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); 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="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><p> 
<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.