Hi,<br><br>It can be codec negotiation error or else plese try to print hangupcause sent from telco<br><br><br><br><div class="gmail_quote">On Wed, Apr 18, 2012 at 4:27 PM, Tech <span dir="ltr"><<a href="mailto:tech@digital-select.com">tech@digital-select.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div link="blue" vlink="purple" lang="EN-GB"><div><p class="MsoNormal">Hi,<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">I have a problem where calling "out" of asterisk when the call is answered dahdi hangs up immediately.<u></u><u></u></p><p class="MsoNormal">For example: Sip Client A calls external number. Route: SIP -> FXO -> GSM Gateway ->External Landline.<u></u><u></u></p>
<p class="MsoNormal">However when that external landline answers the call dahdi hangs up immediately .<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">Going the other way is fine (External Landline -> GSM Gateway -> FXO -> SIP).<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">I've tried multiple different internet searches and can't seem to find any information on this problem.<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">Below are my config files.<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal"><b>Sip.conf<u></u><u></u></b></p><p class="MsoNormal">[office-phone](!) <u></u><u></u></p><p class="MsoNormal">
type=friend <u></u><u></u></p><p class="MsoNormal">context=sipofficephone <u></u><u></u></p><p class="MsoNormal">host=dynamic <u></u><u></u></p><p class="MsoNormal">nat=yes <u></u><u></u></p><p class="MsoNormal">
#secret=xxxx <u></u><u></u></p><p class="MsoNormal">dtmfmode=auto <u></u><u></u></p><p class="MsoNormal">disallow=all <u></u><u></u></p><p class="MsoNormal">;allow=ulaw <u></u><u></u></p><p class="MsoNormal">
allow=alaw <u></u><u></u></p><p class="MsoNormal">allow=GSM<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">[lewisphone](office-phone);lewis mobile<u></u><u></u></p><p class="MsoNormal">
secret=xxxx<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal"><b>Chan_dahdi.conf<u></u><u></u></b></p><p class="MsoNormal">[channels]<u></u><u></u></p><p class="MsoNormal">signalling=fxs_ks <u></u><u></u></p>
<p class="MsoNormal">context=pstnincomming<u></u><u></u></p><p class="MsoNormal">group=0<u></u><u></u></p><p class="MsoNormal">channel => 1<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">
<u></u> <u></u></p><p class="MsoNormal"><b>Extensions.conf<u></u><u></u></b></p><p class="MsoNormal">[sipofficephone]<u></u><u></u></p><p class="MsoNormal">exten => _X.,1,Verbose(2,Call from VoIP network to ${EXTEN})<u></u><u></u></p>
<p class="MsoNormal"> same => n,Dial(DAHDI/1/${EXTEN})<u></u><u></u></p><p class="MsoNormal"> same => n,Hangup()<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">[pstnincomming]Diamon<u></u><u></u></p>
<p class="MsoNormal">exten => s,1,Answer()<u></u><u></u></p><p class="MsoNormal"> same => n,Dial(SIP/lewisphone)<u></u><u></u></p><p class="MsoNormal"> same => n,Hangup()<u></u><u></u></p><p class="MsoNormal">
<u></u> <u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal"><b>Asterisk CLI Output (Verbose 3)<u></u><u></u></b></p><p class="MsoNormal">My comments bold.<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal"> == Using SIP RTP CoS mark 5<u></u><u></u></p><p class="MsoNormal"> -- Executing [xxxx@sipofficephone:1] Verbose("SIP/lewisphone-0000000a", "2,Call from VoIP network to xxxx") in new stack<u></u><u></u></p>
<p class="MsoNormal"> == Call from VoIP network to xxxx<u></u><u></u></p><p class="MsoNormal"> -- Executing [xxxx@sipofficephone:2] Dial("SIP/lewisphone-0000000a", "DAHDI/1/xxxx") in new stack<u></u><u></u></p>
<p class="MsoNormal"> -- Called DAHDI/1/xxxx<u></u><u></u></p><p class="MsoNormal"> -- DAHDI/1-1 answered SIP/lewisphone-0000000a <b>GSM Gateway Answering Call then Sending it out.<u></u><u></u></b></p><p class="MsoNormal">
-- Hanging up on 'DAHDI/1-1' <b>Dest answering call to which DAHDI hangs up<u></u><u></u></b></p><p class="MsoNormal"> -- Hungup 'DAHDI/1-1'<u></u><u></u></p><p class="MsoNormal"> == Spawn extension (sipofficephone, xxxx, 2) exited non-zero on 'SIP/lewisphone-0000000a'<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal"><span style lang="EN-US">Best Regards<u></u><u></u></span></p><p class="MsoNormal">
<u><span style="font-size:6.0pt" lang="EN-US"> <u></u><u></u></span></u></p>
<p class="MsoNormal"><span style lang="EN-US">Lewis <u></u><u></u></span></p><p class="MsoNormal"><span style><img src="cid:image001.gif@01CD1D54.4DD81E80" alt="digitalselect-e" height="91" width="230"></span><span style lang="EN-US"><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Verdana","sans-serif"" lang="EN-US"><a href="http://www.digital-select.com/" target="_blank">www.Digital-Select.com</a><u></u><u></u></span></p>
<p class="MsoNormal"><u><span style="font-size:6.0pt" lang="EN-US"> </span></u><span style="font-size:6.0pt" lang="EN-US"><u></u><u></u></span></p>
<p class="MsoNormal"><u></u> <u></u></p></div></div><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><br></blockquote></div><br>