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<DIV dir=ltr align=left><SPAN class=773294105-18082006><FONT face=Arial
color=#0000ff size=2>Chandra,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=773294105-18082006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=773294105-18082006><FONT face=Arial
color=#0000ff size=2>Unfortunately, I can't help you too much, because I've not
worked a lot with Zap. However, this message:</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=773294105-18082006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=773294105-18082006><FONT face=Arial
size=2>Aug 17 19:45:41 ERROR[10449]: callerid.c:276 callerid_feed:
fsk_serie made mylen < 0 (-8)</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=773294105-18082006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=773294105-18082006><FONT face=Arial
color=#0000ff size=2>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:</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=773294105-18082006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=773294105-18082006><FONT face=Arial
color=#0000ff size=2>[incoming]<BR>exten => s,1,Wait(4)<BR>exten =>
s,n,Answer</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=773294105-18082006><FONT face=Arial
color=#ff0000 size=2>exten => s,n,Verbose(2|CallerID info received:
${CALLERID(all)}) ; shows CID info</FONT></SPAN></DIV>
<DIV dir=ltr align=left><FONT face=Arial><FONT size=2><SPAN
class=773294105-18082006><FONT color=#0000ff><FONT color=#ff0000>exten =>
s,n,Verbose(2|Presentation Setting: ${CALLINGPRES}) ; shows CID
presentation</FONT><BR>exten => s,n,SetMusicOnHold(default)<BR>exten =>
s,n,Set(TIMEOUT(digit)=5)<BR>exten => s,n,Set(TIMEOUT(response)=10)<BR>exten
=> s,n,Background(/tmp/virg2)<BR>exten => s,n,Goto(s,1)<BR>exten =>
s,n,Hangup()<BR>include => leader</FONT></SPAN><SPAN
class=773294105-18082006></SPAN></FONT></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><SPAN class=773294105-18082006></SPAN><FONT color=#0000ff><FONT
face=Arial><FONT size=2>H<SPAN class=773294105-18082006>ope this is helpful in
some way...</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT color=#0000ff><SPAN class=773294105-18082006></SPAN></FONT><SPAN
class=773294105-18082006></SPAN><FONT color=#0000ff><FONT face=Arial><FONT
size=2>R<SPAN
class=773294105-18082006>ushowr</SPAN></FONT></FONT></FONT><BR></DIV>
<BLOCKQUOTE style="MARGIN-RIGHT: 0px">
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Crazy
Boy<BR><B>Sent:</B> Friday, August 18, 2006 1:14 AM<BR><B>To:</B> Asterisk
Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> RE:
[asterisk-users] CallerID is not displaying for my incoming
calls<BR></FONT><BR></DIV>
<DIV></DIV>Hi Rushowr,<BR><BR>Thank you for response.<BR><BR>Here I am giving
my config files and error message. Please see it.<BR><BR><SPAN
style="FONT-WEIGHT: bold">zaptel.conf contents:</SPAN><BR>loadzone =
us<BR>defaultzone=us<BR>fxsks=1-4<BR><BR><SPAN
style="FONT-WEIGHT: bold">zapata.conf
contents:</SPAN><BR>[channels]<BR>context=incoming<BR>signalling=fxs_ks<BR>busydetect=1<BR>busycount=7<BR>relaxdtmf=yes<BR>callwaiting=yes<BR>callwaitingcallerid=yes<BR>threewaycalling=yes<BR>cancallforward=yes<BR>echocancelwhenbridged=yes<BR>rxgain=0.0<BR>txgain=0.0<BR>callerid=asreceived<BR>language=en<BR>usecallerid=yes<BR>hidecallerid=no<BR>echocancel=yes<BR>transfer=yes<BR>immediate=no<BR>musiconhold=default<BR>ringtimeout=8000<BR>cidsignalling=dtmf<BR>cidstart=ring<BR>group=1<BR>callgroup=1<BR>pickupgroup=1<BR>channel
=> 1<BR><BR><SPAN style="FONT-WEIGHT: bold">sip.conf
contents:</SPAN><BR>[105]<BR>type=friend<BR>username=105<BR>secret=ravi<BR>callerid="RaviKanth"<BR>host=dynamic<BR>context=leader<BR>canreinvite=no<BR>nat=yes<BR>dtmfmode=rfc2833<BR>allow=all<BR><BR><SPAN
style="FONT-WEIGHT: bold">extensions.conf
contents:</SPAN><BR>[incoming]<BR>exten => s,1,Wait(4)<BR>exten =>
s,n,Answer<BR>exten => s,n,SetMusicOnHold(default)<BR>exten =>
s,n,Set(TIMEOUT(digit)=5)<BR>exten =>
s,n,Set(TIMEOUT(response)=10)<BR>exten =>
s,n,Background(/tmp/virg2)<BR>exten => s,n,Goto(s,1)<BR>exten =>
s,n,Hangup()<BR>include => leader<BR><BR>[leader]<BR>exten =>
105,1,Dial(SIP/105,15)<BR>exten => 105,2,Voicemail(u105)<BR>exten =>
105,3,Voicemail(b105)<BR>exten => 105,4,Hangup<BR>exten =>
_9XXXXXXXXXX,1,Dial(Zap/1/${EXTEN:1}) ; Mobile phone<BR>exten
=> _5XXXXXXXX,1,Dial(Zap/1/${EXTEN:1}) ;
Local Landline<BR>include => internal<BR><BR>[internal]<BR>exten => 105,
1, Dial(SIP/105,15)<BR><BR>When somebody calls from outside (Eg: mobile), I am
getting this below error message on Asterisk console:<BR><BR><SPAN
style="FONT-WEIGHT: bold">Error Message:</SPAN><BR>Aug 17 19:45:41
ERROR[10449]: callerid.c:276 callerid_feed: fsk_serie made mylen < 0
(-8)<BR>Aug 17 19:45:41 WARNING[10449]: chan_zap.c:6087 ss_thread:
CallerID feed failed: Success<BR>Aug 17 19:45:41 WARNING[10449]:
chan_zap.c:6131 ss_thread: CallerID returned with error on channel
'Zap/1-1'<BR><BR>Please tell me the solution. Looking forward to your kind
response. <BR><BR>Thank you.<BR><BR>Regards,<BR>Chandra.<BR><BR><B><I>Rushowr
<rushowr@phreaker.net></I></B> wrote:
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style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: rgb(16,16,255) 2px solid">
<META content="MSHTML 6.00.2900.2963" name=GENERATOR>
<DIV dir=ltr align=left><SPAN class=351264601-18082006><FONT face=Arial
color=#0000ff size=2>What's the Dial command being used to pass the call to
the Softphones? </FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Crazy
Boy<BR><B>Sent:</B> Wednesday, August 16, 2006 3:23 AM<BR><B>To:</B>
radamson@routers.com; Asterisk Users Mailing List - Non-Commercial
Discussion<BR><B>Subject:</B> Re: [asterisk-users] CallerID is not
displaying for my incoming calls<BR></FONT><BR></DIV>
<DIV></DIV>Hi,<BR><BR>As you said, I have changed my configurations. But,
callerid is not displaying. What I have to do? Please tell
me.<BR><BR>Thanks&Regards,<BR>Chandra.<BR><BR><B><I>Rich Adamson
<radamson@routers.com></I></B> wrote:
<BLOCKQUOTE class=replbq
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: rgb(16,16,255) 2px solid">Crazy
Boy wrote:<BR>> Hi Friends,<BR>> <BR>> We have installed
Asterisk with Digium 04B card (4 FXO ports). Now, I <BR>> have
connected my PSTN line directly to first port. I am making outgoing
<BR>> calls and receiving incoming calls successfully through my
Asterisk. The <BR>> problem is: When I am receiving a call from
outside (PSTN), I am not <BR>> getting the callerid number and
getting callerid as "Asterisk" in my <BR>> softphones (XLite). Here I
am giving my configuration files.<BR>> <BR>> zaptel.conf file
contents:<BR>> <BR>> loadzone = us<BR>> defaultzone=us<BR>>
fxsks=1-4<BR>> <BR>> zapata.conf file contents:<BR>> <BR>>
[channels]<BR>> context=incoming<BR>> signalling=fxs_ks<BR>>
busydetect=1<BR>> busycount=7<BR>> relaxdtmf=yes<BR>>
callwaiting=yes<BR>> callwaitingcallerid=yes<BR>>
threewaycalling=yes<BR>> cancallforward=yes<BR>>
echocancelwhenbridged=yes<BR>> rxgain=0.0<BR>> txgain=0.0<BR>>
callerid=asreceived<BR>> language=en<BR>> usecallerid=yes<BR>>
hidecallerid=no<BR>> echocancel=yes<BR>> transfer=yes<BR>>
immediate=no<BR>> group=1<BR>> callgroup=9<BR>>
pickupgroup=9<BR>> channel => 1<BR><BR>The above entries appear to
be reasonable and correct. If you have not <BR>properly set rxgain and
txgain, it "could" impact callerid. If those <BR>gains are too high/low,
asterisk will not properly read the callerid <BR>data when sent to
you.<BR><BR>> extensions.conf file contents:<BR>> <BR>>
[incoming]<BR>> exten => s,1,Answer<BR>> exten =>
s,2,SetMusicOnHold(default)<BR>> exten =>
s,3,DigitTimeout,5<BR>> exten => s,4,ResponseTimeout,10<BR>>
exten => s,5,Background(/tmp/virg2)<BR>> exten =>
s,6,Goto(s,1)<BR>> include => leader<BR><BR>> Got event 18
(Ring Begin)...<BR>> Aug 14 14:11:58 WARNING[26744]: pbx.c:5869
pbx_builtin_dtimeout: <BR>> DigitTimeout is deprecated, please use
Set(TIMEOUT(digit)=timeout) instead.<BR>> Aug 14 14:11:58
WARNING[26744]: pbx.c:5845 pbx_builtin_rtimeout: <BR>>
ResponseTimeout is deprecated, please use Set(TIMEOUT(response)=timeout)
<BR>> instead.<BR><BR>The above two WARNING statements are telling
you that either you are <BR>copying those exten=> statements from
someone's old config files, or, <BR>you haven't read the asterisk
documentation. The message is telling you <BR>that your statement "exten
=> s,3,DigitTimeout,5" should be replaced <BR>with the
Set(TIMEOUT(digit)=timeout) syntax. Your statements are still
<BR>executing properly today, but the next time you upgrade asterisk
code, <BR>they are likely to fail due to the old syntax not being
supported.<BR><BR>Try 'show function TIMEOUT' from your CLI and read
it.<BR><BR>> What I have to do to display the PSTN caller number on
my softphones? <BR>> Please tell me. Looking forward to your
response. Thank
you.<BR><BR>_______________________________________________<BR>--Bandwidth
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