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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I’d be interested to see a “show voice call status” from the
2801 while a call is in progress. That will allow us to confirm 100% that the
gateway is using the proper codec, which I assume is G729r8. Also, if you could
share that 2801 config, that’d be great.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] <b>On
Behalf Of </b>Ratko Dodevski<br>
<b>Sent:</b> Saturday, October 03, 2009 7:21 AM<br>
<b>To:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> [cisco-voip] SIP calls problem<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoPlainText style='text-indent:.5in'><span style='font-size:10.0pt;
font-family:"Courier New"'>This is my situation; I have a Cisco VG 2801 whit
FXS and BRI ports. Currently I have analogue phone connected in one of the FXS
ports. This VG needs to connect to softswitch using SIP UA. IN front of the
softswitch there is a SBC (session border controller) which handles the calls.I
managed to create connection and made them to negotiate same codec and use the
SBC as SIP proxy, but I still don’t hear anything. In the attach there is the
current configuration and a test call from and to the phone on the gateway.
Also, I see on the softswitch that when I try to reach a number registered on
the softswitch I get SIP message 488 Not Acceptable Here. I should mention that
calls to other networks (PSTN, GSM) are just fine, problem is just with the
calls to phones on the softswitch. Also, this is a production network, and the
VG is the only new component. Also calls to VG from any network (including VoIP
phones)are working. </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>Only
Difference when debugging is bridge done</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>
Preferred Codec : g729r8, bytes :20</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>
Preferred DTMF relay : rtp-nte</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>
Preferred NTE payload : 101</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>
Early Media :
No</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>
Delayed Media : No</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>
Bridge Done :</span><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:red'> <b>No</b></span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>
New
Media
: No</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>
DSP DNLD Reqd : No</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>When I
receive a call Bridge Done is </span><b><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:red'>NO</span></b><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'> and when
I make a call is </span><b><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:red'>YES</span></b><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>.</span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>What does
this means? Is it something relevant?</span><o:p></o:p></p>
<p class=MsoPlainText style='text-indent:.5in'><o:p> </o:p></p>
<p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>
|-----VoIP_Phones</span><o:p></o:p></p>
<p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>
|</span><o:p></o:p></p>
<p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>Analogue_Phone-----(FXS)CiscoVG(SIP)--------SBC----------SoftSwitch</span><o:p></o:p></p>
<p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>
|</span><o:p></o:p></p>
<p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>
|-----------PSTN</span><o:p></o:p></p>
<p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>
|</span><o:p></o:p></p>
<p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>
|-----------GSM</span><o:p></o:p></p>
<p class=MsoPlainText style='text-indent:.5in'><o:p> </o:p></p>
<p class=MsoNormal>Can anyone provide me with some help or share an expirience?<br
clear=all>
<br>
-- <br>
Ratko<o:p></o:p></p>
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