Re: [asterisk-users] Problem with call transfer from one server to another server
Server A ( which contain pri line) *chan_dahdi.conf* [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=no echocancelwhenbridged=no rxgain=0.5 txgain=0.5 relaxdtmf=yes 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] context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=all nat=yes callerid = LITE externip= externhost= autocreatepeer=yes autodomain=yes localnet=192.168.53.197/255.255.255.0 canreinvite=yes language=En allowtransfer=yes realm=telunet domain=192.168.53.197 maxexpiry=3600 defaultexpiry=200 useragent=LITE PBX usereqphone = yes dtmfmode = rfc2833 alwaysauthreject = no regcontext=sipregistrations rtptimeout=60 rtpholdtimeout=300 rtcachefriends=yes ;--- SIP DEBUGGING --- sipdebug = yes registertimeout=60 registerattempts=5 callgroup=1 pickupgroup=1 callevents=yes ;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 [4003] type=friend context=outbound defaultuser=4003 secret=4003 callerid=EXT3 host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all [4004] type=friend context=outbound defaultuser=4004 secret=4004 callerid=EXT4 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
Re: [asterisk-users] Problem with call transfer from one server to another server
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
[asterisk-users] Problem with call transfer from one server to another server
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
Re: [asterisk-users] Problem with call transfer from one server to another server
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
