Hi, Thank you for your reply. 77 ext. number is connected with my asterisk. so any one want to talk with jitsi(pc), they have to dial 77 then 2000#(jitsi sip user number).
my pbx is sending callerid. i can see on other analog phone display. Yes pbx is sending callerid. When i dial any ext. number from jitsi. On the recipient phone display shows 77 ext number. i tried all combination from https://wiki.asterisk.org/wiki/display/AST/Caller+ID+in+India but it does not work. any help On Mon, Dec 10, 2012 at 9:39 PM, Christopher Harrington <ch...@acsdi.com>wrote: > From the last time you sent this to the list, here's the response from Richard > Mudgett <rmudg...@digium.com>... > > > my scenario is below > > > > analog phone (10 to 99)------> pbx------>(77)asterisk--------> > > jitsi(2000) > > > > i have analog telephone interface numbered 77 attached with asterisk > > and > > other sip user is 2000 on jitsi. > > > > I can call from any number from 10 to 99(in intercom) on 77 and ivr > > response will come then i can typed 2000# and call go to 2000 named > > user > > in asterisk. > > > > Now my problem is when i am calling from 10 to 99 (any number) this > > number > > should display to sip 2000's user. But its not showing to user. Its > > shows > > asterisk@my_asterisk_server_ip. > > > > my config. as follow > > > > extension.conf > > > > exten => s,1,Goto(phrase-menu,s,1) > > > > [phrase-menu] > > > > exten => s,1,Answer() > > exten => s,2,Wait(1) > > exten => s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip) > > exten => s,4,Wait(2) > > exten => s,5,Set(CALLERID(num,CID)=${CALLERID}) > > Remove the CID option. It does nothing in this case because > it does not apply. The CID option here only applies to reading > not writing. Please re-read the documentation for CALLERID(). > > > > exten => s,6,Dial(SIP/${PHRASEID},40,tT) > > exten => h,1,Hangup() > > > > > > and in chan_dahdi.conf > > > > ; General options > > [channels] > > usecallerid=yes > > hidecallerid=no > > callwaiting=yes > > threewaycalling=yes > > transfer=yes > > echocancel=yes > > echocancelwhenbridged=yes > > > cidsignalling=dtmf > > cidstart=polarity > > callerid=asreceived > > > rxgain=0.0 > > txgain=0.0 > > ;FXO Modules > > group=1 > > echocancel=yes > > signalling=fxs_ks > > context=default > > channel=1-20 > > > > #include dahdi-channels.conf > > From your description, the link between the pbx and (77)asterisk > is analog. Analog can only pass caller id information in one > direction. It looks like you have it setup to pass caller id > from the pbx to (77)asterisk. Is the pbx even sending caller id? > Is it sending it in the form you have configured in Asterisk? > (dtmf, polarity start, dtmfcidlevel=???) > > > On Sun, Dec 9, 2012 at 11:42 PM, Harish Mandowara < > asteriskhelp2...@gmail.com> wrote: > >> my scenario is below >> >> analog phone (10 to 99)------> pbx------>(77)asterisk--------> jitsi(2000) >> >> i have analog telephone interface numbered 77 attached with asterisk and >> other sip user is 2000 on jitsi. >> >> I can call from any number from 10 to 99(in intercom) on 77 and ivr >> response will come then i can typed 2000# and call go to 2000 named user >> in asterisk. >> >> Now my problem is when i am calling from 10 to 99 (any number) this number >> should display to sip 2000's user. But its not showing to user. Its >> showsasterisk@my_asterisk_server_ip >> <https://webmail.cdac.in/twig/index.php?&s[mailbox]=mail%2Fsent-mail&s[mailGroup]=%2A&s[mail_startmsg]=1&s[sortby]=date&s[sortbyway]=1&s[delete-return]=msgview&s[mailtree]=0%7C&c[f]=mail&c[a]=compose&form[to]=asterisk@my_asterisk_server_ip>. >> >> my config. as follow >> >> extension.conf >> >> exten => s,1,Goto(phrase-menu,s,1) >> >> [phrase-menu] >> >> exten => s,1,Answer() >> exten => s,2,Wait(1) >> exten => s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip) >> exten => s,4,Wait(2) >> exten => s,5,Set(CALLERID(num,CID)=${CALLERID}) >> exten => s,6,Dial(SIP/${PHRASEID},40,tT) >> exten => h,1,Hangup() >> >> >> and in chan_dahdi.conf >> >> ; General options >> [channels] >> usecallerid=yes >> hidecallerid=no >> callwaiting=yes >> threewaycalling=yes >> transfer=yes >> echocancel=yes >> echocancelwhenbridged=yes >> cidsignalling=dtmf >> cidstart=polarity >> callerid=asreceived >> rxgain=0.0 >> txgain=0.0 >> ;FXO Modules >> group=1 >> echocancel=yes >> signalling=fxs_ks >> context=default >> channel=1-20 >> >> #include dahdi-channels.conf >> >> >> any help >> >> thanks.. >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> > > > > -- > -Chris Harrington > ACSDi Office: 763.559.5800 > Mobile Phone: 612.326.4248 > > > > -- > _____________________________________________________________________ > -- 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
