<div dir="ltr">Hi all,<br>I am having a problem with sip uri incoming calls. I have 2 asterisk servers both are <a href="http://1.4.2.">1.4.2.</a> i dial sip uri from one asterisk server which sends the call to the other asterisk server by seeing its domain name in the uri. Invite reaches the recieving asterist server but the call is not autenticated. Everytime i see the following NOTICE on the asterisk server (caller end)<br>
<br>[Sep 17 15:38:24] NOTICE[4594]: chan_sip.c:11968 handle_response_invite: Failed to authenticate on INVITE to '"rizwan" <<a href="http://sip:abc@192.168.0.7:9860">sip:abc@192.168.0.7:9860</a>>;tag=as089d4adb'<br>
<br>My dialplan on caller end is:<br><br>[directcall]<br>exten=> 123,1,Dial(SIP/<a href="http://abc:0786@tulip.axvoice.com:9060">abc:0786@tulip.axvoice.com:9060</a>)<br>exten=> 123,2,Hangup()<br><br>exten=> 456,1,Dial(SIP/<a href="http://adf:123@tulip.axvoice.com:9060">adf:123@tulip.axvoice.com:9060</a>)<br>
exten=> 456,2,Hangup()<br><br>SIP general settings on receiving end are:<br><br>[general]<br>context=uricall-incoming <br>allowoverlap=no <br>bindport=9060 <br>bindaddr=<a href="http://0.0.0.0">0.0.0.0</a> <br>srvlookup=yes <br>
disallow=all<br>allow=ulaw <br>allow=alaw<br>allow=g729<br>allow=gsm<br>relaxdtmf=yes <br>useragent=Asterisk PBX <br>dtmfmode = rfc2833 <br>nat=no<br>canreinvite=yes <br><br>peer settings on receiving end:<br>
<br>[adf]<br>username=adf<br>type=friend<br>secret=XXX<br>qualify=25000<br>nat=yes<br>insecure=port,invite<br>host=dynamic<br>dtmfmode=rfc2833<br>context=sipuri-incoming<br>canreinvite=yes<br>callerid="adf xyz" <123><br>
accountcode=6:0:adf<br>amaflags=default<br>disallow=all<br>allow=g729<br>allow=ulaw<br>allow=alaw<br>allow=gsm<br><br>am i doing something wrong here?<br><br clear="all"><br>-- <br>Best Regards<br>Rizwan Hisham<br><br>
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