[SR-Users] kamailio and asterisk proiblem

[Daniel-Constantin Mierla]

  Hello,

you must configure asterisk to trust the traffic coming from Kamailio IP 
and not authenticate.

Cheers,
Daniel

On 8/29/10 5:00 PM, chuks at cybernergies.com wrote:
>
>     Hello
>      I have been having issues with kamailio and asterisk realtime. I
>     have used all the configurations posted, but it just has not
>     worked for me. What I am trying to do is to use asterisk as PSTN &
>     voicemail for kamailio. But I keep getting this 401 not authorized
>     from asterisk like this:
>
>     ========
>     --- (19 headers 19 lines) ---
>     Sending to 99.89.26.17:5060 (NAT)
>     Using INVITE request as basis request - LCklNT_HoIeTxCB_8cSIf9efRNvkcR
>     > doing dnsmgr_lookup for '99.89.26.17'
>         -- adding dns manager for '99.89.26.17'
>     Scheduling destruction of SIP dialog
>     '3d67147a652fe8641c536eb92383ad74 at 99.89.26.17' in 32000 ms
>     (Method: NOTIFY)
>     Reliably Transmitting (NAT) to 99.89.26.17:5060:
>     NOTIFY sip:1000 at 99.89.26.17 SIP/2.0
>     Via: SIP/2.0/UDP 99.89.26.18:5060;branch=z9hG4bK00545e8e;rport
>     Max-Forwards: 70
>     From: "asterisk" <sip:1000 at 99.89.26.17>;tag=as4d28adb7
>     To: <sip:1000 at 99.89.26.17>
>     Contact: <sip:1000 at 99.89.26.18:5060>
>     Call-ID: 3d67147a652fe8641c536eb92383ad74 at 99.89.26.17
>     CSeq: 102 NOTIFY
>     User-Agent: Asterisk PBX
>     Event: message-summary
>     Content-Type: application/simple-message-summary
>     Content-Length: 84
>
>     Messages-Waiting: no
>     Message-Account: sip:1 at 99.89.26.17
>     Voice-Message: 0/0 (0/0)
>
>     ---
>     Found peer '1000' for '1000' from 99.89.26.17:5060
>
>     <--- Reliably Transmitting (NAT) to 99.89.26.17:5060 --->
>     SIP/2.0 401 Unauthorized
>     Via: SIP/2.0/UDP
>     99.89.26.17;branch=z9hG4bK354f.e07b39b2.0;received=99.89.26.17;rport=5060
>     Via: SIP/2.0/UDP
>     192.168.1.101:5060;branch=z9hG4bKj4sndhbgkh863bu8fpr61tb;rport=5060
>     From: <sip:1000 at 99.89.26.17>;tag=5tnt79v6phhc689kd5vh
>     To: <sip:+2348023098407 at 99.89.26.17;user=phone>;tag=as21d7a164
>     Call-ID: LCklNT_HoIeTxCB_8cSIf9efRNvkcR
>     CSeq: 1710 INVITE
>     Server: Asterisk PBX
>     Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
>     NOTIFY, INFO, PUBLISH
>     Supported: replaces, timer
>     WWW-Authenticate: Digest algorithm=MD5, realm="99.89.26.17",
>     nonce="16c11ac7"
>     Content-Length: 0
>
>     ========
>
>     asterisk and kamailio are on different server, and I have put the
>     IP of asterisk in trusted table in kamailio db. My kamailio.cfg is:
>
>

-- 
Daniel-Constantin Mierla
http://www.asipto.com

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