Server B(child server) *chan_dahdi.conf*
[trunkgroups] [channels] group=1 context=outbound usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes faxdetect=both callprogress=no progzone=in pulsedial=yes ;busydetect=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.5 txgain=0.5 callgroup=1 pickupgroup=1 pritimer => t309,6000 immediate=no switchtype=euroisdn context=outgoing signalling=pri_cpe pridialplan=unknown group=1 channel => 1-15,17-31 context=outgoing signalling=pri_cpe pridialplan=unknown group=2 channel => 32-46,48-62 context=outgoing signalling=pri_cpe pridialplan=unknown group=3 channel => 63-77,79-93 context=outgoing signalling=pri_cpe pridialplan=unknown group=4 channel => 94-108,110-124 *Sip.conf* [general] pear=type context=hunt_incoming port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=all nat=yes callerid = LITE externip= externhost= autocreatepeer=yes autodomain=yes localnet=192.168.14.112/255.255.255.0 canreinvite=yes language=En allowtransfer=yes realm=telunet domain=192.168.14.112 maxexpiry=3600 defaultexpiry=200 useragent=LITE PBX usereqphone = yes dtmfmode = rfc2833 alwaysauthreject = no regcontext=sipregistrations rtptimeout=3600 rtpholdtimeout=300 rtcachefriends=yes ;--------------------------- SIP DEBUGGING --------------------------------------------------- sipdebug = yes registertimeout=60 registerattempts=5 callgroup=1 pickupgroup=1 callevents=yes Disallow=all Allow=all ;Allow=ulaw ;Allow=gsm Canreinvite=no ;register => <username>:<password>:<username>@<Sip Proxy IP or domain name> [authentication] [4001] type=friend context=outbound defaultuser=4001 secret=4001 callerid="EXT1" host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all [4002] type=friend context=outbound defaultuser=4002 secret=4002 callerid="EXT2" host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all On Sun, Oct 20, 2013 at 11:14 AM, Mitul Limbani <mi...@enterux.in> wrote: > Paste ur chan_dahdi.conf n sip.conf on both servers to pastebin and fwd > link here. > > Mitul > On Oct 20, 2013 11:07 AM, "akhilesh chand" <omakhileshch...@gmail.com> > wrote: > >> Dear All, >> >> 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. >> >> >> exten => 8561,1,Dial(SIP/4001@192.168.14.110,120,tT) >> exten => 8561,n,hangup() >> >> exten => 8562,1,Dial(SIP/4001@192.168.14.110,120,tT) >> exten => 8562,n,hangup() >> >> Call comes into first server successful.But problem with second server >> when call came into second server i got following error: >> >> * chan_sip.c:20063 handle_request_invite: Call from '' to extension >> '4001' rejected because extension not found.* >> >> In one more scenario: >> >> 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: >> >> exten => 5001,1,Dial(SIP/4001@192.168.14.110,120,tT) >> exten => 5001,n,hangup() >> >> >> Regards >> Akhilesh >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users