<div dir="ltr">Server B(child server) <br><div><br><b>chan_dahdi.conf</b><br><br>[trunkgroups]<br><br>[channels]<br>group=1<br>context=outbound<br>usecallerid=yes<br>hidecallerid=no<br>callwaiting=yes<br>usecallingpres=yes<br>
callwaitingcallerid=yes<br>threewaycalling=yes<br>transfer=yes<br>canpark=yes<br>cancallforward=yes<br>faxdetect=both<br><br><br>callprogress=no<br>progzone=in<br>pulsedial=yes<br>;busydetect=yes<br><br>callreturn=yes<br>
echocancel=yes<br>echocancelwhenbridged=yes<br>rxgain=0.5<br>txgain=0.5<br>callgroup=1<br>pickupgroup=1<br><br>pritimer => t309,6000<br><br>immediate=no<br><br>switchtype=euroisdn<br><br>context=outgoing<br>signalling=pri_cpe<br>
pridialplan=unknown<br>group=1<br>channel => 1-15,17-31<br><br>context=outgoing<br>signalling=pri_cpe<br>pridialplan=unknown<br>group=2<br>channel => 32-46,48-62<br><br>context=outgoing<br>signalling=pri_cpe<br>pridialplan=unknown<br>
group=3<br>channel => 63-77,79-93<br><br>context=outgoing<br>signalling=pri_cpe<br>pridialplan=unknown<br>group=4<br>channel => 94-108,110-124<br><br><b>Sip.conf</b><br><br>[general]<br>pear=type<br>context=hunt_incoming<br>
port=5060<br>bindaddr=0.0.0.0<br>srvlookup=yes<br>disallow=all<br>allow=all<br>nat=yes<br>callerid = LITE<br>externip=<br>externhost=<br>autocreatepeer=yes<br>autodomain=yes<br>localnet=<a href="http://192.168.14.112/255.255.255.0">192.168.14.112/255.255.255.0</a><br>
canreinvite=yes<br>language=En<br>allowtransfer=yes<br>realm=telunet<br>domain=192.168.14.112<br>maxexpiry=3600<br>defaultexpiry=200<br>useragent=LITE PBX<br>usereqphone = yes<br>dtmfmode = rfc2833<br>alwaysauthreject = no<br>
regcontext=sipregistrations<br>rtptimeout=3600<br>rtpholdtimeout=300<br>rtcachefriends=yes<br>;--------------------------- SIP DEBUGGING ---------------------------------------------------<br>sipdebug = yes<br>registertimeout=60<br>
registerattempts=5<br>callgroup=1<br>pickupgroup=1<br>callevents=yes<br><br>Disallow=all<br>Allow=all<br>;Allow=ulaw<br>;Allow=gsm<br>Canreinvite=no<br><br>;register => <username>:<password>:<username>@<Sip Proxy IP or domain name><br>
<br><br>[authentication]<br><br><br><br>[4001]<br>type=friend<br>context=outbound<br>defaultuser=4001<br>secret=4001<br>callerid="EXT1"<br>host=dynamic<br>nat=no<br>dtfmode=rfc2833<br>disallow=all<br>subscribecontext=outbound<br>
canreinvite=no<br>allow=all<br><br>[4002]<br>type=friend<br>context=outbound<br>defaultuser=4002<br>secret=4002<br>callerid="EXT2"<br>host=dynamic<br>nat=no<br>dtfmode=rfc2833<br>disallow=all<br>subscribecontext=outbound<br>
canreinvite=no<br>allow=all<br><br><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Sun, Oct 20, 2013 at 11:14 AM, Mitul Limbani <span dir="ltr"><<a href="mailto:mitul@enterux.in" target="_blank">mitul@enterux.in</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p>Paste ur chan_dahdi.conf n sip.conf on both servers to pastebin and fwd link here.</p>
<p>Mitul</p>
<div class="gmail_quote"><div><div class="h5">On Oct 20, 2013 11:07 AM, "akhilesh chand" <<a href="mailto:omakhileshchand@gmail.com" target="_blank">omakhileshchand@gmail.com</a>> wrote:<br type="attribution">
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5">
<div dir="ltr"><div><div><div>Dear All,<br><br></div>I have pri with E1 facility that have 30 line and 100 pri number which is provided by service provider.Number started like 23568561,23568562,23568563 and so on. Service provider provide last four digit number for did mapping like 4561,4562,4563.<br>


<br><br>exten => 8561,1,Dial(SIP/<a href="mailto:4001@192.168.14.110" target="_blank">4001@192.168.14.110</a>,120,tT)<br>exten => 8561,n,hangup()<br><br>exten => 8562,1,Dial(SIP/<a href="mailto:4001@192.168.14.110" target="_blank">4001@192.168.14.110</a>,120,tT)<br>


exten => 8562,n,hangup()<br><br></div>Call comes into first server successful.But problem with second server when call came into second server i got following error:<br><br><b> chan_sip.c:20063 handle_request_invite: Call from '' to extension '4001' rejected because extension not found.</b><br>


<br></div><div>In one more scenario:<br><br></div><div>when i create one extension and call forwarding with this extension that time I'm able to transfer call successful the code is given below:   <br><br>exten => 5001,1,Dial(SIP/<a href="mailto:4001@192.168.14.110" target="_blank">4001@192.168.14.110</a>,120,tT)<br>


exten => 5001,n,hangup()<br><br><br></div><div>Regards<br>Akhilesh<br></div></div>
<br></div></div><span class="HOEnZb"><font color="#888888">--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
               <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
   <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></font></span></blockquote></div>
<br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
               <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
   <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br></div>