Puppy Phone - VOIP using SIP

Under development: PCMCIA, wireless, etc.
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smokey01
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#166 Post by smokey01 »

Sylvander wrote: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 :oops:

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. :oops:

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

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smokey01
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#167 Post by smokey01 »

pemasu wrote: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. :lol: 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

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Lobster
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#168 Post by 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 :)

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pemasu
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#169 Post by 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.

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smokey01
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#170 Post by smokey01 »

pemasu wrote: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.

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pemasu
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#171 Post by pemasu »

Smokey01. Thank you of the clarification. Then is probably something in my adsl box firewall, which brakes the connection. I am quite confident anyway that something in my firewall setup is the reason.
And now I remember that in build house rj45`s (4 of them) are connected first to my router and after that to the adsl box.
So...firewall setups in one or both boxes inhibits probably the conference calls. Now I would need to know what I should enable, lol.

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#172 Post by Sylvander »

1. "PM me your pjsua.cfg file."
Now sent.

2. "From left to right, Answer, Chat & Hangup."
They need those words incorporated in the buttons somehow.

3. "You also have to make sure you close Psip properly"
Now using "Phone->Quit and shutdown pjsua" as a matter of routine.
These 4 entries in the "Phone" menu seem handy to me.

4. "To stop the music press hangup. Remember it's a phone."
OK, I've got it.
Does the Help file explain that? Can't remember, but if it doesn't it aught to.

5. "The voice mail account is free"
Nice; must set one up.

6. "We are looking into video, lobster is doing that."
SUPERB!
This looks like it's going to be all I could hope for.

7. "I saw you online when I had you in my buddies as sylvander, no capital S."
Hmm, strange; I'm sure I typed my user display name with a capital S...
Always do.
Perhaps the server auto-eliminates all upper-case characters by changing them to lower-case.

Sylvander
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#173 Post by Sylvander »

@smokey01
1. Got your PM including the fixed version of my pjsua.cfg file I sent you just 2 hours before.

2.
(A) Deleted my copy and put your copy in it's place, then ran PSIP, and...
IT WORKS! :D 8)

(B) I have 6 distinguished Puppy buddies listed in the buddy-pane, plus me.
I'm the only one identified as online at this time [15:18 Europe/London time].

3. Tried "IPTEL->Music test call" [good music], heard on both my headset and speakers at the same time [I'm using a cable-splitter to feed both at the same time]. :D

4. Need some program or feature of PSIP for testing each of my 2 microphones [one on the headset, the other USB freestanding desktop mic]...
(a) Need it to allow me to choose which to enable and/or test. [Skype does this methinks]
And...
(b) Would be nice if it would display a REAL graph or meter that indicates the REAL signal stregth being detected coming from the chosen mic.
I've seen this when doing this job with my mic in Windows.

(c) Tried using "IPTEL->Echo test call", but heard only [very faint clicks at what may have been the start and finish, and] silence.

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#174 Post by Lobster »

6. "We are looking into video, lobster is doing that."
SUPERB!
This looks like it's going to be all I could hope for.
Grant has found that the alpha of Pjsua (our engine) contains video support 8)
Anyone feel like compiling and we can test on Slacko? 8)
http://trac.pjsip.org/repos/wiki/Video_Users_Guide

We may have to release two versions if modem audio support is compromised or the alpha build is unstable . . .

I tried to set up ipad2 to Puppy call no luck so far - next gonna try with Android again . . .

PSIP2 once set up seems OK
Grants help documentation is now updated and comprehensive
http://www.smokey01.com/help/psip/psip.html
Puppy Raspup 8.2Final 8)
Puppy Links Page http://www.smokey01.com/bruceb/puppy.html :D

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smokey01
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#175 Post by smokey01 »

Sylvander wrote:@smokey01
1. Got your PM including the fixed version of my pjsua.cfg file I sent you just 2 hours before. Excellent

2.
(A) Deleted my copy and put your copy in it's place, then ran PSIP, and...
IT WORKS! :D 8) Nice


(B) I have 6 distinguished Puppy buddies listed in the buddy-pane, plus me.
I'm the only one identified as online at this time [15:18 Europe/London time].Probably no one else was online


3. Tried "IPTEL->Music test call" [good music], heard on both my headset and speakers at the same time [I'm using a cable-splitter to feed both at the same time]. :D Excellent


4. Need some program or feature of PSIP for testing each of my 2 microphones [one on the headset, the other USB freestanding desktop mic]...
(a) Need it to allow me to choose which to enable and/or test. [Skype does this methinks]Have you tried the in built voice test ?

And...
(b) Would be nice if it would display a REAL graph or meter that indicates the REAL signal stregth being detected coming from the chosen mic.
I've seen this when doing this job with my mic in Windows.Use mhwaveedit, it has this feature and works well. You could try my little SVR that I posted in this forum


(c) Tried using "IPTEL->Echo test call", but heard only [very faint clicks at what may have been the start and finish, and] silence.
Recording and Playback can be a problem but it's not usually attributed to Psip, more likely your hardware. Sometimes it can be difficult to sort out. Persevere with mhwaveedit. If you can record and play back files with it then Psip should work.

At least you are still heading in the right direction, forward. :lol:

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#176 Post by smokey01 »

Lobster wrote:
6. "We are looking into video, lobster is doing that."
SUPERB!
This looks like it's going to be all I could hope for.
Grant has found that the alpha of Pjsua (our engine) contains video support 8)
Anyone feel like compiling and we can test on Slacko? 8)
http://trac.pjsip.org/repos/wiki/Video_Users_Guide

We may have to release two versions if modem audio support is compromised or the alpha build is unstable . . .

I tried to set up ipad2 to Puppy call no luck so far - next gonna try with Android again . . .

PSIP2 once set up seems OK
Grants help documentation is now updated and comprehensive
http://www.smokey01.com/help/psip/psip.html
lobster it may be better to continue this development as Psip rather than Psip2 as this may become confusing when and if we make traction with Pjsua2. I think Psip=pjsua and Psip2=Pjsua2 would better.

What do you think?

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#177 Post by Caneri »

Here is something I see...E
EDIT: This is Fatdog64 beta5
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#178 Post by pemasu »

I have got those also. I have launched psip many times from console just to see messages. But....if I try to kill leftover processes there arent any. If I use killall pjsua in console it hasnt ever found processes to kill.

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#179 Post by smokey01 »

pemasu wrote:I have got those also. I have launched psip many times from console just to see messages. But....if I try to kill leftover processes there arent any. If I use killall pjsua in console it hasnt ever found processes to kill.
pemasu, you might need to do it like this:

Code: Select all

pidof pjsua && killall pjsua || pjsua
You have to identify the PID first before you kill it.

This was going to be my next attempt at ensuring pjsua is closed properly.

Let me know if it works.

I'm off to work.

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#180 Post by Lobster »

lobster it may be better to continue this development as Psip rather than Psip2 as this may become confusing when and if we make traction with Pjsua2. I think Psip=pjsua and Psip2=Pjsua2 would better.

What do you think?
We have to know what we are talking about
and what end users see

For Final release Slacko menu
and eventually woof. Psip Puppy Phone

We do indeed use Psip to refer to the updates
and just call it PsIp even when running on a compiled for 64 bit version
and using the latest GUI

Psip2=Pjsua2, referring to a video enabled compile of the alpha Pjsua2 makes sense.

Alpha of Pjsua2 (for compiling please) here: :)
http://www.pjsip.org/download.htm
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#181 Post by Lobster »

This Sunday conference meet, Grant took us through all the updates.
Very good help file now available online
http://www.smokey01.com/help/psip/psip.html
Which chromium (on my system) is still incapable of displaying. Que?

Thanks to everyone who turned up
We need more users. Puppy phone certainly works.
We managed to boost or lower peoples mic settings.
and discuss Puppy Pi
http://elinux.org/RaspberryPiBoard
http://www.raspberrypi.org/?page_id=8

Next conference call
Sunday 18th September 2011 at 1300h GMT daylight saving time
Puppy Raspup 8.2Final 8)
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#182 Post by Sylvander »

Testing my 2 microphones:
1. USB standalone desktop mic by Logitech.
2. Swivel mic included on my headset = "Plantronics Gamecom 367".

A. Tested my USB mic in Win2000Pro using the sound recorder program, and it worked OK.

B. Rebooted into "Midnightsun-003-Tmix", and used the sound recorder included there.
It was hearing my headset mic, but rather faint, so in Retrovol I ticked the "mic boost" & "mic boost capture", which boosted the sound very noticeably.

C. Rebooted into my number-1 Puppy = Lupu -526.
Ran the Puppy_phone program...
Used the headset mic switch to switch on the mic, and could immediately hear myself in the headphones.
Clicked "IPTEL->Echo_test_call" and heard everything I said being echoed back through the headphones.
What a GREAT way to do the job!
So both headset mic and headphones are functioning well.
All ready to go methinks.

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#183 Post by dogle »

We managed to boost or lower peoples mic settings.
- this was enlightening. Smokey01 and I had had a chat shortly before the conference, and tweaked our mic settings then. At the start I could barely hear Lobster, and Caneri was pretty faint.

It seems that the path through the SIP process attenuates the sound level quite a bit, and setting the mic using the Test Sound menu does not reflect this. I wonder, would it be possible for Test Sound to 'simulate' this attenuation, thus inducing the user to set the mic a tad higher?

I find that I need a mic setting close to max +20dB boost. This prevents me from using a handset, which sets up acoustic feedback howl across itself.

Thinking that there must be some spurious electrical crosstalk in the system to cause this, I spent some fruitless hours chasing my tail trying to nail it ..... then the elephant in the room broke wind, kaboom! - ALSA is simply set up to feed mic input to speaker by default! Quite reasonable huh, except when you don't want this to happen. Not good for VoIP, you want to hear the other guy's voice, not your own (loud).

Unfortunately none of the various mixer GUIs seem to offer any way to overcome this. Trying to figure out a way from the ALSA documentation I just got savaged ..... my head is still spinning.

Any ALSA gurus out there?

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#184 Post by Lobster »

Think you mentioned you had this Grant.
Just in case still relevant . . . the enclosed quickstart guide goes in the resources directory and is for off-line usage.

Can we make use of it? 8)
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#185 Post by mcewanw »

dogle wrote:ALSA is simply set up to feed mic input to speaker by default! Quite reasonable huh, except when you don't want this to happen. Not good for VoIP, you want to hear the other guy's voice, not your own (loud).
Not sure if this is what you are after but if you start the program alsamixer you don't want any mike gain on the Playback settings (only on the Record or Capture settings). That is, tab to the alsamixer Playback control settings, move the cursor to the Mic setting control, once there make sure the gain for Mic playback is set at an absolute minimum and also press key M to toggle the Mic control such that is not selected at all in Playback mode.
github mcewanw

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