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bogus@does.not.exist.com
2011-Sep-02 08:59 UTC
No subject

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 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]=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:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>


--=20
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248

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<div style=3D"font-family:arial,sans-serif;font-size:13px">From
the last time you sent this to the list, here&#39;s the response
from=C2=A0<span name=3D"Richard Mudgett" class=3D""
style=3D"font-size:13px">Richard Mudgett</span><span
style=3D"white-space:nowrap">=C2=A0</span><span
class=3D"" style=3D"white-space:nowrap">&lt;<a
href=3D"mailto:rmudgett at
digium.com">rmudgett@digium.com</a>&gt;...</span></div>
<div
style=3D"font-family:arial,sans-serif;font-size:13px"><br></div><div
style=3D"font-family:arial,sans-serif;font-size:13px">&gt; my
scenario is below<br>&gt;<br>&gt; analog phone (10 to
99)------&gt; pbx------&gt;(77)asterisk--------&gt;<br>
&gt;=C2=A0<span>jitsi</span>(2000)<br>
&gt;<br>&gt; i have analog telephone interface numbered 77
attached with asterisk<br>&gt; and<br>&gt; other sip user is
2000 on=C2=A0<span>jitsi</span>.<br>&gt;<br>&gt;
I can call from any number from 10 to 99(in intercom) on 77 and ivr<br>

&gt; response will come then i can typed 2000# and call go to 2000
named<br>&gt; user<br>&gt; in
asterisk.<br>&gt;<br>&gt; Now my problem is when i am
calling from 10 to 99 (any number) this<br>&gt;
number<br>&gt; should display to sip 2000&#39;s user. But its not
showing to user. Its<br>

&gt; shows<br>&gt; asterisk at
my_asterisk_server_ip.<br>&gt;<br>&gt; my config. as
follow<br>&gt;<br>&gt;
extension.conf<br>&gt;<br>&gt; exten =3D&gt;
s,1,Goto(phrase-menu,s,1)<br>&gt;<br>&gt;
[phrase-menu]<br>&gt;<br>&gt; exten =3D&gt;
s,1,Answer()<br>

&gt; exten =3D&gt; s,2,Wait(1)<br>&gt; exten =3D&gt;
s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)<br>&gt; exten
=3D&gt; s,4,Wait(2)<br>&gt; exten =3D&gt;
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 apply. =C2=A0The CID option here only applies to
reading</span><br
style=3D"font-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 documentation for CALLERID().</span><br
style=3D"font-family:arial,sans-serif;font-size:13px">

<div
style=3D"font-family:arial,sans-serif;font-size:13px"><br>&gt;
exten =3D&gt; s,6,Dial(SIP/${PHRASEID},40,tT)<br>&gt; exten
=3D&gt; h,1,Hangup()<br>&gt;<br>&gt;<br>&gt;
and in chan_dahdi.conf<br>&gt;<br>&gt; ; General
options<br>

&gt; [channels]<br>&gt; usecallerid=3Dyes<br>&gt;
hidecallerid=3Dno<br>&gt; callwaiting=3Dyes<br>&gt;
threewaycalling=3Dyes<br>&gt; transfer=3Dyes<br>&gt;
echocancel=3Dyes<br>&gt;
echocancelwhenbridged=3Dyes<br><br>&gt;
cidsignalling=3Ddtmf<br>

&gt; cidstart=3Dpolarity<br>&gt;
callerid=3Dasreceived<br><br>&gt; rxgain=3D0.0<br>&gt;
txgain=3D0.0<br>&gt; ;FXO Modules<br>&gt;
group=3D1<br>&gt; echocancel=3Dyes<br>&gt;
signalling=3Dfxs_ks<br>&gt; context=3Ddefault<br>&gt;
channel=3D1-20<br>

&gt;<br>&gt; #include
dahdi-channels.conf<br><br></div><span
style=3D"font-family:arial,sans-serif;font-size:13px">From your
description, the link between the pbx and (77)asterisk</span><br
style=3D"font-family:arial,sans-serif;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"font-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-size: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:arial,sans-serif;font-size:13px">Is it sending
it in the form you have configured 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">&lt;<a
href=3D"mailto:asteriskhelp2013 at gmail.com"
target=3D"_blank">asteriskhelp2013 at
gmail.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0
.8ex;border-left:1px #ccc
solid;padding-left:1ex"><tt><pre>my scenario is below

analog phone (10 to 99)------&gt;
pbx------&gt;(77)asterisk--------&gt; 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&#39;s user. But its not showing to user. Its
shows
<a
href=3D"https://webmail.cdac.in/twig/index.php?&amp;s[mailbox]=3Dmail%2Fsent-mail&amp;s[mailGroup]=3D%2A&amp;s[mail_startmsg]=3D1&amp;s[sortby]=3Ddate&amp;s[sortbyway]=3D1&amp;s[delete-return]=3Dmsgview&amp;s[mailtree]=3D0%7C&amp;c[f]=3Dmail&amp;c[a]=3Dcompose&amp;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&gt; s,1,Goto(phrase-menu,s,1)

[phrase-menu]

exten =3D&gt; s,1,Answer()
exten =3D&gt; s,2,Wait(1)
exten =3D&gt; s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
exten =3D&gt; s,4,Wait(2)
exten =3D&gt; s,5,Set(CALLERID(num,CID)=3D${CALLERID})
exten =3D&gt; s,6,Dial(SIP/${PHRASEID},40,tT)
exten =3D&gt; 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.com"
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://www.asterisk.org/hello"
target=3D"_blank">http://www.asterisk.org/hello</a><br>
<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-users"
target=3D"_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</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--
asterisk users - Sep 2011 - No subject

No subject
