GOOD NEWS GVSIP-NAF with Ubuntu

xekon

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I do not see a way to format things as code and nobody likes unformatted text mixed with code, so I just dropped it in pastebin: https://pastebin.com/36azPQn0

Inbound calls are ringing then being picked up by googles voicemail
and outbound calls while monitoring the asterisk CLI appear to be trying to use GVsip, but I get an error:

[2018-07-01 15:56:40] WARNING[1545][C-00000001]: channel.c:6108 request_channel: No channel type registered for 'PJSIP'
[2018-07-01 15:56:40] WARNING[1545][C-00000001]: app_dial.c:2516 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 66 - Channel not implemented)

I am wondering how I would check if my machine is acutally signing in / registering with google.
With Motif I would set all that up in the GUI, but it seems most of this is done in the pjsip_custom.conf file, so I am wondering what all still needs to be done in the GUI for things to work. I have setup the extensions, currently I have 3 extensions with a PAP2T(ata), HT802(ata), and CSipSimple (cell).

Appreciate any help :)
 
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restamp

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Here's a Unix shellscript I wrote sometime back that I've found quite useful in resolving these matters. Hope it helps:
Code:
echo '============================== XMPP ===================================='
asterisk -x 'xmpp show connections' | grep gmailcom
echo '=============================== SIP ===================================='
asterisk -x 'sip show registry' | cut -c 1-30,48-80
asterisk -x 'sip show peers' | grep -v Unmonitored | cut -c 1-48,96-115
echo '============================== IAX2 ===================================='
asterisk -x 'iax2 show peers' | cut -c 1-33,49,58-62,105-132
echo '============================== PJSIP ==================================='
asterisk -x 'pjsip show contacts' | cut -c 1-59,71-
 

xekon

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This is now resolved :) the primary issue was modules.conf had everything related to pjsip set as noload
 

dicko

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After looking at the git source, and you NEED a bogus ASTERISKVERSION reported, I suggest that in the root of your clone you

echo 13.21.99 > .version

BEFORE you ./configure or make it, you can of course be as bogus as you want with the content of that file.
 
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skyeman

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I do not see a way to format things as code and nobody likes unformatted text mixed with code, so I just dropped it in pastebin: https://pastebin.com/36azPQn0

Inbound calls are ringing then being picked up by googles voicemail
and outbound calls while monitoring the asterisk CLI appear to be trying to use GVsip, but I get an error:

[2018-07-01 15:56:40] WARNING[1545][C-00000001]: channel.c:6108 request_channel: No channel type registered for 'PJSIP'
[2018-07-01 15:56:40] WARNING[1545][C-00000001]: app_dial.c:2516 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 66 - Channel not implemented)

I am wondering how I would check if my machine is acutally signing in / registering with google.
With Motif I would set all that up in the GUI, but it seems most of this is done in the pjsip_custom.conf file, so I am wondering what all still needs to be done in the GUI for things to work. I have setup the extensions, currently I have 3 extensions with a PAP2T(ata), HT802(ata), and CSipSimple (cell).

Appreciate any help :)
 

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