[asterisk-users] CallerID is not displaying for my incoming calls
Crazy Boy
crazymoonboy at yahoo.com
Sun Aug 20 22:44:06 MST 2006
Hi Rushowr,
Thank you for your response. As you said, I executed these below lines:
exten => s,n,Verbose(2|CallerID info received: ${CALLERID(all)}) ; shows CID info
exten => s,n,Verbose(2|Presentation Setting: ${CALLINGPRES}) ; shows CID presentation
And Asterisk is showing this below error on console:
Executing Verbose("Zap/1-1", "3|CallerID info received: "" <>") in new stack
CallerID info received : "" <>
Executing Verbose("Zap/1-1", "3|Presentation setting: 0") in new stack
Presentation setting: 0
As per my knowledge, I have to do some modifications in chan_zap.c file to get callerid in India. But, I dont know what modifications i have to do? Can you pleaes tell me.
Looking forward to your reply. Than you.
Regards,
Chandra.
Rushowr <rushowr at phreaker.net> wrote: Chandra,
Unfortunately, I can't help you too much, because I've not worked a lot with Zap. However, this message:
Aug 17 19:45:41 ERROR[10449]: callerid.c:276 callerid_feed: fsk_serie made mylen < 0 (-8)
Seems interesting...My guess is that the callerid information is corrupted or something, because it's a negative value, not a 0 or positive. Possibly you have your CID Signalling set to the wrong value... One thing you could try just to get a better idea of what (if anything) is actually read from the callerid and what the presentation is set to, is to modify the your dialplan to output the data to your console (I use verbose 2 so I don't have to read the extra info:
[incoming]
exten => s,1,Wait(4)
exten => s,n,Answer
exten => s,n,Verbose(2|CallerID info received: ${CALLERID(all)}) ; shows CID info
exten => s,n,Verbose(2|Presentation Setting: ${CALLINGPRES}) ; shows CID presentation
exten => s,n,SetMusicOnHold(default)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(/tmp/virg2)
exten => s,n,Goto(s,1)
exten => s,n,Hangup()
include => leader
Hope this is helpful in some way...
Rushowr
---------------------------------
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Crazy Boy
Sent: Friday, August 18, 2006 1:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] CallerID is not displaying for my incoming calls
Hi Rushowr,
Thank you for response.
Here I am giving my config files and error message. Please see it.
zaptel.conf contents:
loadzone = us
defaultzone=us
fxsks=1-4
zapata.conf contents:
[channels]
context=incoming
signalling=fxs_ks
busydetect=1
busycount=7
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
cancallforward=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callerid=asreceived
language=en
usecallerid=yes
hidecallerid=no
echocancel=yes
transfer=yes
immediate=no
musiconhold=default
ringtimeout=8000
cidsignalling=dtmf
cidstart=ring
group=1
callgroup=1
pickupgroup=1
channel => 1
sip.conf contents:
[105]
type=friend
username=105
secret=ravi
callerid="RaviKanth"
host=dynamic
context=leader
canreinvite=no
nat=yes
dtmfmode=rfc2833
allow=all
extensions.conf contents:
[incoming]
exten => s,1,Wait(4)
exten => s,n,Answer
exten => s,n,SetMusicOnHold(default)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(/tmp/virg2)
exten => s,n,Goto(s,1)
exten => s,n,Hangup()
include => leader
[leader]
exten => 105,1,Dial(SIP/105,15)
exten => 105,2,Voicemail(u105)
exten => 105,3,Voicemail(b105)
exten => 105,4,Hangup
exten => _9XXXXXXXXXX,1,Dial(Zap/1/${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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