<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Verdana; font-size: 10pt; color: #000000'><P><SPAN>when I dail the number that is behind the SIP trunk, I get a busy signal.</SPAN></P>
<P><SPAN>On the pico cell side I don't get any signalling events, so it seems that the call doesn't leave the CCM.<BR><BR>Anyone ideas?<BR></SPAN><BR>----- Oorspronkelijk bericht -----<BR>Van: "Ryan Ratliff" <rratliff@cisco.com><BR>Aan: "Marty van de Veerdonk" <marty@voiceidentity.nl><BR>Cc: "cisco-voip" <cisco-voip@puck.nether.net><BR>Verzonden: Vrijdag 7 mei 2010 15:39:08<BR>Onderwerp: Re: [cisco-voip] Trace call over SIP trunk from CCM<BR><BR>CCM traces or a packet capture will clearly show you the sip signaling.</P>
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<DIV>-Ryan</DIV></SPAN></DIV><BR>
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<DIV>On May 7, 2010, at 6:59 AM, Marty van de Veerdonk wrote:</DIV><BR class=Apple-interchange-newline>
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<DIV style="FONT-SIZE: 10pt; COLOR: rgb(0,0,0); FONT-FAMILY: Verdana">I have a CCM with a SIP trunk to the pico cell gateway. Between a cell phone and internal DN we want to dail via 5 digits.<BR>The call from the cell phone towards the fixed phone is working, the call from the deskphone towards the cell phone isn't.<BR><BR>In the DNA tool of CCM I see that the right SIP trunk is selected (via route pattern, route list, route group), but still the call is handled correctly.<SPAN class=Apple-converted-space> </SPAN><BR>I don't have access to the pico cell gateway.<BR><BR>How can I validate that the call is sent over the SIP trunk with the proper format?<BR><BR>regards, Marty<BR><SPAN><BR></SPAN></DIV>_______________________________________________<BR>cisco-voip mailing list<BR><A href="mailto:cisco-voip@puck.nether.net" target=_blank>cisco-voip@puck.nether.net</A><BR><A href="https://puck.nether.net/mailman/listinfo/cisco-voip" target=_blank>https://puck.nether.net/mailman/listinfo/cisco-voip</A><BR></DIV></SPAN></BLOCKQUOTE></DIV><BR></DIV></div></body></html>