>From the last time you sent this to the list, here's the response from Richard
Mudgett <[email protected]>...
> 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 <
[email protected]> 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..
>
>
> --
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--
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248
--
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