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 Forum index » Advanced Topics » Cutting edge
Puppy Phone - VOIP using SIP
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nooby

Joined: 29 Jun 2008
Posts: 10557
Location: SwedenEurope

PostPosted: Thu 17 Nov 2011, 13:08    Post_subject:  

Thanks for one of them are totally dependent on one man
giving us that for free using his server.

Quote:
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
Quote:
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.

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dogle

Joined: 11 Oct 2007
Posts: 332

PostPosted: Thu 17 Nov 2011, 17:48    Post_subject:  

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!
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Hogweed

Joined: 12 Feb 2011
Posts: 96

PostPosted: Tue 31 Jan 2012, 12:46    Post_subject:  

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.
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Hogweed

Joined: 12 Feb 2011
Posts: 96

PostPosted: Tue 31 Jan 2012, 20:09    Post_subject:  

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

Joined: 05 Feb 2012
Posts: 901
Location: London, England

PostPosted: Sun 05 Feb 2012, 21:39    Post_subject:  

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
Official Crustacean


Joined: 04 May 2005
Posts: 15117
Location: Paradox Realm

PostPosted: Sun 05 Feb 2012, 22:56    Post_subject:  

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/ Smile

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russoodle


Joined: 12 Sep 2008
Posts: 665
Location: Down-Under in South Oz

PostPosted: Mon 06 Feb 2012, 02:45    Post_subject:  

@OscarTalks: Welcome to the kennels! Very Happy

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

Joined: 05 Feb 2012
Posts: 901
Location: London, England

PostPosted: Mon 06 Feb 2012, 08:44    Post_subject:  

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

Joined: 05 Feb 2012
Posts: 901
Location: London, England

PostPosted: Mon 06 Feb 2012, 12:05    Post_subject:  

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


Joined: 30 Dec 2006
Posts: 1829
Location: South Australia

PostPosted: Mon 06 Feb 2012, 15:50    Post_subject:  

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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OscarTalks

Joined: 05 Feb 2012
Posts: 901
Location: London, England

PostPosted: Tue 07 Feb 2012, 09:10    Post_subject:  

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.
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smokey01


Joined: 30 Dec 2006
Posts: 1829
Location: South Australia

PostPosted: Tue 07 Feb 2012, 15:47    Post_subject:  

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.

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smokey01


Joined: 30 Dec 2006
Posts: 1829
Location: South Australia

PostPosted: Tue 07 Feb 2012, 17:08    Post_subject:  

OscarTalks try this:

http://smokey01.com/Tman/apps/Multiple_Soundcard_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.

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OscarTalks

Joined: 05 Feb 2012
Posts: 901
Location: London, England

PostPosted: Tue 07 Feb 2012, 21:56    Post_subject:  

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.
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smokey01


Joined: 30 Dec 2006
Posts: 1829
Location: South Australia

PostPosted: Sat 02 Jun 2012, 05:09    Post_subject:  

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/portal/articles/374286

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

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