<div dir="ltr">If you want us to be able to figure out your H.323 negotiation problem without codec transparent in place, you need to provide these debugs:<div>debug h225 asn1</div><div>debug h245 asn1</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, May 11, 2015 at 6:33 AM, s m <span dir="ltr"><<a href="mailto:sam.gh1986@gmail.com" target="_blank">sam.gh1986@gmail.com</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"><div>thank you Ryan,<br><br></div>i have no problem with sip and it is ok (although i do not know it has MTP or not). i think transcoder may not needed because as i know, it translate two different codecs to each other but in my scenario, both side uses g711alaw. please let me know if i misunderstand it.<br><br></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On Mon, May 11, 2015 at 11:31 AM, Ryan Huff <span dir="ltr"><<a href="mailto:ryanhuff@outlook.com" target="_blank">ryanhuff@outlook.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir="ltr">The reason that is happening is due to media negotiation failure as you mention (both call legs are not offering the same codec capabilities). In that exact configuration, you would need a transcoder (which you could run on the router if you have enough DSP).</p>
<p dir="ltr">Are you sold on h323 or can you do a full SIP trunk (with MTP) between cucm and asterisk?</p><span>
<p dir="ltr">Thanks,</p>
<p dir="ltr">Ryan</p>
<br><br>-------- Original Message --------<br>From: s m <<a href="mailto:sam.gh1986@gmail.com" target="_blank">sam.gh1986@gmail.com</a>><br></span><div><div>Sent: Monday, May 11, 2015 12:25 AM<br>To: Ryan Huff <<a href="mailto:ryanhuff@outlook.com" target="_blank">ryanhuff@outlook.com</a>><br>Subject: Re: [cisco-voip] how codec transparent works?<br>CC: <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br><br><div dir="ltr"><div>hello Ryan,<br><br></div>thank you for your reply. without codec transparent, my phone rings but when i answer i have no voice and it hangs up after 5 seconds. asterisk says "no answer". this is so strange for me. i think media negotiation failed, right? is there any hint to have h323 trunk to asterisk with specific codec (not transparent one)???<br></div><div class="gmail_extra"><br><div class="gmail_quote">On Sun, May 10, 2015 at 4:32 PM, Ryan Huff <span dir="ltr"><<a href="mailto:ryanhuff@outlook.com" target="_blank">ryanhuff@outlook.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir="ltr">Codec transparent just passes sdp through to the other call leg without trying to do media negotiations. </p>
<p dir="ltr">So without codec transparent, what happens?<br>
Thanks,</p>
<p dir="ltr">Ryan</p><div><div>
<br><br>-------- Original Message --------<br>From: s m <<a href="mailto:sam.gh1986@gmail.com" target="_blank">sam.gh1986@gmail.com</a>><br>Sent: Sunday, May 10, 2015 01:19 AM<br>To: <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>Subject: [cisco-voip] how codec transparent works?<br><br><div dir="ltr"><div><div><div>hello everybody,<br><br></div>anybody knows how codec transparent works?<br><br>i
have a strange problem. i want to set h323 trunk between asterisk and
cisco 2800. it only works when i set codec transparent in dial-peer
nodes. show commands in cisco shows that i have a call with g711alaw but
if i set codec g711alaw in dial-peers, i do not have any success call. i
know it is codec compatibility problem. is there any difference
between g711 codecs which cisco and
asterisk utilize? what happened when codec is set to transparent? dose
anyone know anything about it? <br><br></div>thanks is advance<br></div>SAM</div>
</div></div></blockquote></div><br></div>
</div></div></blockquote></div><br></div>
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