pjsua.cfg (configuration) file were put there to help people get started.

If you get stuck there is quite a bit of help in these forums. It's worth reading them and you will see the progress that was made. It will also give you an inside into the issues that were discovered and solved.

Anyway, once you get it working you will find the quality is very good. The delay is almost unnoticeable. Benny, The guy who wrote PJSUA did a great job. Hairywill who did most of the coding and compiling also did a great job.

There were plans to development Psip even further however, competing priorities and time is always a challenge.

Smokey

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

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

Here is another show stopper that I just discovered.

When you are adding buddies and don't finish properly you may end up with something like this in your configuration file:
--add-buddy=sip:

You need to either finish this line with a user address or delete it as it will stop Psip from working.

As I and hairywill have said earlier, the error checking needs work.

If your eye for detail is good you should be ok.

Smokey