Okay, contact_user seems like do the job. Thanks is contact_user can be anything, or it should be same as username ? I would like to use contact_user for transmitting trunk name into agi script
On Sep 2, 2014, at 7:04 PM, Rainer Piper <[email protected]> wrote: > I use in pjsip.conf > [sipgate1] > type=registration > transport=transport-udp > outbound_auth=sipgate1_auth > server_uri=sip:sipgate.de > client_uri=sip:[email protected] > contact_user=sipgatefilter ; goto the filter in extensions.conf > retry_interval=60 > forbidden_retry_interval=600 > expiration=3600 > > extensions.conf ; i'm cutting the dialed number out of the invite Header and > goto/jump to the extensions > ; incoming VOIP 9716716x SIPGATE > exten => sipgatefilter,1,NoOp(*** ${PJSIP_HEADER(read,To)} *** > ${CALLERID(num)} ***) > same => n,Set(gotoadr=${CUT(PJSIP_HEADER(read,To),@,1)}) > same => n,NoOp(**** 49${gotoadr:-11} ****) > same => n,Goto(49${gotoadr:-11},1) > > ; the filter is jumping to the extensions ... > > ; incoming VOIP 97167160 SIPGATE -> MENU > exten => > 4922897167160,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000&PJSIP/7004&PJSIP/7003&PJSIP/7005,,r) > ; incoming VOIP 97167161 SIPGATE > exten => 4922897167161,1,Dial(PJSIP/7000&PJSIP/7001&PJSIP/7003&PJSIP/7004,,r) > ; incoming VOIP 97167162 SIPGATE ECHO TEST > exten => > 4922897167162,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r) > ; incoming VOIP 97167163 SIPGATE > exten => > 4922897167163,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r) > ; incoming VOIP 97167164 SIPGATE > exten => > 4922897167164,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r) > ; incoming VOIP 97167165 SIPGATE > exten => > 4922897167165,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r) > ; incncoming VOIP 97167166 Mailbox > exten => > 4922897167166,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r) > ; incoming VOIP 97167167 CONF. 1 > exten => > 4922897167167,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r) > ; incoming VOIP 97167168 CONF. 2 > ;exten => > 4922897167168,1,Dial(PJSIP/sip:${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r) > exten => 4922897167168,1,Answer > same => n,echo() > same => n,Hangup() > ; incoming VOIP 97167169 FAX > ;exten => 4922897167169,1,Dial(PJSIP/${EXTEN}@sip.soho-piper.de&PJSIP/7000,,r) > > > Regards > Rainer > > Am 02.09.2014 um 15:08 schrieb Joshua Colp: >> Nick Awesome wrote: >>> register => 73432260005:pass@10001 >>> register => 73432260050:pass@10002 >>> >>> [10001] >>> type=peer >>> host=80.75.132.66 >>> context=dialmap >>> [10002] >>> type=peer >>> host=80.75.132.66 >>> context=dialmap >> >> Can you provide a sip debug of calls to both of these? I'm confused how >> that... works... >> > > > -- > Rainer Piper > Integration engineer > Koeslinstr. 56 > 53123 BONN > GERMANY > Phone: +49 228 97167161 > P2P: sip:[email protected]:5072 (pjsip-test) > -- > _____________________________________________________________________ > -- 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
