Hi,
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)=${CALLERID})
exten => s,6,Dial(SIP/${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..
Do not bother about below message. That is auto-generated by my mail
server.
--
With Warm Regards
Harish Mandowara
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On Fri, Nov 30, 2012 at 04:54:28PM +0530, Harish Mandowara wrote:> > Do not bother about below message. That is auto-generated by my mail > server.[snip]> ------------------------------------------------------------------------------------------------------------------------------- > > This e-mail is for the sole use of the intended recipient(s) and may > contain confidential and privileged information. If you are not the > intended recipient, please contact the sender by reply e-mail and destroy > all copies and the original message. Any unauthorized review, use, > disclosure, dissemination, forwarding, printing or copying of this email > is strictly prohibited and appropriate legal action will be taken. > -------------------------------------------------------------------------------------------------------------------------------I realize this probably seems silly, but I do not think it's in my best interest to ignore threats of appropriate legal action for forwarding this email to someone who might be able to help or archiving on a message board, etc.. Do you think you could talk to the people who manage your mail server and have the disclaimer removed? They may be interested in: http://www.goldmark.org/jeff/stupid-disclaimers -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org
> 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
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
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)=${CALLERID})
exten => s,6,Dial(SIP/${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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