(old)Puppy Linux Discussion Forum

This computer is quite old (7 years ish) and in fairness is short on RAM but I was hoping I might be able to get it to work. Might try on a higher spec computer later.

If I want to try 0.26 should I uninstall 1.3 first? Would prefer to stick with 1.3 if possible though.

OS is Lucid 5.2.8-004
Have read the Help file now and think I do pretty much understand everything.
Wonder if some sort of codec management should be included, disable, order of preference perhaps.

At the moment I can't log in with any sip account or any client, even in windows, so beginning to suspect my ISP is restricting something.

Installing Psip 0.26 will not affect your 1.3 installation as it completely installs to a different directory.

You will end up with two menu entries under Internet but they are different icons. You will be able to run each Psip separately, just don't run them at the same time.

Also if you use 0.26 make sure you shut it down properly from quit on the drop down menu or it may to continue to run in memory.

OscarTalks try this:

http://smokey01.com/Tman/apps/Multiple_ ... Wizard.pet

I think it tells puppy to use a specific sound card and isolate the others. I haven't tried it but others have with success.

After losing all sip connectivity while experimenting I switched everything off and went for a walk to clear my head. With the problem on all computers and all clients and all accounts I thought it must be either the ISP or the router.

I rebooted the router and could log in again so at least that was something.

Then I hooked up my new, high-spec computer but found that the symptoms were pretty much the same. Iptel utilities not working and call drops after exactly 32 seconds on calls to certain domains including iptel and alcazarnetworks. Calls to some other domains were fine though.

Still thinking the problem must be related to the router I UNTICKED the "enable SIP ALG" option in the router's advanced settings and found that iptel utilities suddenly burst into life and the calls didn't drop after 32 seconds any more

BUT

Some of the other calls that I was able to make before were now not working. I was getting a connection but it wasn't resolving the audio paths so I could hear nothing.

Tried ice but it didn't seem to make any difference.

Stun has always seemed to help with audio resolution problems in the past, so even though iptel advise against it I set one (stun.noc.ams-ix.net) in the field in the "network" tab of the PSIP set-up and got audio back on all the calls, so looks like I may be finally making progress. Don't want to speak too soon though, need to do more testing.

With the help of OscarTalks we have managed to get Psip working without having to register with a SIP provider such as Iptel. Psip actually works quite well in peer to peer mode so you don't really need to connect via a SIP provider. The main problem though is dynamic IP addresses which most people have.

OscarTalks has suggested a solution which works very well. The answer is no-ip.

Check out this web site: http://support.no-ip.com/customer/porta ... les/374286

No matter what you IP address is the no-ip site will be able to resolve your address.

smokey01 wrote:With the help of OscarTalks we have managed to get Psip working without having to register with a SIP provider such as Iptel. Psip actually works quite well in peer to peer mode so you don't really need to connect via a SIP provider. The main problem though is dynamic IP addresses which most people have.

OscarTalks has suggested a solution which works very well. The answer is no-ip.

Check out this web site: http://support.no-ip.com/customer/porta ... les/374286

No matter what you IP address is the no-ip site will be able to resolve your address.

+1

smokey01 wrote:With the help of OscarTalks we have managed to get Psip working without having to register with a SIP provider such as Iptel. Psip actually works quite well in peer to peer mode so you don't really need to connect via a SIP provider. The main problem though is dynamic IP addresses which most people have.

OscarTalks has suggested a solution which works very well. The answer is no-ip.

Check out this web site: http://support.no-ip.com/customer/porta ... les/374286

No matter what you IP address is the no-ip site will be able to resolve your address.

One of the oldest and most reliable of these types of providers is DynDNS.

I have been a user for more than 12 years without ANY issues. I have installed this solution in more than 100 accounts. Its free, friendly, always up, has a 5-star reputation and, most of all, instantly reliable.

PSIP Users can and do connect to you using the URL you pick when you sign-up to DynDNS. The ONLY thing required after you sign up is to insure that your router passes the port number for PSIP to your PC. Its that simple.

Hope this experience helps others.

gcmartin wrote:One of the oldest and most reliable of these types of providers is DynDNS.

I have been a user for more than 12 years without ANY issues.

Lucky for you, new users may not have the same luck:
http://tech.slashdot.org/story/11/12/17 ... ns-options

According to the linked page, the free service is being drastically cut back for new users (one free hostname, rather than five, and from a shorter list of branded domains), but not ended entirely. Existing users, it says, will see no changes "as long as you keep your hostnames active and up-to-date. If you allow your account or hostnames to expire, you will have to select from the new domains instead and will be limited to the one free hostname."

so best to have a couple options

Back in February, OscarTalks reported:

For dynamic IP address problems I have used no-ip.com but you need to have the "Dynamic Update Client" running to maintain the connection from your name domain to the dynamic IP and I don't know how to install that in Puppy.

Is there now a fix/workaround for this?

In fact there is a Linux version of the Dynamic Update Client available at the http://no-ip.com site which is very easy to install. I am still a relative newcomer to Puppy.

Incidentally if folks wish to test their PSIP Puppy Phone with a live human being rather than an echo test they can give me a call via the Iptel account when it is on-line. Sharing experiences and observations may help the further development of PSIP.

Unable to get my sound working on my new PC.
See details of mobo and sound here.
I'm looking for a PDF manual for the mobo.

There are 3 sockets on the back:
Pink=mic? | green=line out? | blue=line-in?
Mic connected to pink.
Amplified spkrs connected to green, with headphones connected to the headphone socket on the front of the powered spkrs.
Speakers given power, and switched on.

Missed the opportunity to test in 1st run of Psip-1.2.

OK...
Just now managed to use the "Media Test" files stored on my Flash Drive.
Played the "wheelie.mov" file, and the sound was OK.

So how do I test it and set it up in Psip-1.2?
Should be easy, right?!

Now managed to play music in Psip.
Tried the echo, but heard nothing, even with mic switched on and plugged in.
Retrovol has mic [and "Front mic" (its mobo header not connected to any socket)] boxes ticked and volumes up.
I have SVR installed, but heard nothing when I tried to record and playback [don't know what I'm doing].

Hello Sylvander

Sylvander wrote:Unable to get my sound working on my new PC.
See details of mobo and sound here.
I'm looking for a PDF manual for the mobo.

I wonder if this link is any good to you

CatDude
.

Thank you.

Got it.

Guys, a note, we will be needing the executable binary used in Psip compiled for ARM (armv6) architecture.

Barry,

Attached is Puppy Phone (PSIP) compiled in sap6
- pjproject 1.14 (I thought this was the latest, but then I checked it after compilation I saw they have released 1.14.2 ... anyway I'm not going to recompile unless people have problems with it - pjproject compilation took hours!)
- osxcart 1.1 (latest)
- psip proper from fossil trunk (latest - I haven't updated anything since last november)

This is a straight-compile, I haven't optimised anything (or enable optimisation flags etc).

jamesbond wrote:Barry,

Attached is Puppy Phone (PSIP) compiled in sap6
- pjproject 1.14 (I thought this was the latest, but then I checked it after compilation I saw they have released 1.14.2 ... anyway I'm not going to recompile unless people have problems with it - pjproject compilation took hours!)
- osxcart 1.1 (latest)
- psip proper from fossil trunk (latest - I haven't updated anything since last november)

This is a straight-compile, I haven't optimised anything (or enable optimisation flags etc).

Thanks for that! I'll put together a "puppy phone" psip pet and include it in the next sap6, for further testing.

I have been trying to get PSIP 1.3 up and running in Precise Puppy 5.4.1.1 on an Asus EEEPC. My main goal is to call landlines and cellphones.

My sip provider is diamondcard.us (www.diamondcard.us). I think all of my PSIP setup is correct. I can connect to landlines and my own cellphone but I invariably get an audio disconnect (and the call disappears from the active-call list) after about 10 seconds.

I have set my console log level to 6, but the activity log and dmesg do not seem to have any useful information.

I have been using Twinkle/diamondcard.us as a softphone/sip provider under Precise Puppy 5.4.1.1 with no problems, but I would prefer to use the Puppy softphone.

Anyone know what causes the disconnects and how to eliminate them?

A long time ago I had a problem of calls disconnecting after a fixed period of 30 seconds and disabling SIP ALG in my router cured it (if I recall correctly).

Doing that stopped the disconnects, but did introduce a problem of connection established but no audio (sometimes one way - sometimes both ways) which in turn I resolved by making sure to always set a STUN server.

I have since changed my ISP and router so can't reproduce the problem now. These SIP softphones can be a bit temperamental and there are lots of variables which is why I like to have 2 available. Puppy Phone usually works well and gives great audio. I have Linphone as my alternative.

Is Twinkle still being developed? I didn't think so. I recall trying it once in one of my Puppies and it didn't perform very well.

Thanks for your response, OscarTalks. I know zippo about how routers work, but I did manage to get inside mine and check all of its settings. However I could find no "SIP ALG" to enable/disable, and saw nothing about a STUN server. Just my inexperience as to where to look I guess.

Skype, Ekiga and Twinkle work with my router and its settings without disconnecting the call, which to my way of thinking, makes me doubt that router settings are the real issue when PSIP disconnects.

I want to use PSIP as a VOIP when travelling out-of-country, where international roaming charges on a cell phone would be prohibitive. Surely all of the wi-fi hotspots, airports, hotels and internet cafes in the world must adhere to the same protocols, or VOIP would be a nightmare!

I think you are right. Twinkle may no longer be actively developed. Maybe I should give Linphone a try.

LoboGrande wrote:I've been wanting to use MagicJack with Puppy. Anybody done this before?

Has there been any resolution to this question?