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free calling using freepbx and asterisk?
I wanted to make free calls to my family over internet. I have one of the online.net servers which I have installed asterisk and freepbx.
using just this setup, can I setup and make calls over Internet? Any providers I can lookup to make free calls? Any help is appreciated.
or am I asking for too much here :-)
Thanks in advance
Comments
Voip client on your end and on their end and you're set...
Thanks @ndelaespada. Any pointers? Any more details on how to accomplish the same? Can I use my nexus5 as a VOIP client as it allows Internet calls?
@plumberg You can install Zoiper on your Nexus 5, it is a good VoIP client.
Create two or more extensions on your Asterisk server using FreePBX GUI, and put each extension on Zoiper on each person's phone, respectively, then they can call each other.
Thanks @hwdsl2 I think I got it working for 1 extn. Any idea for any native android apps which allow video calls through freepbx/ asterisk?
I think video calls are supported in both Zoiper and CSipSimple:
http://code.google.com/p/csipsimple/
To make it work, you need to first enable video in the FreePBX GUI "Asterisk SIP Settings" menu, and reload.
@hwdsl2
Configuration is simple. I have online.net server where FreePbx w Asterisk is installed. I created 2 extensions (111 & 333) on CHAN_SIP. Both extensions are registered on nexus 5 and nexus4 respectively.
[2015-04-11 16:03:51] VERBOSE[9945] chan_sip.c: -- Registered SIP '111' at 192.0.X.X:35408While making a call from 111 to 333, I added a contact in phone [email protected] . When I place the call, it says, called from OUT_OF_NETWORK. On Asterisk server, i see:
[2015-04-11 16:04:04] NOTICE[4770] res_pjsip/pjsip_distributor.c: Request from '"111" <sip:[email protected]>' failed for '172.56.X.X:58814' (callid: [email protected]) - No matching endpoint foundWhat am I doing wrong here? Phone is on Tmobile H network. Any help is appreciated.
Free voip / calls for personal use only ? why you don't go with Skype, gTalk, WhatsApp etc ...
In case you have some brilliant Idea and want to do it at commercial level / large scale then please contact some professional. like http://www.ictinnovations.com/
@plumberg
Zoiper only supports the SIP/IAX protocol, while in Asterisk 12 or 13, port 5060 defaults to the new PJSIP protocol. You need to either change your phone to connect to port 5061 which is the SIP port, or you can switch the PJSIP and SIP ports in FreePBX. Please see:
http://community.freepbx.org/t/cant-register-any-phones/25202
http://pbxinaflash.com/community/index.php?threads/no-matching-endpoint-found.16048/
http://community.freepbx.org/t/asterisk-12-chan-sip-technology-does-not-register/23062
To be pedantic: chan_pjsip and chan_sip both implement the SIP protocol, the chan_pjsip module just uses a third party SIP stack called PJSIP to do a lot of the heavy work. FreePBX has each running on different ports so you need to ensure you are using the correct port for what you have configured. That's what your log message indicated, you had a device configured in chan_sip but it connected to the port running chan_pjsip which had no idea about it.
To be further pedantic: FreePBX is the one who has set things up this way, it's not the default Asterisk configuration. The chan_pjsip module has no defaults like this - you have to explicitly configure it to listen on a port.