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 Post subject: pjsip realtime with a2billing
PostPosted: Mon Sep 28, 2015 5:07 pm 
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Joined: Wed Apr 23, 2014 9:35 am
Posts: 19
Hi, i am testing the new pjsip stack to work with a2billing realtime. Has anyone been successful on this? i am using asterisk13, freepbx 13, a2billing 2.2.0. After configuring everything, my sip clients created in a2billing are being populated by asterisk realtime, but sip clients not regstering, pjsip saying '' No matching endpoint found ''. Any tips will be highly appreciated.

here is my realtime load sipusers from a2billing sippeers.
asrx13*CLI>realtime load sipusers name 9840716502
Column Name Column Value
-------------------- --------------------
id 13
id_cc_card 18
name 9840716502
accountcode 9840716502
regexten 9840716502
amaflags billing
callgroup
callerid
canreinvite YES
context a2billing
DEFAULTip
dtmfmode RFC2833
fromuser
fromdomain
host dynamic
insecure
language
mailbox
md5secret
nat yes
deny
permit
mask
pickupgroup
port
qualify yes
restrictcid
rtptimeout
rtpholdtimeout
secret 77530hu9li0jjcfuc5p5
type friend
username 9840716502
disallow ALL
allow ulaw,alaw,gsm,g729
musiconhold
regseconds 0
ipaddr
cancallforward yes

And here is the pjsip's error that endpoint not found.
[2015-09-28 17:49:50] NOTICE[9125]: res_pjsip/pjsip_distributor.c:347 log_unidentified_request: Request from '<sip:[email protected]>' failed for '192.168.1.68:56272' (callid: 76589N2Y3YTliY2Y4YjdiNTM5MGRkOGFlYmE1NDBhMWMyOGE) - No matching endpoint found
[2015-09-28 17:49:50] NOTICE[9125]: res_pjsip/pjsip_distributor.c:347 log_unidentified_request: Request from '<sip:[email protected]>' failed for '192.168.1.68:56272' (callid: 76589N2Y3YTliY2Y4YjdiNTM5MGRkOGFlYmE1NDBhMWMyOGE) - No matching endpoint found


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 Post subject: Re: pjsip realtime with a2billing
PostPosted: Sun Nov 13, 2016 7:37 pm 
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Joined: Mon Jul 01, 2013 12:08 am
Posts: 2
Hi,

did you able to find an answer to your problem? because i have just installed centos 7 asterisk 13 and freepbx13 and finally a2billing 2.2 and followed the instructions but i got the same problem when i call the access number which is routed to the a2billing.

Thanks


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 Post subject: Re: pjsip realtime with a2billing
PostPosted: Tue Dec 13, 2016 9:18 am 
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Joined: Fri Nov 04, 2016 3:53 pm
Posts: 3
I also tried it briefly but didn't get it to work. a2billing will interact with your asterisk sip_chan stack so you would need to change some code in a2billing (my guess).


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