Hi,<br><br>Thank you for your reply. <br>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).<br><br> my pbx is sending callerid. i can see on other analog phone display.<br>
<br>Yes pbx is sending callerid. When i dial any ext. number from jitsi. On the recipient phone display shows 77 ext number.<br><br>i tried all combination from <a href="https://wiki.asterisk.org/wiki/display/AST/Caller+ID+in+India">https://wiki.asterisk.org/wiki/display/AST/Caller+ID+in+India</a><br>
<br>but it does not work.<br><br><br>any help<br> <br><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Dec 10, 2012 at 9:39 PM, Christopher Harrington <span dir="ltr"><<a href="mailto:chris@acsdi.com" target="_blank">chris@acsdi.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="font-family:arial,sans-serif;font-size:13px">From the last time you sent this to the list, here's the response from <span name="Richard Mudgett" style="font-size:13px">Richard Mudgett</span><span style="white-space:nowrap"> </span><span style="white-space:nowrap"><<a href="mailto:rmudgett@digium.com" target="_blank">rmudgett@digium.com</a>>...</span></div>
<div class="im">
<div style="font-family:arial,sans-serif;font-size:13px"><br></div><div style="font-family:arial,sans-serif;font-size:13px">> my scenario is below<br>><br>> analog phone (10 to 99)------> pbx------>(77)asterisk--------><br>
> <span>jitsi</span>(2000)<br>
><br>> i have analog telephone interface numbered 77 attached with asterisk<br>> and<br>> other sip user is 2000 on <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 display to sip 2000's user. But its not showing to user. Its<br>
> shows<br>> asterisk@my_asterisk_server_ip.<br>><br>> my config. as follow<br>><br>> extension.conf<br>><br>> exten => s,1,Goto(phrase-menu,s,1)<br>><br>> [phrase-menu]<br>><br>> exten => s,1,Answer()<br>
> exten => s,2,Wait(1)<br>> exten => s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)<br>> exten => s,4,Wait(2)<br>> exten => s,5,Set(CALLERID(num,CID)=${CALLERID})<br><br></div></div><span style="font-family:arial,sans-serif;font-size:13px">Remove the CID option. It does nothing in this case because</span><br style="font-family:arial,sans-serif;font-size:13px">
<span style="font-family:arial,sans-serif;font-size:13px">it does not apply. The CID option here only applies to reading</span><br style="font-family:arial,sans-serif;font-size:13px"><span style="font-family:arial,sans-serif;font-size:13px">not writing. Please re-read the documentation for CALLERID().</span><div class="im">
<br style="font-family:arial,sans-serif;font-size:13px">
<div style="font-family:arial,sans-serif;font-size:13px"><br>> exten => s,6,Dial(SIP/${PHRASEID},40,tT)<br>> exten => h,1,Hangup()<br>><br>><br>> and in chan_dahdi.conf<br>><br>> ; General options<br>
> [channels]<br>> usecallerid=yes<br>> hidecallerid=no<br>> callwaiting=yes<br>> threewaycalling=yes<br>> transfer=yes<br>> echocancel=yes<br>> echocancelwhenbridged=yes<br><br>> cidsignalling=dtmf<br>
> cidstart=polarity<br>> callerid=asreceived<br><br>> rxgain=0.0<br>> txgain=0.0<br>> ;FXO Modules<br>> group=1<br>> echocancel=yes<br>> signalling=fxs_ks<br>> context=default<br>> channel=1-20<br>
><br>> #include dahdi-channels.conf<br><br></div></div><span style="font-family:arial,sans-serif;font-size:13px">From your description, the link between the pbx and (77)asterisk</span><br style="font-family:arial,sans-serif;font-size:13px">
<span style="font-family:arial,sans-serif;font-size:13px">is analog. Analog can only pass caller id information in one</span><br style="font-family:arial,sans-serif;font-size:13px"><span style="font-family:arial,sans-serif;font-size:13px">direction. It looks like you have it setup to pass caller id</span><br style="font-family:arial,sans-serif;font-size:13px">
<span style="font-family:arial,sans-serif;font-size:13px">from the pbx to (77)asterisk. Is the pbx even sending caller id?</span><br style="font-family:arial,sans-serif;font-size:13px"><span style="font-family:arial,sans-serif;font-size:13px">Is it sending it in the form you have configured in Asterisk?</span><br style="font-family:arial,sans-serif;font-size:13px">
<span style="font-family:arial,sans-serif;font-size:13px">(dtmf, polarity start, dtmfcidlevel=???)</span><br>
<div class="gmail_extra"><br><br><div class="gmail_quote"><div><div class="h5">On Sun, Dec 9, 2012 at 11:42 PM, Harish Mandowara <span dir="ltr"><<a href="mailto:asteriskhelp2013@gmail.com" target="_blank">asteriskhelp2013@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5"><tt><pre>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
<a href="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" target="_blank">asterisk@my_asterisk_server_ip</a>.
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..</pre></tt>
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<br clear="all"><div><br></div>-- <br>-Chris Harrington<br>
<div>ACSDi Office: 763.559.5800</div><div><div>Mobile Phone: 612.326.4248</div></div><div><br></div><br>
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