Puppy Phone - VOIP using SIP

Under development: PCMCIA, wireless, etc.
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russoodle
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#501 Post by russoodle »

Hi nooby, if i'm stll awake, i should be there this time :)
[i][color=Green][size=92]The mud-elephant, wading thru the sea, leaves no tracks..[/size][/color][/i]

DPUP5520
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Joined: Wed 16 Feb 2011, 05:38

#502 Post by DPUP5520 »

Same here
[url=http://www.murga-linux.com/puppy/viewtopic.php?t=69651][b][i]PupRescue 2.5[/i][/b][/url]
[url=http://www.murga-linux.com/puppy/viewtopic.php?t=72178][b][i]Puppy Crypt 528[/i][/b][/url]

nooby
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#503 Post by nooby »

My local time is now 9.43 something in the morning and
I am still in my dreams but them slowly fade away. 3 hours to go
I use Google Search on Puppy Forum
not an ideal solution though

Sylvander
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Joined: Mon 15 Dec 2008, 11:06
Location: West Lothian, Scotland, UK

#504 Post by Sylvander »

1. Working in...
Dpup-squeeze-5.X.9-SCSI
And...
PSIP 1.2-pre ef5ea635e4

2. When I try to set "Preferences" on my 2003 hardware...
The window freezes Psip.

3. Cannot see any way to reduce the sampling rate.
There's no setting opportunity available in...
Setup->Miscellaneous".
Or anywhere else that I can see.

nooby
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#505 Post by nooby »

Sylvander, I have time to talk but I fail to make use of the
new 1.2 I can only use the very old one from 2008
so if you need to compare what to write in the config
then I can not help but I can help with how it sounds
but so can Dpup and he most likely is online.
I use Google Search on Puppy Forum
not an ideal solution though

dogle
Posts: 409
Joined: Thu 11 Oct 2007, 12:41

#506 Post by dogle »

jamesbond wrote: 1. "Freeze problem" when closing the setup dialog - this is hardware problem. Computer is not powerful enough or
soundcard is not powerful enough to run psip. Fix - quality settings must be reduced from default 5 or 4 or 3 or 2...
No, that's mixing up two different probs - lowering the quality setting improved my voice quality in Psip0.26, but the problem of Psip32 refusing to play with old hardware, and freezing up without making the .psip.config file, remains.

We we discussing this in the conference today, and DPUP5520 very kindly sent me a ready-made .psip.config template just in case I could fool Psip32 into thinking it didn't have a hardware problem after all .... no joy, still freezes as usual (no great surprise really, because Psip32 is seen to grumble about the hardware right at the outset when started from the CLI).

Now here's a puzzle ... having a non-high-quality internet connection, I have frequently needed several attempts to get voice on the automated IPTel utilities. This seemed to get worse after I had reduced the (0.26) call quality setting; last time it was very hard to get anything except the conference room, which worked with no problem at all.

Today I had the exact opposite: all the IPTel voice utilities worked first time every time .... except the conference room! I resorted to text thinking conference must be u/s today after many attempts, and I'm still not sure how I finally managed to get conference voice.

Now, I'm using a dedicated box for Psip and I haven't touched anything since the previous time, so I'm confident that all my settings are unchanged .... which is telling me that something has changed at the IPTel end of things. But what? Has anyone an insight as to why the relative voice accessibilty on conference v. the rest of the utils should suddenly flip over?

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russoodle
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#507 Post by russoodle »

Have to agree with Dogle regarding the freeze issue not necessarily being caused by hardware not being powerful enough, because i experienced the very same frustrating issue a couple of weeks back on the same hardware i'm using now, ie. one Puppy or another running on an Intel Mac C2D 2.4GHz machine, 3GB RAM and HDA Intel audio adapter. While it's not exactly a you-beaut top-of-the-line system, it's not a slug, either.

Last few days i've been using the integrated Psip 1.2-RC2 64bit on Lighthouse Pup 64, prior to that was Fatdog 521 64 with Psip 1.2 and i haven't had the freeze issue for some time now.

I'm not really fussed on the Alsa Mixer though, so i use Retrovol..

It can be quite confusing, trying to sort out these issues, as there've been so many iterations of Psip over the past couple of months, + 32bit and 64bit versions + individuals' different Puppys and wide-ranging hardware specs as well...

If anyone here can make head or tail of it all, you're definitely a better man than i! :wink:
[i][color=Green][size=92]The mud-elephant, wading thru the sea, leaves no tracks..[/size][/color][/i]

nooby
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#508 Post by nooby »

I gave the modern 32 a try which is psip 1.2 as I get it.

I either have done something very wrong or it is same issue that
dpup5520 had with the router.

The very old psip 0.12 just works every time and I spent an hour
on the new one three times so I give up on it now.
I will continue using the oldest because it just works.
I use Google Search on Puppy Forum
not an ideal solution though

nooby
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Location: SwedenEurope

#509 Post by nooby »

I have the old Psip running and it works rather well.
I have spent hour on two or three day and totally
fail to get teh modern Puppy phone going.

Now a neighbor asked me if I had Skype.

No but I have Puppy Phone I told her.

Can I call her too over the puppy phone? Would that work?


sure she has to tell me her sip number. Does skype have a
general sip or do them only talk to each other and landlines?
I use Google Search on Puppy Forum
not an ideal solution though

DPUP5520
Posts: 800
Joined: Wed 16 Feb 2011, 05:38

#510 Post by DPUP5520 »

They use two different protocols so no you can't call skype from Psip

Edit: I may have been wrong check these out

http://scopezoom.com/guide9.htm

http://gigaom.com/2009/02/09/gizmo5-lau ... -ip-phone/
[url=http://www.murga-linux.com/puppy/viewtopic.php?t=69651][b][i]PupRescue 2.5[/i][/b][/url]
[url=http://www.murga-linux.com/puppy/viewtopic.php?t=72178][b][i]Puppy Crypt 528[/i][/b][/url]

nooby
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Location: SwedenEurope

#511 Post by nooby »

Thanks for one of them are totally dependent on one man
giving us that for free using his server.
Keep in mind that the middleman making this possible Net2Max.com can only provide so many channels, so "Please don't abuse it, or We'll all lose it"
And Gizmo write this
What we’ve done is create a SIP alias for every Skype user. So if you want to call a Skype user named echo123 you simply dial echo123@opensky.gizmo5.com from any SIP aware device (which is just about every piece of VOIP equipment). Users can even have any SIP call forwarded to their Skype address using my.gizmo5.com.

All calls up to five minutes are free, while longer calls are going to cost you money.
which shows it only works fhorugh a middle man.

Unless some clever person look into the server that Opera use
and if one can make such a thing on every Opera browser.

I doubt that is possible or one would have heard of it by now?
What I could do is talk to her and her Fiance or what name to use.
Her BoyFriend and them both get puppy to boot in frugally or through
VBox or something and that way them can talk to each other for free.
I use Google Search on Puppy Forum
not an ideal solution though

dogle
Posts: 409
Joined: Thu 11 Oct 2007, 12:41

#512 Post by dogle »

Thanks nooby for confirming that the Old Original PSip is still fit for purpose - so that should maintain retro-capability to the 4-series Puppys, which is lacking in this year's versions? - I must check that out, I fear that voice quality issues may persist with the 4-series sound setup, but nooby sounded pretty good in the last conference; I surmise however that he was using a later Puppyversion.

Thanks DPUP5520 for the links re. Skype access .... I have no interest in M$Skype, but was intrigued by those net2max.com offerings re. PSTN/VoIP connections .... so much so that I've spent an hour or two trawling through their blurb. (I hope that they are good guys and deserve my good wishes for success in their innovative enterprises, but their screed scares me and ensures that there's no way they get my bank card details).

I am dismayed by the apparent general lack of enthusiasm here for PSip, which the regular contributors to this thread know to work remarkably well once sound/Puppyversion/hardware/duff connection hurdles have been surmounted.

Perhaps The Great Leap Forward for Psip will be the availabilty of cheap calls to/from those without a SIP code, i.e. the PSTN? So far nobody has reported success with this. If you have found a service provider which connects PSip - public networks calls reliably and economically, please, do tell!

Hogweed
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Joined: Sat 12 Feb 2011, 19:37

#513 Post by Hogweed »

Ok Just installed psip32 1.3 on Lucid Puppy 5.2-update1 (yes I'll get round to upgrading to latest 5.2.8 some time). Gave it my sipgate details and it seems to work with a couple of problems.

1) If I enter a proxy sipgate.co.uk as sipgate recommend then the login never completes and hangs for ever. EDIT:Found the problem - I needed a "sip:" before sipgate.co.uk. Why do I need the "sip:" field in front of URL, registrar and proxy anyway? Don't need this for other clients I've tried. If I forget it it should not hang for ever with no error in any case.

2) Incoming PSTN callers to my sipgate Direct Dial In phone number do not hear a ring tone. They just hear silence after dialing until I answer then they hear my voice.

Are these known issues? Any suggestions?

Just to confirm I can successfully make and receive calls to/from the public telephone network via psip32 and sipgate.

Hogweed
Posts: 96
Joined: Sat 12 Feb 2011, 19:37

#514 Post by Hogweed »

gcmartin wrote:
Has any member[/b] of this thread-forum used PuppyPhone to "connect" to a "non-PC client" SIP device? (There are plenty on SIP devices that don't need PCs for calling)
Yes. I am using it and can call real phones (no client needed) and to receive calls. Works ok except user calling from a real phone does not hear ringtone. (they do if I use other clients than psip). Perhaps the psip client is missing sending a signal to say it is ringing? See my previous message in thread. EDIT: Actually it appears incoming callers are not getting charged by their Telephone Co. either even though the call connects!! Probably a timeout limit on that I guess but it looks like psip may be missing some signalling.

sipgate is pay as you go and it is totally free if you just use it for incoming calls - they give you a number for free. Any of tbe developers tried it with psip?

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OscarTalks
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#515 Post by OscarTalks »

Hello Folks.

Recently started using Puppy and really like it.

Would like to get the SIP phone working reliably and properly. Using Lucid 5.2.8.004 with PSIP 0.12 included. Are there newer versions available?

I read through the 16 page thread linked from the client and then discovered this one with 35 pages. Will take a while to read it all.

I can get PSIP to work sometimes but often it seems to stop working. Difficult to figure out what is wrong when the "fault" is intermittent like that.

I have two sound cards and want to be able to route the telephony audio to my second card (as I do with Skype). I like to separate it from other audio sources running on the computer. There is something about device ID's in the pjsua info but I wasn't clear about what it meant. Where do I find the soundcard device ID's?

Happy to do test calls or conferences. Using sip:oscartalks@iptel.org as my main URI for now.

The reg timeout default is 55 but in other clients it often is 3600. I take it this is not a problem. Also tried putting "iptel.org" into the realm field but don't think it makes a difference from wildcard.

If PSIP is still in development, can I install other sip clients in Puppy and how do I do that?

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Lobster
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#516 Post by Lobster »

OscarTalks wrote: Would like to get the SIP phone working reliably and properly. Using Lucid 5.2.8.004 with PSIP 0.12 included. Are there newer versions available?
click on red phone
http://www.smokey01.com/menu/ :)
Puppy Raspup 8.2Final 8)
Puppy Links Page http://www.smokey01.com/bruceb/puppy.html :D

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russoodle
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#517 Post by russoodle »

@OscarTalks: Welcome to the kennels! :D

You might by now have followed Lobster's link and downloaded/installed the latest version. If not, i recommend you do so and, if you log on to Psip tonight, (Oz time, don't know where you are), there are usually a couple of us around and perhaps we can help you get sorted if you still need help.

Best way to start with it is to open Psip, add your own contact details, open the "Setup" menu at top left, enter your connection details in the "Account" tab ( BTW, just use the asterisk wildcard for realm).....at this point, before worrying about "Audio", "Network" or "Misc" tabs, quit Psip so it will create a configuration file. When you relaunch it, see if the Audio settings are correct and make any other necessary adjustments.

The current versions of Psip only use one .conf file, so there are no pjsua thingies to play with..

To be able to connect with us, you'll need to add us to your contact list, so you can see if we're online, otherwise we don't show up, AFAIK ....you'll find me at sip:russoodle@iptel.org. Smokey01 is one of Psip's developers, so he's much more knowledgeable about it and usually checks in at some stage. CatDude is also very clued up and they're both with iptel.org as well (using all lowercase usernames).

HTH
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OscarTalks
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#518 Post by OscarTalks »

I am logged in to iptel and can make some calls but the iptel utilities including conference don't seem to be working for me. With conference I get a split second of audio, sounds like the start of a welcome greeting but then all goes quiet. Echo and music test give nothing. I got some IM's from you (saying you were in conference) and tried to reply but don't think you received it.

The first call from you worked for a bit but I answered it with the config window open and then when the saving ram message came up the call dropped.

Think the audio devices selector is working OK.

I can call the Wideband Audio Demo (demonstrates the G722 codec)
sip:wbdemo@conf.zipdx.com

I can call Mouselike test facility
sip:904@mouselike.org

Calling toll free US numbers is working eg voice-interactive "Tell Me"
sip:18005558355@tf.callwithus.com

I am in England and there is a provider of free UK numbers which they route to a sip softphone. The caller pays normal national rates to call from a landline (mobiles may cost more). Useful if you want to advertise a number publicly and answer the calls with PSIP. (uknumber.co.uk)

Still not sure if I should use ice. Iptel say that use of stun is not recommended but with some other clients I found I needed it.

Will do more testing and see what's what.

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OscarTalks
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#519 Post by OscarTalks »

Connecting to music@iptel if I tick and untick the "hold" I get a very brief burst of audio. Looking at the "stats" button I see it is trying to use speex. I wonder if it is a problem with this codec as mentioned in the previous version conf file.

How do I disable speex codec in this version? The conf file for this is located in the ~ directory and not in /root/.psip is this correct?

Regarding peer-to-peer, I have used audio servers quite a bit in this mode, Icecast, Shoutcast, that sort of thing. Anyone can connect to my PC directly and receive whatever audio I am sending. Usually the client needs to be listening for connections and if you are behind a router you need to set the NAT port-forwarding manually.

With PSIP I am not sure that just having the client running would maintain ports open and a path through NAT to listen for any other client attempting to make the connection and call you.

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.

I would be up for some testing though at some point. I have a single port modem/router which gives me an internet IP address (which I could give you at the time) rather than a LAN one. I think it will work, the problem would be how to make it user-friendly for the general public.

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smokey01
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#520 Post by smokey01 »

OscarTalks what sort of computer do you have?

Is it older than about 7 years? If so that may be your problem as some others have difficult running the later Psip on older computers. There are a few people that use the 0.26 version which works quite well on older computers.

I have just compiled 1.3 for Saluki which seems to be quite stable.

What OS are you using?

Have you read the help file?

Psip is very lightweight with exceptional sound quality. As russoodle said, a few of us get together most mornings (UK time) for a bit of a chat.

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