<div dir="ltr">thats what i am passing<br><br>exten=> 456,1,Dial(SIP/<a href="http://adf:123@tulip.axvoice.com:9060/" target="_blank">adf:123@tulip.axvoice.com:9060</a><br><br>adf is username and 123 is the password<br>
<br><br><br><div class="gmail_quote">On Wed, Sep 17, 2008 at 6:01 PM, Alex Balashov <span dir="ltr"><<a href="mailto:abalashov@evaristesys.com">abalashov@evaristesys.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
If there is a secret= on the receiving peer, the sending peer needs to<br>
provide that secret. Along with a username.<br>
<div class="Ih2E3d"><br>
Rizwan Hisham wrote:<br>
<br>
> Hi all,<br>
> I am having a problem with sip uri incoming calls. I have 2 asterisk<br>
</div>> servers both are <a href="http://1.4.2." target="_blank">1.4.2.</a> <<a href="http://1.4.2" target="_blank">http://1.4.2</a>.> i dial sip uri from one asterisk<br>
<div class="Ih2E3d">> server which sends the call to the other asterisk server by seeing its<br>
> domain name in the uri. Invite reaches the recieving asterist server but<br>
> the call is not autenticated. Everytime i see the following NOTICE on<br>
> the asterisk server (caller end)<br>
><br>
> [Sep 17 15:38:24] NOTICE[4594]: chan_sip.c:11968 handle_response_invite:<br>
> Failed to authenticate on INVITE to '"rizwan" <<a href="http://sip:abc@192.168.0.7:9860" target="_blank">sip:abc@192.168.0.7:9860</a><br>
</div>> <<a href="http://sip:abc@192.168.0.7:9860" target="_blank">http://sip:abc@192.168.0.7:9860</a>>>;tag=as089d4adb'<br>
<div class="Ih2E3d">><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" target="_blank">abc:0786@tulip.axvoice.com:9060</a><br>
</div>> <<a href="http://abc:0786@tulip.axvoice.com:9060" target="_blank">http://abc:0786@tulip.axvoice.com:9060</a>>)<br>
<div class="Ih2E3d">> exten=> 123,2,Hangup()<br>
><br>
> exten=> 456,1,Dial(SIP/<a href="http://adf:123@tulip.axvoice.com:9060" target="_blank">adf:123@tulip.axvoice.com:9060</a><br>
</div>> <<a href="http://adf:123@tulip.axvoice.com:9060" target="_blank">http://adf:123@tulip.axvoice.com:9060</a>>)<br>
<div class="Ih2E3d">> 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>
</div>> bindaddr=<a href="http://0.0.0.0" target="_blank">0.0.0.0</a> <<a href="http://0.0.0.0" target="_blank">http://0.0.0.0</a>><br>
<div><div></div><div class="Wj3C7c">> 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>
><br>
> --<br>
> Best Regards<br>
> Rizwan Hisham<br>
><br>
><br>
</div></div>> ------------------------------------------------------------------------<br>
><br>
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<br>
--<br>
Alex Balashov<br>
Evariste Systems<br>
Web : <a href="http://www.evaristesys.com/" target="_blank">http://www.evaristesys.com/</a><br>
Tel : (+1) (678) 954-0670<br>
Direct : (+1) (678) 954-0671<br>
Mobile : (+1) (706) 338-8599<br>
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To UNSUBSCRIBE or update options visit:<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Best Regards<br>Rizwan Hisham<br><br>
</div>