[asterisk-users] CallerID is not displaying for my incoming calls
Crazy Boy
crazymoonboy at yahoo.com
Thu Aug 17 22:14:23 MST 2006
Hi Rushowr,
Thank you for response.
Here I am giving my config files and error message. Please see it.
zaptel.conf contents:
loadzone = us
defaultzone=us
fxsks=1-4
zapata.conf contents:
[channels]
context=incoming
signalling=fxs_ks
busydetect=1
busycount=7
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
cancallforward=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callerid=asreceived
language=en
usecallerid=yes
hidecallerid=no
echocancel=yes
transfer=yes
immediate=no
musiconhold=default
ringtimeout=8000
cidsignalling=dtmf
cidstart=ring
group=1
callgroup=1
pickupgroup=1
channel => 1
sip.conf contents:
[105]
type=friend
username=105
secret=ravi
callerid="RaviKanth"
host=dynamic
context=leader
canreinvite=no
nat=yes
dtmfmode=rfc2833
allow=all
extensions.conf contents:
[incoming]
exten => s,1,Wait(4)
exten => s,n,Answer
exten => s,n,SetMusicOnHold(default)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(/tmp/virg2)
exten => s,n,Goto(s,1)
exten => s,n,Hangup()
include => leader
[leader]
exten => 105,1,Dial(SIP/105,15)
exten => 105,2,Voicemail(u105)
exten => 105,3,Voicemail(b105)
exten => 105,4,Hangup
exten => _9XXXXXXXXXX,1,Dial(Zap/1/${EXTEN:1}) ; Mobile phone
exten => _5XXXXXXXX,1,Dial(Zap/1/${EXTEN:1}) ; Local Landline
include => internal
[internal]
exten => 105, 1, Dial(SIP/105,15)
When somebody calls from outside (Eg: mobile), I am getting this below error message on Asterisk console:
Error Message:
Aug 17 19:45:41 ERROR[10449]: callerid.c:276 callerid_feed: fsk_serie made mylen < 0 (-8)
Aug 17 19:45:41 WARNING[10449]: chan_zap.c:6087 ss_thread: CallerID feed failed: Success
Aug 17 19:45:41 WARNING[10449]: chan_zap.c:6131 ss_thread: CallerID returned with error on channel 'Zap/1-1'
Please tell me the solution. Looking forward to your kind response.
Thank you.
Regards,
Chandra.
Rushowr <rushowr at phreaker.net> wrote: What's the Dial command being used to pass the call to the Softphones?
---------------------------------
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Crazy Boy
Sent: Wednesday, August 16, 2006 3:23 AM
To: radamson at routers.com; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID is not displaying for my incoming calls
Hi,
As you said, I have changed my configurations. But, callerid is not displaying. What I have to do? Please tell me.
Thanks&Regards,
Chandra.
Rich Adamson <radamson at routers.com> wrote: Crazy Boy wrote:
> Hi Friends,
>
> We have installed Asterisk with Digium 04B card (4 FXO ports). Now, I
> have connected my PSTN line directly to first port. I am making outgoing
> calls and receiving incoming calls successfully through my Asterisk. The
> problem is: When I am receiving a call from outside (PSTN), I am not
> getting the callerid number and getting callerid as "Asterisk" in my
> softphones (XLite). Here I am giving my configuration files.
>
> zaptel.conf file contents:
>
> loadzone = us
> defaultzone=us
> fxsks=1-4
>
> zapata.conf file contents:
>
> [channels]
> context=incoming
> signalling=fxs_ks
> busydetect=1
> busycount=7
> relaxdtmf=yes
> callwaiting=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> cancallforward=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> callerid=asreceived
> language=en
> usecallerid=yes
> hidecallerid=no
> echocancel=yes
> transfer=yes
> immediate=no
> group=1
> callgroup=9
> pickupgroup=9
> channel => 1
The above entries appear to be reasonable and correct. If you have not
properly set rxgain and txgain, it "could" impact callerid. If those
gains are too high/low, asterisk will not properly read the callerid
data when sent to you.
> extensions.conf file contents:
>
> [incoming]
> exten => s,1,Answer
> exten => s,2,SetMusicOnHold(default)
> exten => s,3,DigitTimeout,5
> exten => s,4,ResponseTimeout,10
> exten => s,5,Background(/tmp/virg2)
> exten => s,6,Goto(s,1)
> include => leader
> Got event 18 (Ring Begin)...
> Aug 14 14:11:58 WARNING[26744]: pbx.c:5869 pbx_builtin_dtimeout:
> DigitTimeout is deprecated, please use Set(TIMEOUT(digit)=timeout) instead.
> Aug 14 14:11:58 WARNING[26744]: pbx.c:5845 pbx_builtin_rtimeout:
> ResponseTimeout is deprecated, please use Set(TIMEOUT(response)=timeout)
> instead.
The above two WARNING statements are telling you that either you are
copying those exten=> statements from someone's old config files, or,
you haven't read the asterisk documentation. The message is telling you
that your statement "exten => s,3,DigitTimeout,5" should be replaced
with the Set(TIMEOUT(digit)=timeout) syntax. Your statements are still
executing properly today, but the next time you upgrade asterisk code,
they are likely to fail due to the old syntax not being supported.
Try 'show function TIMEOUT' from your CLI and read it.
> What I have to do to display the PSTN caller number on my softphones?
> Please tell me. Looking forward to your response. Thank you.
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
---------------------------------
How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060817/f9bc5bab/attachment.htm
More information about the asterisk-users
mailing list