<div>Hi Guys,</div><div>Hope fully somebody out there will have experienced this and can shed some light on how it was overcome.</div><div><br></div><div>Current setup includes asterisk 1.6.2.11, GNU GK and a Quintum Tenor CMS on the same lan. Earlier I was unable to make a sip call from the CMS back to a sip client registered on my asterisk box. So I moved onto passing the call from the Quintum CMS to a Quintum Tenore DX which is also on the same lan and is registered as a sip client to the available asterisk machine but can't get it to route a call out to another sip client. Following are the resulting logs when this call fails.</div>
<div><br></div><div>10.152.0.7 Quintum CMS</div><div>10.152.0.120 Asterisk</div><div>10.152.0.155 GnuGK</div><div>10.152.0.248 Quintum DX</div><div><br></div><div><br></div><div><div>ch |01/01| 2010/09/10|16:53:42:560 |h323[-1774588488] [0]: h323mgr:RcvIncomingCall</div>
<div> [ch] Call Source <<a href="http://10.152.0.155:46124">10.152.0.155:46124</a>></div><div><br></div><div><span class="Apple-tab-span" style="white-space:pre"> </span> ch |01/01| 2010/09/10|16:53:42:635 |CH: iprg name=IPRG-default.</div>
<div> ocall[177]: FaxRelay:1 FaxModemCoding:0 RXGain:0 TXGain:0 IdleNoiseLevel:-7000 QOSType:0 QOSValue:176</div><div> ocall[177]: rejectNoCID:0 minimumANILength:1 rejectNoCIDCause:21</div>
<div> ocall[177]:RcvSetup() iprgIndex=0 numIncCalls=0/-1 maxTalkTime=0 extRouteReq=0.</div><div> bandwidth info: max=-1 cur=12600.</div><div>
ocall [177]: Fast start element present.</div><div>ch |01/01| 2010/09/10|16:53:42:640 |calling->called media type=9(4).</div><div> called->calling media type=9(4).</div>
<div> H323 [177]:Setting remote rtp port=<a href="http://10.152.0.7:10312">10.152.0.7:10312</a> ps=0.</div><div>ch |01/01| 2010/09/10|16:53:42:645 |ocall [177]:Remote side packet saver version = 3.</div>
<div> translateCID =12345678.</div><div> UNSPECIFIED_INDEX</div><div><br></div><div> h323[177] [0]: tcall:doTranslation inc=1 iprgIndex=0.</div>
<div> CallInfo [178]: origCalled.digit(61008) callingparty (12345678)</div><div>.</div><div>ch |01/01| 2010/09/10|16:53:42:655 |h323[178] [-1976852028]: ocall:stackSendCallProc</div>
<div> ocall:setIPMedia():Setting My IP to 10.152.0.248</div><div> H323OrigCall::stackSendCallProc(), EncryptRTP(10356-->0x147a)</div><div>
ch |01/01| 2010/09/10|16:53:42:660 |Routing requested for:</div><div> Orig#=61008 NPI=1(public) TON=1 Normalized#=61008 NPI=1(public) TON=1</div><div> Incoming SRC:10.152.0.7 CallingParty:12345678</div>
<div> Route code= selected TG=0</div><div> 0 match(es) found:</div><div> Route response [178]: result=0 cause=34.</div>
<div> use the cause from previous attempt ifone available</div><div> CallInfo[178]: fail event. cause=34 legno=0 leg=0 sentLeg=0.</div><div>
<br></div><div> CallInfo [178]: discTickm(-179930606)connTickm(0) duration (0)</div><div>.</div><div> CallInfo[178]: send eventFailed 0.</div>
<div> h323[178] [-1976852028]: ocall:stackSendRelease</div></div><div><br></div><div><br></div><div>Any ideas on what i can try please? let me know if you need additional configuration files please. thank you.</div>
<div><br></div><br clear="all">Regards,<br>Jas<br>
<pre>-------------------------------------------------
IMPORTANT NOTICE : The information in this email is confidential and may also be privileged.
If you are not the intended recipient, any use or dissemination of the information and any
disclosure or copying of this email is unauthorised and strictly prohibited. If you have received
this email in error, please promptly inform us by reply email or telephone. You should also delete
this email and destroy any hard copies produced.