Hi Rushowr,<br><br>Thank you for your response. As you said, I executed these below lines:<br><div dir="ltr" align="left"><span class="773294105-18082006"><font color="#ff0000" face="Arial" size="2"><br>exten => s,n,Verbose(2|CallerID info received: ${CALLERID(all)}) ; shows CID info</font></span></div> <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<br><br>And Asterisk is showing this below error on console:<br><br>Executing Verbose("Zap/1-1", "3|CallerID info received: "" <>") in new stack<br>CallerID info received : "" <><br></font></font></span></font></font><font face="Arial"><font size="2"><span class="773294105-18082006"><font color="#0000ff"><font color="#ff0000">Executing Verbose("Zap/1-1", "3|Presentation setting: 0") in new stack<br></font></font></span></font></font><font face="Arial"><font
size="2"><span class="773294105-18082006"><font color="#0000ff"><font color="#ff0000">Presentation setting: 0</font></font></span></font></font><br><font face="Arial"><font size="2"><span class="773294105-18082006"><font color="#0000ff"><font color="#ff0000"><br></font></font></span></font></font>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.<br><br>Looking forward to your reply. Than you.<br><br>Regards,<br>Chandra.<br><br><br><b><i>Rushowr <rushowr@phreaker.net></i></b> wrote:<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> <meta http-equiv="Content-Type" content="text/html; charset=us-ascii"> <meta content="MSHTML 6.00.2900.2963" name="GENERATOR"> <div dir="ltr" align="left"><span class="773294105-18082006"><font color="#0000ff" face="Arial"
size="2">Chandra,</font></span></div> <div dir="ltr" align="left"><span class="773294105-18082006"><font color="#0000ff" face="Arial" size="2"></font></span> </div> <div dir="ltr" align="left"><span class="773294105-18082006"><font color="#0000ff" face="Arial" 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 color="#0000ff" face="Arial" 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 color="#0000ff" face="Arial" size="2"></font></span> </div> <div dir="ltr" align="left"><span class="773294105-18082006"><font color="#0000ff" face="Arial" 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 color="#0000ff" face="Arial" size="2"></font></span> </div> <div dir="ltr" align="left"><span class="773294105-18082006"><font color="#0000ff" face="Arial" 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 color="#ff0000" face="Arial" 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" dir="ltr" align="left" lang="en-us"> <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: <blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); padding-left: 5px; margin-left: 5px;"> <meta content="MSHTML 6.00.2900.2963" name="GENERATOR"> <div dir="ltr" align="left"><span class="351264601-18082006"><font color="#0000ff" face="Arial" 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" dir="ltr" align="left" lang="en-us"> <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="border-left: 2px solid rgb(16, 16, 255); padding-left: 5px; margin-left: 5px;">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 and Colocation provided by Easynews.com --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br>http://lists.digium.com/mailman/listinfo/asterisk-users<br></blockquote><br> <div>__________________________________________________<br>Do You Yahoo!?<br>Tired of spam? Yahoo! Mail has the best spam protection around <br>http://mail.yahoo.com </div></blockquote>_______________________________________________<br>--Bandwidth and Colocation provided by Easynews.com --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options
visit:<br>http://lists.digium.com/mailman/listinfo/asterisk-users<br></blockquote><br> <div> </div><hr size="1"> How low will we go? Check out Yahoo! Messenger’s low <a href="http://us.rd.yahoo.com/mail_us/taglines/postman8/*http://us.rd.yahoo.com/evt=39663/*http://voice.yahoo.com">PC-to-Phone call rates.</a></blockquote>_______________________________________________<br>--Bandwidth and Colocation provided by Easynews.com --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-users<br></blockquote><br><p> 
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