[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.

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