[asterisk-users] callerid not received from dahdi

[Harish Mandowara]

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 <chris at acsdi.com>wrote:

> From the last time you sent this to the list, here's the response from Richard
> Mudgett <rmudgett at 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 at 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 <
> asteriskhelp2013 at 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 at 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
>
>
>
> --
> _____________________________________________________________________
> -- 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
>
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