[SR-Users] I need you help-----about Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration

[Barry Flanagan]

On 20 May 2013 04:07, zhengyw <zhengyw at neusoft.com> wrote:

> 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!
>
>
I can see that your sipregs  table is now being properly populated. I have
no idea though why Kamailio is sending the Contact: as 106@(null), which
appears to be the main issue.

<--- SIP read from UDP:10.11.2.47:5060 --->
INVITE sip:107 at 10.11.2.47 SIP/2.0
Record-Route: <sip:10.11.2.47;lr=on;ftag=617>
Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK7c6b.86d570e3.0
Via: SIP/2.0/UDP 10.11.2.37:5060;rport=5060;branch=z9hG4bK2720
From: <sip:106 at 10.11.2.47>;tag=617
To: "107" <sip:107 at 10.11.2.47>
Call-ID: 5246
CSeq: 21 INVITE
Contact: <sip:106@(null)>
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Max-Forwards: 69
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Subject: Phone call
Content-Length: 349


You need to look in the Kamailio logs and try to see what the issue is
on the initial INVITE


-Barry


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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> Sent from the Users mailing list archive at Nabble.com.
>
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