On 20 May 2013 09:35, Gertjan Wolzak <gw@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>
>
>
>
> --
> View this message in context: http://sip-router.1086192.n5.nabble.com/I-need-you-help-about-Kamailio-3-3-x-and-Asterisk-10-7-0-Realtime-Integration-tp118248p118567.html
> Sent from the Users mailing list archive at Nabble.com.
>
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