<html><head></head><body bgcolor="#FFFFFF"><div>Hi</div><div><br></div><div>I have had issues with wiring for incoming calls causing what looks like a hangup when answered but in those cases the call stays up and asterisk thinks its a new call. Have seen it on Avaya too</div><div><br></div><div>If it is wiring can you test a different incoming line?</div><div><br></div><div>Cheers duncan <br><br><br></div><div><br>On 19/04/2012, at 1:54 AM, Tech <<a href="mailto:tech@digital-select.com">tech@digital-select.com</a>> wrote:<br><br></div><div></div><blockquote type="cite"><div><meta http-equiv="Content-Type" content="text/html; charset=us-ascii"><meta name="Generator" content="Microsoft Word 12 (filtered medium)"><!--[if !mso]><style>v\:* {behavior:url(#default#VML);}
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</o:shapelayout></xml><![endif]--><div class="WordSection1"><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">Thanks Dhaval for taking the time to look at my question.<o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><o:p> </o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">I have tried to </span><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">print the hangup cause however as you can see below it doesn't show that section of the dialplan.<o:p></o:p></span></p><p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">I have ammended below the CLI and extensions.conf with the changes I made.</span><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><o:p></o:p></span></p><p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><o:p> </o:p></span></b></p><p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">ASTERISK CLI<o:p></o:p></span></b></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> == Using SIP RTP CoS mark 5<o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Executing [01493857917@sipofficephone:1] Verbose("SIP/lewisphone-0000000d", "2,Call from VoIP network to 01493857917") in new stack<o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> == Call from VoIP network to 01493857917<o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Executing [01493857917@sipofficephone:2] Dial("SIP/lewisphone-0000000d", "DAHDI/1/01493857917") in new stack<o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Called DAHDI/1/01493857917<o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- DAHDI/1-1 answered SIP/lewisphone-0000000d<o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Hanging up on 'DAHDI/1-1'<o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Hungup 'DAHDI/1-1'<o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> == Spawn extension (sipofficephone, 01493857917, 2) exited non-zero on 'SIP/lewisphone-0000000d'<o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><o:p> </o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><o:p> </o:p></span></p><p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">extensions.conf<o:p></o:p></span></b></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">[sipofficephone]<o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><o:p> </o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => _X.,1,Verbose(2,Call from VoIP network to ${EXTEN})<o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> same => n,Dial(DAHDI/1/${EXTEN})<o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> <b>same => n,Verbose(2, Hangup Cause ${HANGUPCAUSE})<o:p></o:p></b></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> same => n,Hangup()<o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><o:p> </o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">[pstnincomming]<o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><o:p> </o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => s,1,Answer()<o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> same => n,Dial(SIP/lewisphone)<o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> same => n,Hangup()<o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><o:p> </o:p></span></p><p class="MsoNormal"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:black">Best Regards<o:p></o:p></span></p><p class="MsoNormal"><u><span lang="EN-US" style="font-size:6.0pt;font-family:"Calibri","sans-serif";color:black"> <o:p></o:p></span></u></p><p class="MsoNormal"><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:black">Lewis <o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:black"><image001.gif></span><span lang="EN-US" style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:black"><o:p></o:p></span></p><p class="MsoNormal"><u><span lang="EN-US" style="font-size:6.0pt;font-family:"Calibri","sans-serif";color:black"> </span></u><span lang="EN-US" style="font-size:6.0pt;font-family:"Calibri","sans-serif";color:black"><o:p></o:p></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><o:p> </o:p></span></p><div style="border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm"><p class="MsoNormal"><b><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>DHAVAL INDRODIYA<br><b>Sent:</b> 18 April 2012 13:18<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] FXO -> GSM Gateway Problem<o:p></o:p></span></p></div><p class="MsoNormal"><o:p> </o:p></p><p class="MsoNormal" style="margin-bottom:12.0pt">Hi,<br><br>It can be codec negotiation error or else plese try to print hangupcause sent from telco<br><br><br><o:p></o:p></p><div><p class="MsoNormal">On Wed, Apr 18, 2012 at 4:27 PM, Tech <<a href="mailto:tech@digital-select.com">tech@digital-select.com</a>> wrote:<o:p></o:p></p><div><div><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Hi,<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">I have a problem where calling "out" of asterisk when the call is answered dahdi hangs up immediately.<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">For example: Sip Client A calls external number. Route: SIP -> FXO -> GSM Gateway ->External Landline.<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">However when that external landline answers the call dahdi hangs up immediately .<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Going the other way is fine (External Landline -> GSM Gateway -> FXO -> SIP).<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">I've tried multiple different internet searches and can't seem to find any information on this problem.<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Below are my config files.<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><b>Sip.conf</b><o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">[office-phone](!) <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">type=friend <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">context=sipofficephone <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">host=dynamic <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">nat=yes <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">#secret=xxxx <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">dtmfmode=auto <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">disallow=all <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">;allow=ulaw <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">allow=alaw <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">allow=GSM<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">[lewisphone](office-phone);lewis mobile<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">secret=xxxx<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><b>Chan_dahdi.conf</b><o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">[channels]<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">signalling=fxs_ks <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">context=pstnincomming<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">group=0<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">channel => 1<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><b>Extensions.conf</b><o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">[sipofficephone]<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">exten => _X.,1,Verbose(2,Call from VoIP network to ${EXTEN})<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> same => n,Dial(DAHDI/1/${EXTEN})<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> same => n,Hangup()<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">[pstnincomming]Diamon<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">exten => s,1,Answer()<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> same => n,Dial(SIP/lewisphone)<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> same => n,Hangup()<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><b>Asterisk CLI Output (Verbose 3)</b><o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">My comments bold.<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> == Using SIP RTP CoS mark 5<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Executing [xxxx@sipofficephone:1] Verbose("SIP/lewisphone-0000000a", "2,Call from VoIP network to xxxx") in new stack<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> == Call from VoIP network to xxxx<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Executing [xxxx@sipofficephone:2] Dial("SIP/lewisphone-0000000a", "DAHDI/1/xxxx") in new stack<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Called DAHDI/1/xxxx<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- DAHDI/1-1 answered SIP/lewisphone-0000000a <b>GSM Gateway Answering Call then Sending it out.</b><o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Hanging up on 'DAHDI/1-1' <b>Dest answering call to which DAHDI hangs up</b><o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Hungup 'DAHDI/1-1'<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> == Spawn extension (sipofficephone, xxxx, 2) exited non-zero on 'SIP/lewisphone-0000000a'<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Best Regards<o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><u><span lang="EN-US" style="font-size:6.0pt"> </span></u><o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Lewis <o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><image001.gif><o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span lang="EN-US" style="font-size:10.0pt;font-family:"Verdana","sans-serif""><a href="http://www.digital-select.com/" target="_blank">www.Digital-Select.com</a></span><o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><u><span lang="EN-US" style="font-size:6.0pt"> </span></u><o:p></o:p></p><p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p></div></div><p class="MsoNormal"><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p></div><p class="MsoNormal"><o:p> </o:p></p></div></div></blockquote><blockquote type="cite"><div><span>--</span><br><span>_____________________________________________________________________</span><br><span>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --</span><br><span>New to Asterisk? Join us for a live introductory webinar every Thurs:</span><br><span> <a href="http://www.asterisk.org/hello">http://www.asterisk.org/hello</a></span><br><span></span><br><span>asterisk-users mailing list</span><br><span>To UNSUBSCRIBE or update options visit:</span><br><span> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></span></div></blockquote></body></html>