On 20 May 2013 09:35, Gertjan Wolzak <gw(a)foize.com> wrote:
Hello,
I looked at the asterisk config, looks like you have not configured that.
You need to add a dialplan in the extensions config and trunks in the sip
confiig.
If you want to make the dialplan realtime as well, you should add the
dialplan to the db.
He does have sufficient dialplan in extensions.conf for that he is testing
(calling exten 106 from ext 107)
; add by tm88
[from-sip]
exten => _XXX,1,Dial(SIP/${EXTEN},60)
Also, he does not need any trunk setting in sip.conf as the two extensions
are registering as endpoints with Asterisk.
His issue appears to be that the initial invite from Kamailio has the
header "Contact: 107@(null)" which is incorrect.
-Barry
Subject: Re: [SR-Users] I need you help-----about
Kamailio 3.3.x
andAsterisk 10.7.0 Realtime Integration
Hi Barry:
This issue have not been resolved after following by your method that
modifed the video1_sipregs table struct,attachment is table info and
asterisk log.
Can you help me with this problem? thank you very much!
best
zhengyw
*asterisk log* asterisk_log.txt
<http://sip-router.1086192.n5.nabble.com/file/n118567/asterisk_log.txt>
*dbinfo.txt* dbinfo.txt
<http://sip-router.1086192.n5.nabble.com/file/n118567/dbinfo.txt>
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