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i've sent you an email to <a class="moz-txt-link-abbreviated" href="mailto:brian.west@mac.com">brian.west@mac.com</a> with trace<br>
<br>
Brian West wrote:
<blockquote cite="mid:73349256-0393-41F0-BA8E-CE308D9AF92D@mac.com"
type="cite">pastebin the logs
<div><br class="webkit-block-placeholder">
</div>
<div>/b</div>
<div><br>
<div>
<div>On Mar 13, 2008, at 3:47 PM, Josip Djuricic wrote:</div>
<br class="Apple-interchange-newline">
<blockquote type="cite">
<div bgcolor="#ffffff" text="#000000"> Ok, seems like I can't
include any file larger then 100kb, so I can send the trace to people
that want to check it out.<br>
<br>
As i said I still get no audio, but I'm rid of sip 415. From the trace
you can see that there is a RTP from softphone trough FS, all the way
to Cisco AS5300, but it doesn't get out on PSTN. Also FS get's no RTP
from AS5300. This is all happening when I call from PSTN side into Voip
network. When the call is coming from voip phone everything works well.<br>
<br>
Thank you all<br>
<br>
Djuricic wrote:
<blockquote cite="mid:47D96641.8070307@primatel.hr" type="cite"> Hi
there,<br>
<br>
I am including the new trace, this time with sip and rtp.<br>
<br>
I hope that somebody can see something I don't, cause there is still no
audio.<br>
<br>
This time I am doing the trace behind a NAT with a x-lite softphone
because I am at home.<br>
<br>
Josip Djuricic wrote:
<blockquote cite="mid:47D9506D.5080904@primatel.hr" type="cite">
I meant GTD ;)<br>
<br>
Anyway, the traces now looks alot better, no SIP 415 anymore, but there
is still no voice, and rtp seems to be going only from the phone to the
FS<br>
<br>
Josip Djuricic wrote:
<blockquote cite="mid:47D94B82.5000005@primatel.hr" type="cite">
I can disable GDP but still no audio, I'm checkin the traces now<br>
<br>
Brian West wrote:
<blockquote
cite="mid:DE29084F-DF8B-4FFE-9B1A-F9110397A40A@mac.com" type="cite"><span
class="Apple-style-span" style="font-family: 'Lucida Grande';">
<div>Oh this bug has already been sent to MikeJ and Anthm.
This is one strange invite. anyway its being worked on. You might want
to see what in your config causes the addition of the application/gtd
type. I suspect '<span class="Apple-style-span"
style="font-family: Helvetica; white-space: normal;">supplementary-service
pass-through'</span></div>
<div><font class="Apple-style-span" face="Helvetica"><span
class="Apple-style-span" style="white-space: normal;"><br
class="webkit-block-placeholder">
</span></font></div>
<div><font class="Apple-style-span" face="Helvetica"><span
class="Apple-style-span" style="white-space: normal;">/b</span></font></div>
<div><font class="Apple-style-span" face="Helvetica"><span
class="Apple-style-span" style="white-space: normal;"><br
class="webkit-block-placeholder">
</span></font></div>
<div><br class="webkit-block-placeholder">
</div>
INVITE sip:<a moz-do-not-send="true"
href="mailto:38514001148@89.18.32.19">38514001148@89.18.32.19</a>:5080
SIP/2.0 Via: SIP/2.0/UDP 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From:
<sip:<a moz-do-not-send="true" href="mailto:38513391900@10.0.0.19">38513391900@10.0.0.19</a>>;tag=2D49486C-1855
To: <sip:<a moz-do-not-send="true"
href="mailto:38514001148@89.18.32.19">38514001148@89.18.32.19</a>>
Date: Thu, 13 Mar 2008 14:39:39 GMT Call-ID: <a moz-do-not-send="true"
href="mailto:253738A2-F04211DC-81719B71-D67790FB@10.0.0.19">253738A2-F04211DC-81719B71-D67790FB@10.0.0.19</a>
Supported: timer,100rel Min-SE: 1800 Cisco-Guid:
624255889-4030861788-2171509617-3598160123 User-Agent:
Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK,
PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE
Max-Forwards: 6 Remote-Party-ID: <sip:<a moz-do-not-send="true"
href="mailto:38513391900@10.0.0.19">38513391900@10.0.0.19</a>>;party=calling;screen=yes;privacy=off
Timestamp: 1205419179 Contact: <sip:<a moz-do-not-send="true"
href="mailto:38513391900@10.0.0.19">38513391900@10.0.0.19</a>:5060>
Expires: 180 Allow-Events: telephone-event MIME-Version: 1.0
Content-Type: multipart/mixed;boundary=uniqueBoundary Content-Length:
523 --uniqueBoundary Content-Type: application/sdp v=0
o=CiscoSystemsSIP-GW-UserAgent 5947 1401 IN IP4 10.0.0.19 s=SIP Call
c=IN IP4 10.0.0.19 t=0 0 m=audio 16850 RTP/AVP 8 c=IN IP4 10.0.0.19
a=rtpmap:8 PCMA/8000 a=ptime:20 --uniqueBoundary Content-Type:
application/gtd Content-Disposition: signal;handling=optional IAM,
PRN,isdn*,,NET5*, USI,rate,c,3,c,1 USI,lay1,alaw TMR,02
CPN,04,,1,14001148 CGN,04,,1,y,4,13391900 CPC,09 FCI,,,,,,,y,
GCI,25356391f04211dc816e9b71d67790fb --uniqueBoundary--</span><br>
<div>
<div>On Mar 13, 2008, at 9:51 AM, Josip Djuricic wrote:</div>
<br class="Apple-interchange-newline">
<blockquote type="cite"><span class="Apple-style-span"
style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px;">supplementary-service
pass-through</span></blockquote>
</div>
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