Puppy Phone - VOIP using SIP

[Sylvander]

1. I uninstalled ALL of the psip already existing in my Lupu-526, and installed the new version using psip-0.26.pet.

2. I then noticed the [empty] folder /.psip right at the top of the folder hierarchy.
I don't like it at the top; I want my /00 folder there.
Anything I [or you] can do about that?

3. When I run the command psip in a terminal, it reports:
# psip
# /usr/sbin/psip: line 3: ./psip: No such file or directory

4. /usr/local/psip is empty also.

5. /usr/sbin/psip is 42 bytes.

[smokey01]

1. I uninstalled ALL of the psip already existing in my Lupu-526, and installed the new version using psip-0.26.pet.

2. I notice the [empty] folder /.psip right at the top of the folder hierarchy.
I don't like it at the top; I want my /00 folder there.
Anything I [or you] can do about that?

3. When I run the command psip in a terminal, it reports:
# psip
# /usr/sbin/psip: line 3: ./psip: No such file or directory

4. /usr/local/psip is empty also.

You could rename 00 to .00

[Sylvander]

1.
(a) OK, renamed /00, /01, /02, /03, /04 to /.00, /.01, /.02, /.03, /.04

(b) And now I cannot set my "Firefox->Edit->Preferences->General->Downloads->Save files to..."
I'm not offered the option to save to /.00...
Can only save to unhidden files...
Therefore chose /00, and that folder was auto-made when I chose to re-download psip-0.26.pet to see if it was empty or not.

2. It was somewhat strange as follows:
Folders that had files in them...
When I did the rename....
Xfe crashed and had to be restarted.
Once open there were 2 copies of the folder:
One with the original name; one with the new name.
The new folder had only one of the files in it, and it was of zero size.
So I deleted the zero-size file, copied ALL the files over from old to new, and deleted the old folder including all its copies of the files.

3. psip still doesn't run...
Still gives the same error.

4. I notice the error mentions [folder or file?] ./psip, yet I have no hidden ./
What I have is unhidden /

[smokey01]

1. OK, renamed /00, /01, /02, /03, /04 to /.00, /.01, /.02, /.03, /.04

2. It was somewhat strange as follows:
Folders that had files in them...
When I did the rename....
Xfe crashed and had to be restarted.
Once open there were 2 copies of the folder:
One with the original name; one with the new name.
The new folder had only one of the files in it, and it was of zero size.
So I deleted the zero-size file, copied ALL the files over from old to new, and deleted the old folder including all its copies of the files.

3. psip still doesn't run...
Still gives the same error.

4. I notice the error mentions [folder or file?] ./psip, yet I have no hidden ./
What I have is unhidden /

I haven't tried it on Xfe.

I have tested on Lupu-525
midnightsunpup
Slacko-499.4

All works well on each distro

If you run it from the CLI it should work from /usr/sbin/psip or /usr/sbin/local/psip

Have you tried it running live from a CD?

[Sylvander]

1. I decided to reboot without saving the session, so that the installation would be gone.
Got back into Lupu-526, and "Menu->Internet->Psip VOIP + IM client" was in the menu system.
Clicked on that, and PSIP 0.26 appeared.
The I guess I must have saved the session right after I installed.
Is it necessary to reboot after installing this?

2. The changes I made to my folders had not been saved.
I think I'll leave them as-is since it's going to be difficult to get Firefox to save to any hidden folder.

[smokey01]

1. I decided to reboot without saving the session, so that the installation would be gone.
Got back into Lupu-526, and "Menu->Internet->Psip VOIP + IM client" was in the menu system. It should have changed to Psip Puppy Phone
Clicked on that, and PSIP 0.26 appeared.
The I guess I must have saved the session right after I installed.
Is it necessary to reboot after installing this? Not normally

2. The changes I made to my folders had not been saved.
I think I'll leave them as-is since it's going to be difficult to get Firefox to save to any hidden folder.

Try logging into the conference tomorrow at 1300GMT DST. We will be using channel 528, under heading IPTel.

[dogle]

I've just discovered that an incomplete buddy line in pjsua.cfg, thus

Code: Select all

--add-buddy

is sufficient to prevent pjsua logging in properly.

I was lucky, it only took me an hour or so to figure out what-the-heck because I happened to have working and non-working PSIPs running on different boxes. The trouble is that it is completely baffling to the newcomer - "me PSIP just don't work!" - yet all too easy to induce by over-hasty handling of Add Buddy menu item. Right now I'm a bit short of bright ideas for newbie-proofing this one.

[smokey01]

I've just discovered that an incomplete buddy line in pjsua.cfg, thus

Code: Select all

--add-buddy

is sufficient to prevent pjsua logging in properly.

I was lucky, it only took me an hour or so to figure out what-the-heck because I happened to have working and non-working PSIPs running on different boxes. The trouble is that it is completely baffling to the newcomer - "me PSIP just don't work!" - yet all too easy to induce by over-hasty handling of Add Buddy menu item. Right now I'm a bit short of bright ideas for newbie-proofing this one.

dogle, that's a very old problem that has been discussed many times over the years. There is no error checking of the config file. I haven't been able to find a solution and neither has anyone else that I'm aware of.

I would be very happy if someone could produce some error checking code to solve this issue. Another gotcha is the additional blank line required on the last line of the config file. Without the extra line it sometimes causes problems however, not always.

Generally it won't log in to the server. This is the reason I keep saying put your own address in the buddies list. If you see your name on the status bar online then there is a good chance things are working properly.

It's always the first place I look when things aren't working.

[dogle]

Thanks Smokey - hey, don't you ever sleep?

Although I shall attend tomorrow's conference, sorry folks if I am unable to participate ... reason is, at present my incoming sound is at times so badly square-waved that it takes my feeble brain a second or two to decrypt what is being said, and I can't keep up with the discussion! ... I'm working on it (the sound problem, I mean).

Sorry also to those whom I may have bothered today with duff calls - seems I have a gremlin in the mic channel on my test box.

In trying to troubleshoot the latter, I switched my headset for a handset and I was astonished to find that I could very easily induce acoustic feedback (howl) by e.g. cupping my hand across the handset .... wait for it ... with PSIP disconnected!

This should not be possible, without an electrical link to complete the feedback loop, so it has me puzzled and fascinated to know why. It is happening on different machines with Puppy 4.3.1/AlsaMixer and Wary 5.1.2/Retrovol, so maybe there are wider implications re. the all-important sound setup in PSIP.

AlsaMixer and Retrovol are (I think) cognate to some extent, is there something in their architecture which can permit spurious transfer across 'audio' output and mic input channels ... perhaps at the hands of a nincompoop like me? Any ideas here welcome, if not I'll raise it in another thread.

So far, PSIP 0.26 is looking just grand.

Conference: just to clarify, that's 1300GMT = 1300Z = 1300UST = 1400 BST (British Summer Time = daylight saving time), have I got that right?

[smokey01]

Thanks Smokey - hey, don't you ever sleep?

Not enough ZZZzzzzz.....

Sorry also to those whom I may have bothered today with duff calls - seems I have a gremlin in the mic channel on my test box.

This is why the voicemail facility is so useful.

In trying to troubleshoot the latter, I switched my headset for a handset and I was astonished to find that I could very easily induce acoustic feedback (howl) by e.g. cupping my hand across the handset .... wait for it ... with PSIP disconnected!

That's a new one

So far, PSIP 0.26 is looking just grand. Thanks

Conference: just to clarify, that's 1300GMT = 1300Z = 1300UST = 1400 BST (British Summer Time = daylight saving time), have I got that right?

Same as last Sunday. 1300h BST.

[smokey01]

New help file that now reflects 0.26 with a hyper linked table of contents and everything.

It does need some formatting which gcmartin will address.

dogle, very happy for you to do some proof reading now as I'm sure there are lots of mistakes.

lobster, now I am happier to move forward to an official release date.

caneri, more coffee please.

Fred, good to see you back.

Chat later, 130h GMT BST

Channel 528.

[Sylvander]

smokey01 sent me a PM asking if I'd got this working.
Thought it best to reply here, so he and all can see my reply.

1. I have been having some snags when attempting to set it up and using.

Deleted my lupusave, and restored a backup copy, so as to begin afresh.

Latest install attempt went well.
Deleted ALL I could find of psip.
More than advised.
Re-installed using the latest PET file.
Good result!
Program entry in the Puppy Menu reads "PSIP Puppy Phone".
Program runs [after a reboot].
Looks good.
Added a couple of "Buddies".
[sip:smokey01@iptel.org, sip:lobster@iptel.org, sip:caneri@iptel.org ]

2. PROBLEM:
(a) After I close down and re-run the program, no buddies show.
Clicking the "Refresh Buddies" button doesn't make them appear.
Used "Configure->Edit config file", and added the buddies in there, then...
Shutdown Pjsua, then Start Pjsua, but it doesn't help.
Still no buddies displayed after I click "Refresh Buddies".

(b) Don't know how to setup mic and speakers [test them].
I have a headset with built-in mic [both connections have 3.2mm jacks], and also a USB desk mic [has it's own sound card circuitry built in].
In Skype can specify that this [or that] mic is to be used, but not in PSIP methinks.

3. Found my way to the new online help manual ["PSIP->Help->Help"].
Looks good.
Will begin reading that.

[smokey01]

make sure you add your own address to buddies.

If you don't see yourself online then you have done something wrong although our chat worked well.

Play about with mhwaveedit or that little SVR I put in the thread.

When you can record and playback sound, Psip will probably work.

What Puppy are you using?

[Sylvander]

1. Running Lupu-526.

2. Just now added my own address = sip:Sylvander@iptel.org [Note the capital S]
... In PSIP.
Then closed PSIP, and opened the /root/.psip/pjsua.cfg in leafpad, and there was my address added to the bottom of the list.
Yet none of the buddy names are displayed in the PSIP pane.

3. Regarding things read in the help file:
(a) Is a voice-mail account free, or does it cost?
(b) Pity about there being no plans for video capability.
(c) When your voice call came in, and there was the little box with the 3 buttons, I didn't know which did what, and hovering the cursor over each produced no text identification.
(d) The 4 commands involving pjsua seem good to me; I'm inclined to want to see them kept in the prog.
(e) I noticed in my previous install attempt...
That when I used "IPTEL->Music test call", the music played, but there was no way I could see to end the music.
(f) Very recently I noticed the /root/.psip/pjsua.cfg file included a setting = --add-buddy [with no address coming after] but I removed that just in case it wasn't good to have.
The help files have just now told me it's bad.

4. Just now noticed that my %CPU usage is VERY high [90%?]
6 horizontal lines in the usage display at bottom right.
Will try "Ctrl+Alt+backspace" then xwin, and get back.
---
That restored my normally low %CPU usage [5%?]
Dropping toward a single line in the usage display->now below 1 line.

[pemasu]

I have installed and played with Psippy a little. Probably I dont have 100 % sane install, but...I have now registered. I suppose it does not matter what phone number you give while registering to the iptel.org. I think that information could be updated to the help-manual or whatever will be the source for newbies like me.

pemasu@iptel.org

I am not sure if my conference room call actually proceeds, I get message: "call has been disconnected: unexpected condition". When I start psip in console, I get message about dialog_disconnectedcall script started and then finished, the script was not included in psip-0.26.pet. I found it from some older pet or compressed package. Okay, there is XX-dialog_disconnectedcall, so I made symlink to that script asking dialog_disconnectedcall, but error message stays.... So...I am more experimenting than understanding. Lol.

[smokey01]

1. Running Lupu-526.

2. Just now added my own address = sip:Sylvander@iptel.org [Note the capital S]
... In PSIP.
Then closed PSIP, and opened the /root/.psip/pjsua.cfg in leafpad, and there was my address added to the bottom of the list.
Yet none of the buddy names are displayed in the PSIP pane.

3. Regarding things read in the help file:
(a) Is a voice-mail account free, or does it cost?
(b) Pity about there being no plans for video capability.
(c) When your voice call came in, and there was the little box with the 3 buttons, I didn't know which did what, and hovering the cursor over each produced no text identification.
(d) The 4 commands involving pjsua seem good to me; I'm inclined to want to see them kept in the prog.
(e) I noticed in my previous install attempt...
That when I used "IPTEL->Music test call", the music played, but there was no way I could see to end the music.
(f) Very recently I noticed the /root/.psip/pjsua.cfg file included a setting = --add-buddy [with no address coming after] but I removed that just in case it wasn't good to have.
The help files have just now told me it's bad.

4. Just now noticed that my %CPU usage is VERY high [90%?]
6 horizontal lines in the usage display at bottom right.
Will try "Ctrl+Alt+backspace" then xwin, and get back.
---
That restored my normally low %CPU usage [5%?]
Dropping toward a single line in the usage display->now below 1 line.

PM me your pjsua.cfg file.

The three little icons are there so when you have Psip minimised you can still answer or hangup. They are the same icons as the main buttons.

From left to right, Answer, Chat & Hangup. Might have to add this to the help documents, forgot

You also have to make sure you close Psip properly otherwise you might have more than one instance running and this can cause problems. It's in all the help documents.

To stop the music press hangup. Remember it's a phone. Whenever you finished checking your voice, talking, listen to music etc, hang up. That's hat you would do with your mobile or home phone.

The voice mail account is free, as far as I know.

We are looking into video, lobster is doing that. What he might not know is version 2 of pjsua has video included. Still in Alpha.

I saw you online when I had you in my buddies as sylvander, no capital S.

Cheers

[smokey01]

I have installed and played with Psippy a little. Probably I dont have 100 % sane install, but...I have now registered. I suppose it does not matter what phone number you give while registering to the iptel.org. I think that information could be updated to the help-manual or whatever will be the source for newbies like me.

pemasu@iptel.org correct

I am not sure if my conference room call actually proceeds, I get message: "call has been disconnected: unexpected condition". When I start psip in console, I get message about dialog_disconnectedcall script started and then finished, the script was not included in psip-0.26.pet. I found it from some older pet or compressed package. Okay, there is XX-dialog_disconnectedcall, so I made symlink to that script asking dialog_disconnectedcall, but error message stays.... So...I am more experimenting than understanding. Lol.

You are such a fiddler, just like me. I left that XX-dialog_disconnectedcall file in. I should have deleted it as it's a bit of a pain having to keep on closing all of the dialog boxes each time you disconnect. When you remove the XX from the front of the file it's works as planned. Better XX the file out, more peaceful.

It sounds like you may not be logged in properly. When you click on menu item IPTel and click on Conference Call room 528, you should hear a voice saying your are the first in the conference. If you are not the first you will hear a beep as someone else is already in the room. When someone leaves you will also hear a beep.

The help file will provide some valuable answers.

Hope to chat later.

Cheers

[Lobster]

Me, Eric, and Grant had an impromptu conference call yesterday. Until I was called away for emergency shopping . . .
Another one today
Sunday 11th September 2011 at 1300h GMT daylight saving time

I was hoping to create a wizard for ffmpeg webcam synch
but have not worked it out yet (not essential but fun)
Any help from smarter pups welcome
Smokey (Grant) is interested in the new serverless SIP feature of PJSUA
- any help with that welcome . . .

There is a quick start guide here:
http://puppylinux.org/wikka/Psippy

and also an experimental wizard
http://www.murga-linux.com/puppy/viewto ... 148#559148

Hopefully Grant will soon be incorporating the sound recorder into Puppy phone (Psip2) - allowing you to add recordings to emails

Beta 3 of Slacko might well contain the latest Puppy Phone

Puppy
Growling, howling, barking

[pemasu]

To clarify my situation. I have made now succesful call. I should have now right pjsua.cfg. Conference call still results in unexpected condition error. I checked all the logs in /tmp/psip. I got internal server, 500, error. Then I thought about my net connection arrangement. I have wireless connection to my separate wlan router which is connected by rj45 to the adsl box which is connected to the phone line. Only phone line we have in the house, lol. Finnish way. Old phone lines to make calls are coming obsolete in Finland. They are needed for adsl still.

Anyway, I changed to use our house in build rj45 network and I disabled the wlan card from my laptop. Just press Fn + Wlan button. Frisbee handled with xbindkeys binding the disabling. Nice feature.
With wired house inbuild rj45 connection straight to the adsl modem the conference call didnt give that error message. I just got message about timeout. Maybe it is due to there wasnt anybody in line at conference room.

Anyway, progress. I am testing now in MNS-003 Office edition, but so far calling seems to work in Dpup Exprimo 5.X.6, Polarpup and MNS.
Buddies list refreshes, I just have Question symbol as first symbol. It misled me thinking it does not poll the server. But when I empty /tmp/psip/buddy-list it writes the polled users ok to the file. So it polls and writes on line users to the buddy-list.

[smokey01]

Maybe it is due to there wasn't anybody in line at conference room.

No, if you are the first in the conference, a recorded voice message will say so.