<div dir="ltr"><div><div><div>thank you Brian, yes i have set bind address. when i enable h245 debugging, all messages have no ip address like this:<br>value OpenLogicalChannel ::= <br> {<br> forwardLogicalChannelNumber 1001<br> forwardLogicalChannelParameters <br> {<br> dataType nullData : NULL<br> multiplexParameters none : NULL<br> }<br> reverseLogicalChannelParameters <br> {<br> dataType audioData : g711Ulaw64k : 20<br> multiplexParameters h2250LogicalChannelParameters : <br> {<br> sessionID 1<br> mediaChannel unicastAddress : iPAddress : <br> {<br> network 'C0A80047'H<br> tsapIdentifier 17680<br> }<br> mediaControlChannel unicastAddress : iPAddress : <br> {<br> network 'C0A80047'H<br> tsapIdentifier 17681<br> }<br> }<br> }<br> }<br><br><br><br>Apr 29 05:09:23.499: H245 FS OLC INCOMING ENCODE BUFFER::= 0003E90C6013800A04000100C0A800474511<br>Apr 29 05:09:23.499: <br>Apr 29 05:09:23.499: H245 FS OLC INCOMING PDU ::=<br><br>value OpenLogicalChannel ::= <br> {<br> forwardLogicalChannelNumber 1002<br> forwardLogicalChannelParameters <br> {<br> dataType audioData : g711Ulaw64k : 20<br> multiplexParameters h2250LogicalChannelParameters : <br> {<br> sessionID 1<br> mediaControlChannel unicastAddress : iPAddress : <br> {<br> network 'C0A80047'H<br> tsapIdentifier 17681<br> }<br> }<br> }<br> }<br><br></div>i think it is problem. cisco does not know where should send rtp packets. am i right??? do you have any hint about it???<br><br></div>thank you for your attention.<br></div>SAM<br></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Apr 28, 2015 at 8:04 PM, Brian Meade <span dir="ltr"><<a href="mailto:bmeade90@vt.edu" target="_blank">bmeade90@vt.edu</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Do you have "<span style="color:rgb(51,51,51);font-family:Arial,sans-serif;font-size:16px;line-height:20px">h323-gateway voip bind srcaddr x.x.x.x" configured on an interface?</span><div><span style="color:rgb(51,51,51);font-family:Arial,sans-serif;font-size:16px;line-height:20px"><br></span></div><div><span style="color:rgb(51,51,51);font-family:Arial,sans-serif;font-size:16px;line-height:20px">You'll want to run "debug h245 asn1" to see if media negotiations as well.</span></div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div class="h5">On Tue, Apr 28, 2015 at 3:55 AM, s m <span dir="ltr"><<a href="mailto:sam.gh1986@gmail.com" target="_blank">sam.gh1986@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5"><div dir="ltr"><div><div><div><div><div>hello guys,<br><br></div>i want to have h323 trunk between cisco 2800 and asterisk 11.13.1 with ooh323 module. i configured both side and have successful call from cisco to asterisk. but when call comes from asterisk to cisco, my phone rings but no audio is heard and call is disconnected after 5 second. i enable "debug voice rtp" in cisco and see the source address for receiving rtp packets is 0.0.0.0<br><br> Apr 28 07:46:34.765: RTP(50493): ps rx s=0.0.0.0(0), d=192.168.0.139(17112), pt=8, ts=BF40, ssrc=2C1690C9<br><br></div>any body knows how should i fix it?<br><br></div>this is my cisco config:<br><br>voice service voip <br> allow-connections h323 to sip<br> allow-connections sip to h323<br> allow-connections sip to sip<br> sip<br>!<br>!<br>!<br>voice class codec 1<br> codec preference 1 g711ulaw<br> codec preference 2 g711alaw<br> codec preference 3 g729r8<br>!<br>dial-peer voice 1 voip<br> destination-pattern 2.+<br> voice-class codec 1<br> session protocol sipv2<br> session target ipv4:192.168.0.240<br>!<br>dial-peer voice 2 voip<br> destination-pattern 1.+<br> voice-class codec 1<br> session target ipv4:<a href="http://192.168.0.71:1720" target="_blank">192.168.0.71:1720</a><br><br></div>any comments or hints are really appreciated.<br></div>SAM<br><div><div><br></div></div></div>
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