(old)Puppy Linux Discussion Forum

Hello n0ukf,

I took a glance at the website and MagicJack appears to be a SIP adapter, probably set up to use their own SIP service. It would almost certainly work, but would not involve PSIP Puppy Phone at all. You have to plug in a normal phone and use that.

If you use the cheaper basic version of MagicJack it establishes networking through the USB port so it would just be a question of whether or not that works in Puppy. I guess it would be OK, but anyway if you have the PLUS version of MagicJack it can connect to your router via an ethernet cable so you don't even need to have a computer switched on at all.

There are other adapters available and other options for using SIP for calling (via PSIP or something else). Don't let the advertising of any one product fool you. Investigate the pros and cons bearing in mind your own needs and read the small print before parting with your money.

Hi guys,
I just signed up with Evaphone and I want to use PP as my softphone in Precise, frugal.. I can just about connect to a sip tester line from mouselike but I've no idea if my Evaphone login details have been correctly inserted, or how I can even check?! I've tried calling several cell numbers using the full international prefix but nothing seems to happen, the log just says "outgoing"

Here's a snap of my input (obviously I've hidden my password). If my settings are wrong or I'm missing something obvious, please put me straight. Many thanks.

Try like this in the first instance:-

SIP URL sip:803145157@evaphone.com

Registrar sip:evaphone.com

Auth Realm evaphone.com (or just *)

Auth User 803145157

Secure voice call unticked

Notes:-
Colon not a dot after initial "sip"
Auth Realm may work with just an asterisk (wildcard)

Close preferences AND quit and restart PSIP

If the settings are correct, click the "Logged Out" button in the main GUI
It should go green and show you as "Logged In" almost instantly.

I'm chatting to CatDude in PSIP now and he made this pic:-

^^^^
Thanks so much for taking the time, Oscar.I really appreciate you and Catdude helping out. Sadly it still doesn't work but that's probably just that Puppy and Evaphone are incompatible.

On that note, can anyone recommend a Skype-like service that they have used successfully within PuppyPhone? It needs to be a service I can top-up online.

Thanks

Hi @RedandWhiteStripes

Is there a chance you can turn off your firewall, then test to see if it works for you? Afterwhich you can turn it back on.

There are 3 things which can affect proper dialer operations on PCs

1. your ISP
2. your router blocking the ports to the PC
3. your firewall

This assumes you have typed your setup definitions correctly.

Let us know

I just signed up for an Evaphone account to take a closer look and am logged in now in PSIP Puppy Phone.

The domain just needs and extra "sip." in it (like the Linphone accounts) so the correct login details will be like this:-

SIP URL sip:803145157@sip.evaphone.com

Registrar sip:sip.evaphone.com

Auth Realm *

Auth User 803145157

Password (your password)

Secure voice call unchecked


Then do as before, close, quit and restart PSIP just to be sure, then click the "Logged Out" button. Should log you in straight away.

Voipbuster is another good company that allows voip calls.

In Australia you can make calls to landlines and Mobiles for free. I can with my Samsung Galaxy S4 using MobileVOIP. I'm sure it would work on Psip too.

VoipBuster allows you to have an account for free.

http://www.voipbuster.com/en/dashboard

Well thank you everyone for the comments and help, I am pleased to say I got all my VOIP accounts logged in to PSIP in the end

Now, ahem, how do I make external calls? I tried using the full international prefix in the 'call' line but nothing happens, no dial tone, no answer.

I searched PSIP's help section, this forum and a big chunk of this long thread but I couldn't find any further information.

EDIT: It's just occured to me that maybe PSIP is...errr...SIP only

So then all I need is a means to record my calls from my x-lite softphone. MHWaveedit only captures one side of the conversations.


Anyone is welcome to add me sip:4323284@localphone.com

Thanks again, everyone.

redandwhitestripes wrote:Well thank you everyone for the comments and help, I am pleased to say I got all my VOIP accounts logged in to PSIP in the end ... .

Great to hear.

Would you share what you found to overcome issues, allowing you to connect?

Thx.

redandwhitestripes wrote:Now, ahem, how do I make external calls? I tried using the full international prefix in the 'call' line but nothing happens, no dial tone, no answer.

Try this as the number to call: sip:+number-to-call@yourdomain, for example sip:+00118882472425@localphone.com (I'm assuming that localphone.com is your provider).

If this doesn't work, you need to ask your provider. If you can call using X-Lite, the above should work. Anyway, let us know.

EDIT: It's just occured to me that maybe PSIP is...errr...SIP only

It is SIP only but most VoIP providers provide SIP-to-PSTN gateway at their end, all you need to do is ask how to access it.

So then all I need is a means to record my calls from my x-lite softphone. MHWaveedit only captures one side of the conversations.

Why don't use use a conventional voice recorder placed near your mouth and the laptop speaker?

cheers!

Thanks everyone, I got things to work by putting the "sip:" prefix where required, exactly as Oscar demonstrated.

I still can't make outgoing calls, I request them exactly as you suggested jamesbond but it flashes in the "call" section for a split-second then vanishes. I'm guessing PSIP simply doesn't do that.

It appears Psip won't allow a call unless the full address is submitted, eg: sip:+61855555555@voipbuster.com however, voipbuster is only expecting to see +61855555555.

This may be the same with evaphone.

I have tried many combinations including the command line version without success.

In earlier versions I have called landlines with Psip via Gizmo.

Any ideas jamesbond?

Some of these providers have that extra sip. in the front of the domain name.

Try calling to sip:+6185555555@sip.voipbuster.com

Just to note also that the number should either be the full numerical international prefix or the international prefix using the "+" sign but not both, so for the UK it would be either sip:0044.......@domain or sip:+44.....@domain but NOT sip:+0044......@domain.

I know folks are busy, but it would be really nice if we could get this issue looked at. Really the standard functionality with any SIP softphone should be that you can log in with a provider and if you have credit you should be able to enter a PSTN number via the GUI dialpad and make a call to normal phones that way. Maybe even a PSIP with video support as well?

OscarTalks, I'm not sure if you are aware but pjsip-2.1, the Psip engine, has video support built in. I just haven't manage to get it working yet although I haven't tried very hard either.

Also You were right on the money with the dialing, the following worked perfectly:
The dialing entry for a landline:
sip:+61855555555@sip.voipbuster.com

Mobile:
sip:+614066555555@sip.voipbuster.com

Voipbuster Configuration for Psip

SIP URL: sip:username@viopbuster.com
Registrar: sip:sip.voipbuster.com
Auth Realm: *
Auth User: Username
Password: Password
Secure voice call: Unticked
Sip Proxy: Blank

Add Friend

Nickname: Whatever
SIP address: sip:+61855555555@sip.voipbuster.com
Category: Whatever

Problem solved

Be sure not to leave spaces in the phone number, it won't work.

Cheers

I've updated the help for Psip.

It now includes connection via no-ip and peer to peer.

It also describes how to make calls to landlines and mobile phones.
http://www.smokey01.com/help/psip/psip-help.html

Cheers

Thanks again everyone, I'll look at the help file and get calling!
PSIP is an excellent example of what Puppy is all about: all the efficiency, all the features, none of the bloat.

EDIT: A few things that would be awesome to see at some point in the future:

a) A way to save and/or switch between SIP profiles. (For example, I'm registered with Mouthmun and Localphone)

b) A default, automatic firewall action each time PSIP starts

c) Compatibility with Pmusic for recorded calls. Right now I have to use GMplayer to playback calls.

Keep up the great work with this superb piece of software

I'll have a look to see what I can setup with Psip. I want to actually setup a Brekeke Sip Server which I've been able to complete successfully. I'll post a tutorial how-to today for everyone. I would like to use such to route a call from a DID provider which I will attempt to make Psip work.

I'll keep posted on details.


Kent C.

With some routers Puppy Phone (and Ekiga) seems not to be usable, at least without doing special things.

After reading http://wiki.ekiga.org/index.php/Manual# ... _firewalls I read http://wiki.ekiga.org/index.php/Enable_ ... g_manually (to be forwarded: 5000 to 5100 UDP, ... ports) and I finished in a page like http://portforward.com/english/routers/ ... /Ekiga.htm:

The router_brand router_model will not allow you to forward enough ports, to run Ekiga.

... and ...

You should try using the DMZ portion of this router if it is available. Alternatively you can try switching the router to bridged mode. You will need to contact your ISP to switch to bridged mode, so they can make the required changed on their end.

I don't see DMZ option and I don't wan't to contact my ISP (I'm not so interested in VoIP now).

When the router allows to forward enough ports, for example with http://portforward.com/english/routers/ ... /Ekiga.htm you can see graphically how to do it.

(I use a Live USB with Slacko Puppy version 5.7. on a Toshiba Satellite laptop with a Speedtouch residential gateway -modem, router, ...-)

Finally, I can connect to my VoIP account unbinding the SIP ALG and the UDP 5060 port. This way: Thanks to smokey01: http://murga-linux.com/puppy/viewtopic. ... 421#796421

I can call for example to the iptel.org utilities but I listen no sound at all (I've tried different Setup-Audio-Input/Output options). And if PuppyPhone is closed I can listen to my music files and record my voice via microphone (I've configured the audio channels running retrovol and alsamixer). But if PuppyPhone is running I cannot. I've used Ekiga in other Linux distro successfully, with the audio OK (without sound problems).

Use sip:music@iptel.org to test for incoming audio and then sip:echo@iptel.org (and speak into the microphone) to test if audio is working both ways. You will need to make sure you have the correct playback and capture settings in retrovol or alsamixer. Also you need to make sure you are using the correct device settings in PSIP Puppy Phone.

Sometimes problems of no audio or one-way audio can be resolved by setting a stun server.

If you are using a sip provider it is probably best not to set up any port forwarding in your router. The sip protocol (along with the stun server if needed) is supposed to handle the opening of all the correct ports to establish the call.