[asterisk-users] callerid not received from dahdi

Harish Mandowara asteriskhelp2013 at gmail.com
Mon Dec 10 22:37:42 CST 2012


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..
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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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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