No subject
Fri Sep 2 03:59:05 CDT 2011
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=3D???)
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 numbe=
r
> should display to sip 2000's user. But its not showing to user. Its shows=
asterisk at my_asterisk_server_ip <https://webmail.cdac.in/twig/index.php?&s[m=
ailbox]=3Dmail%2Fsent-mail&s[mailGroup]=3D%2A&s[mail_startmsg]=3D1&s[sortby=
]=3Ddate&s[sortbyway]=3D1&s[delete-return]=3Dmsgview&s[mailtree]=3D0%7C&c[f=
]=3Dmail&c[a]=3Dcompose&form[to]=3Dasterisk at my_asterisk_server_ip>.
>
> my config. as follow
>
> extension.conf
>
> exten =3D> s,1,Goto(phrase-menu,s,1)
>
> [phrase-menu]
>
> exten =3D> s,1,Answer()
> exten =3D> s,2,Wait(1)
> exten =3D> s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
> exten =3D> s,4,Wait(2)
> exten =3D> s,5,Set(CALLERID(num,CID)=3D${CALLERID})
> exten =3D> s,6,Dial(SIP/${PHRASEID},40,tT)
> exten =3D> h,1,Hangup()
>
>
> and in chan_dahdi.conf
>
> ; General options
> [channels]
> usecallerid=3Dyes
> hidecallerid=3Dno
> callwaiting=3Dyes
> threewaycalling=3Dyes
> transfer=3Dyes
> echocancel=3Dyes
> echocancelwhenbridged=3Dyes
> cidsignalling=3Ddtmf
> cidstart=3Dpolarity
> callerid=3Dasreceived
> rxgain=3D0.0
> txgain=3D0.0
> ;FXO Modules
> group=3D1
> echocancel=3Dyes
> signalling=3Dfxs_ks
> context=3Ddefault
> channel=3D1-20
>
> #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:
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>
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>
--=20
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248
--f46d04339cdef94d0004d081cb80
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable
<div style=3D"font-family:arial,sans-serif;font-size:13px">From the last ti=
me you sent this to the list, here's the response from=C2=A0<span name=
=3D"Richard Mudgett" class=3D"" style=3D"font-size:13px">Richard Mudgett</s=
pan><span style=3D"white-space:nowrap">=C2=A0</span><span class=3D"" style=
=3D"white-space:nowrap"><<a href=3D"mailto:rmudgett at digium.com">rmudgett=
@digium.com</a>>...</span></div>
<div style=3D"font-family:arial,sans-serif;font-size:13px"><br></div><div s=
tyle=3D"font-family:arial,sans-serif;font-size:13px">> my scenario is be=
low<br>><br>> analog phone (10 to 99)------> pbx------>(77)aste=
risk--------><br>
>=C2=A0<span>jitsi</span>(2000)<br>
><br>> i have analog telephone interface numbered 77 attached with as=
terisk<br>> and<br>> other sip user is 2000 on=C2=A0<span>jitsi</span=
>.<br>><br>> I can call from any number from 10 to 99(in intercom) on=
77 and ivr<br>
> response will come then i can typed 2000# and call go to 2000 named<br=
>> user<br>> in asterisk.<br>><br>> Now my problem is when i am=
calling from 10 to 99 (any number) this<br>> number<br>> should disp=
lay to sip 2000's user. But its not showing to user. Its<br>
> shows<br>> asterisk at my_asterisk_server_ip.<br>><br>> my confi=
g. as follow<br>><br>> extension.conf<br>><br>> exten =3D> s=
,1,Goto(phrase-menu,s,1)<br>><br>> [phrase-menu]<br>><br>> exte=
n =3D> s,1,Answer()<br>
> exten =3D> s,2,Wait(1)<br>> exten =3D> s,3,Read(PHRASEID,/var=
/lib/asterisk/sounds/custom/soip)<br>> exten =3D> s,4,Wait(2)<br>>=
exten =3D> s,5,Set(CALLERID(num,CID)=3D${CALLERID})<br><br></div><span =
style=3D"font-family:arial,sans-serif;font-size:13px">Remove the CID option=
. =C2=A0It does nothing in this case because</span><br style=3D"font-family=
:arial,sans-serif;font-size:13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px">it does not app=
ly. =C2=A0The CID option here only applies to reading</span><br style=3D"fo=
nt-family:arial,sans-serif;font-size:13px"><span style=3D"font-family:arial=
,sans-serif;font-size:13px">not writing. =C2=A0Please re-read the documenta=
tion for CALLERID().</span><br style=3D"font-family:arial,sans-serif;font-s=
ize:13px">
<div style=3D"font-family:arial,sans-serif;font-size:13px"><br>> exten =
=3D> s,6,Dial(SIP/${PHRASEID},40,tT)<br>> exten =3D> h,1,Hangup()<=
br>><br>><br>> and in chan_dahdi.conf<br>><br>> ; General op=
tions<br>
> [channels]<br>> usecallerid=3Dyes<br>> hidecallerid=3Dno<br>>=
callwaiting=3Dyes<br>> threewaycalling=3Dyes<br>> transfer=3Dyes<br>=
> echocancel=3Dyes<br>> echocancelwhenbridged=3Dyes<br><br>> cidsi=
gnalling=3Ddtmf<br>
> cidstart=3Dpolarity<br>> callerid=3Dasreceived<br><br>> rxgain=
=3D0.0<br>> txgain=3D0.0<br>> ;FXO Modules<br>> group=3D1<br>> =
echocancel=3Dyes<br>> signalling=3Dfxs_ks<br>> context=3Ddefault<br>&=
gt; channel=3D1-20<br>
><br>> #include dahdi-channels.conf<br><br></div><span style=3D"font-=
family:arial,sans-serif;font-size:13px">From your description, the link bet=
ween the pbx and (77)asterisk</span><br style=3D"font-family:arial,sans-ser=
if;font-size:13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px">is analog. =C2=
=A0Analog can only pass caller id information in one</span><br style=3D"fon=
t-family:arial,sans-serif;font-size:13px"><span style=3D"font-family:arial,=
sans-serif;font-size:13px">direction. =C2=A0It looks like you have it setup=
to pass caller id</span><br style=3D"font-family:arial,sans-serif;font-siz=
e:13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px">from the pbx to=
(77)asterisk. =C2=A0Is the pbx even sending caller id?</span><br style=3D"=
font-family:arial,sans-serif;font-size:13px"><span style=3D"font-family:ari=
al,sans-serif;font-size:13px">Is it sending it in the form you have configu=
red in Asterisk?</span><br style=3D"font-family:arial,sans-serif;font-size:=
13px">
<span style=3D"font-family:arial,sans-serif;font-size:13px">(dtmf, polarity=
start, dtmfcidlevel=3D???)</span><br>
<div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Sun, Dec 9=
, 2012 at 11:42 PM, Harish Mandowara <span dir=3D"ltr"><<a href=3D"mailt=
o:asteriskhelp2013 at gmail.com" target=3D"_blank">asteriskhelp2013 at gmail.com<=
/a>></span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><tt><pre>my scenario is below
analog phone (10 to 99)------> pbx------>(77)asterisk--------> jit=
si(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 sho=
ws
<a href=3D"https://webmail.cdac.in/twig/index.php?&s[mailbox]=3Dmail%2F=
sent-mail&s[mailGroup]=3D%2A&s[mail_startmsg]=3D1&s[sortby]=3Dd=
ate&s[sortbyway]=3D1&s[delete-return]=3Dmsgview&s[mailtree]=3D0=
%7C&c[f]=3Dmail&c[a]=3Dcompose&form[to]=3Dasterisk at my_asterisk_=
server_ip" target=3D"_blank">asterisk at my_asterisk_server_ip</a>.
my config. as follow
extension.conf
exten =3D> s,1,Goto(phrase-menu,s,1)
[phrase-menu]
exten =3D> s,1,Answer()
exten =3D> s,2,Wait(1)
exten =3D> s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
exten =3D> s,4,Wait(2)
exten =3D> s,5,Set(CALLERID(num,CID)=3D${CALLERID})
exten =3D> s,6,Dial(SIP/${PHRASEID},40,tT)
exten =3D> h,1,Hangup()
and in chan_dahdi.conf
; General options
[channels]
usecallerid=3Dyes
hidecallerid=3Dno
callwaiting=3Dyes
threewaycalling=3Dyes
transfer=3Dyes
echocancel=3Dyes
echocancelwhenbridged=3Dyes
cidsignalling=3Ddtmf
cidstart=3Dpolarity
callerid=3Dasreceived
rxgain=3D0.0
txgain=3D0.0
;FXO Modules
group=3D1
echocancel=3Dyes
signalling=3Dfxs_ks
context=3Ddefault
channel=3D1-20
#include dahdi-channels.conf
any help
thanks..</pre></tt>
<br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href=3D"http://www.api-digital.c=
om" target=3D"_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"http://ww=
w.asterisk.org/hello" target=3D"_blank">http://www.asterisk.org/hello</a><b=
r>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
=C2=A0 =C2=A0<a href=3D"http://lists.digium.com/mailman/listinfo/asterisk-u=
sers" target=3D"_blank">http://lists.digium.com/mailman/listinfo/asterisk-u=
sers</a><br></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
-Chris Harrington<br>
<div>ACSDi=C2=A0Office: 763.559.5800</div><div><div>Mobile Phone:=C2=A0612.=
326.4248</div></div><div><br></div><br>
</div>
--f46d04339cdef94d0004d081cb80--
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