<div dir="ltr">16<br><br><br>Jonathan (debug below...)<br><br><br>CCME#<br>Aug 6 14:43:21.208: //435/1D842714808A/CCAPI/cc_api_call_connected:<br> Interface=0x851ACE38, Data Bitmask=0x1, Progress Indication=NULL(0),<br>
Connection Handle=0<br>Aug 6 14:43:21.208: //435/1D842714808A/CCAPI/cc_api_call_connected:<br> Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)<br>Aug 6 14:43:21.212: //434/xxxxxxxxxxxx/CCAPI/ccConferenceCreate:<br>
(confID=0x836F9A74, callID1=0x1B2, callID2=0x1B3, tag=0x0)<br>Aug 6 14:43:21.212: //434/1D842714808A/CCAPI/ccConferenceCreate:<br> Conference Id=0x836F9A74, Call Id1=434, Call Id2=435, Tag=0x0<br>Aug 6 14:43:21.212: //434/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:<br>
<br>Aug 6 14:43:21.212: cc_api_get_xcode_stream : 4369<br>Aug 6 14:43:21.216: //434/xxxxxxxxxxxx/CCAPI/cc_api_bridge_done:<br> Conference Id=0x65, Source Interface=0x84E27B40, Source Call Id=434,<br> Destination Call Id=435, Disposition=0x0, Tag=0x0<br>
Aug 6 14:43:21.216: //435/xxxxxxxxxxxx/CCAPI/cc_api_bridge_done:<br> Conference Id=0x65, Source Interface=0x851ACE38, Source Call Id=435,<br> Destination Call Id=434, Disposition=0x0, Tag=0xFFFFFFFF<br>Aug 6 14:43:21.216: //434/1D842714808A/CCAPI/cc_generic_bridge_done:<br>
Conference Id=0x65, Source Interface=0x851ACE38, Source Call Id=435,<br> Destination Call Id=434, Disposition=0x0, Tag=0xFFFFFFFF<br>Aug 6 14:43:21.220: //434/1D842714808A/CCAPI/ccConferenceCreate:<br> Call Entry(Conference Id=0x65, Destination Call Id=435)<br>
Aug 6 14:43:21.220: //435/1D842714808A/CCAPI/ccConferenceCreate:<br> Call Entry(Conference Id=0x65, Destination Call Id=434)<br>Aug 6 14:43:21.220: //435/1D842714808A/CCAPI/cc_api_caps_ind:<br> Destination Interface=0x84E27B40, Destination Call Id=434, Source Call Id=435,<br>
Caps(Codec=0x1, Fax Rate=0x1, Vad=0x1,<br> Modem=0x0, Codec Bytes=20, Signal Type=3)<br>Aug 6 14:43:21.220: //435/1D842714808A/CCAPI/cc_api_caps_ind:<br> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),<br>
Playout Max=250(ms), Fax Nom=300(ms))<br>Aug 6 14:43:21.224: //434/1D842714808A/CCAPI/cc_api_caps_ind:<br> Destination Interface=0x851ACE38, Destination Call Id=435, Source Call Id=434,<br> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,<br>
Modem=0x0, Codec Bytes=160, Signal Type=2)<br>Aug 6 14:43:21.224: //434/1D842714808A/CCAPI/cc_api_caps_ind:<br> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),<br> Playout Max=250(ms), Fax Nom=300(ms))<br>
Aug 6 14:43:21.224: //434/1D842714808A/CCAPI/cc_api_caps_ack:<br> Destination Interface=0x851ACE38, Destination Call Id=435, Source Call Id=434,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),<br>
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=2132)<br>Aug 6 14:43:21.228: //435/1D842714808A/CCAPI/cc_api_caps_ack:<br> Destination Interface=0x84E27B40, Destination Call Id=434, Source Call Id=435,<br>
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=2132)<br>Aug 6 14:43:21.228: //434/1D842714808A/CCAPI/cc_process_notify_bridge_done:<br>
Conference Id=0x65, Call Id1=434, Call Id2=435<br>Aug 6 14:43:21.232: //435/1D842714808A/CCAPI/cc_api_voice_mode_event:<br> Call Id=435<br>Aug 6 14:43:21.232: //435/1D842714808A/CCAPI/cc_api_voice_mode_event:<br> Call Entry(Context=0x836C1400)<br>
Aug 6 14:43:21.249: //434/1D842714808A/CCAPI/ccCallConnect:<br> Progress Indication=NULL(0), Data Bitmask=0x1<br>Aug 6 14:43:21.249: //434/1D842714808A/CCAPI/ccCallConnect:<br> Call Entry(Connected=TRUE, Responsed=TRUE)<br>
Aug 6 14:43:21.253: //434/1D842714808A/CCAPI/ccCallNotify:<br> Data Bitmask=0x7, Call Id=434<br>Aug 6 14:43:21.253: //435/1D842714808A/CCAPI/ccCallFeature:<br> Feature Type=25, Call Id=435<br>Aug 6 14:43:21.585: //435/1D842714808A/CCAPI/ccGenerateToneInfo:<br>
Stop Tone On Digit=FALSE, Tone=Null,<br> Tone Direction=Sum Network, Params=0x0, Call Id=435<br>Aug 6 14:43:21.585: //434/1D842714808A/CCAPI/cc_api_call_disconnected:<br> Cause Value=16, Interface=0x84E27B40, Call Id=434<br>
Aug 6 14:43:21.585: //434/1D842714808A/CCAPI/cc_api_call_disconnected:<br> Call Entry(Responsed=TRUE, Cause Value=16, Retry Count=0)<br>Aug 6 14:43:21.589: //434/1D842714808A/CCAPI/ccConferenceDestroy:<br> Conference Id=0x65, Tag=0x0<br>
Aug 6 14:43:21.589: //434/xxxxxxxxxxxx/CCAPI/cc_api_bridge_drop_done:<br> Conference Id=0x65, Source Interface=0x84E27B40, Source Call Id=434,<br> Destination Call Id=435, Disposition=0x0, Tag=0x0<br>Aug 6 14:43:21.593: //435/xxxxxxxxxxxx/CCAPI/cc_api_bridge_drop_done:<br>
Conference Id=0x65, Source Interface=0x851ACE38, Source Call Id=435,<br> Destination Call Id=434, Disposition=0x0, Tag=0x0<br>Aug 6 14:43:21.593: //434/1D842714808A/CCAPI/cc_generic_bridge_done:<br> Conference Id=0x65, Source Interface=0x851ACE38, Source Call Id=435,<br>
Destination Call Id=434, Disposition=0x0, Tag=0x0<br>Aug 6 14:43:21.597: //434/1D842714808A/CCAPI/ccCallDisconnect:<br> Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=16)<br>Aug 6 14:43:21.597: //434/1D842714808A/CCAPI/ccCallDisconnect:<br>
Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)<br>Aug 6 14:43:21.597: //435/1D842714808A/CCAPI/ccCallDisconnect:<br> Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)<br>
Aug 6 14:43:21.601: //435/1D842714808A/CCAPI/ccCallDisconnect:<br> Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)<br>Aug 6 14:43:21.601: //435/1D842714808A/CCAPI/cc_api_get_transfer_info:<br> Transfer Number Is Null<br>
Aug 6 14:43:21.605: //434/1D842714808A/CCAPI/cc_api_call_disconnect_done:<br> Disposition=0, Interface=0x84E27B40, Tag=0x0, Call Id=434,<br> Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)<br>
Aug 6 14:43:21.609: //434/1D842714808A/CCAPI/cc_api_call_disconnect_done:<br> Call Disconnect Event Sent<br>Aug 6 14:43:21.609: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:<br> <br>Aug 6 14:43:21.609: :cc_free_feature_vsa freeing 84F834A8<br>
Aug 6 14:43:21.609: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:<br> <br>Aug 6 14:43:21.609: vsacount in free is 1<br>Aug 6 14:43:21.621: //435/1D842714808A/CCAPI/cc_api_call_feature:<br> Feature Type=6, Interface=0x851ACE38, Call Id=435<br>
Aug 6 14:43:21.661: //435/1D842714808A/CCAPI/cc_api_call_disconnect_done:<br> Disposition=0, Interface=0x851ACE38, Tag=0x0, Call Id=435,<br> <b>Call Entry(Disconnect Cause=16</b>, Voice Class Cause Code=0, Retry Count=0)<br>
Aug 6 14:43:21.661: //435/1D842714808A/CCAPI/cc_api_call_disconnect_done:<br> Call Disconnect Event Sent<br>Aug 6 14:43:21.661: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:<br> <br>Aug 6 14:43:21.661: :cc_free_feature_vsa freeing 84F83658<br>
CCME#<br>Aug 6 14:43:21.661: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:<br> <br>Aug 6 14:43:21.661: vsacount in free is 0<br><br><div class="gmail_quote">On Wed, Aug 6, 2008 at 2:47 PM, ROZA, Ariel <span dir="ltr"><<a href="mailto:Ariel.ROZA@la.logicalis.com">Ariel.ROZA@la.logicalis.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>
<div dir="ltr" align="left"><font size="2" color="#0000ff" face="Arial"><span>What´s the disconnect cause code shown in a debug voip
ccapi inout?</span></font></div><font size="2" color="#0000ff" face="Arial"></font><br>
<div dir="ltr" align="left" lang="en-us">
<hr>
<font size="2" face="Tahoma"><div class="Ih2E3d"><b>From:</b> <a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>
[mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] <b>On Behalf Of </b>Jonathan
Charles<br></div><b>Sent:</b> Miércoles, 06 de Agosto de 2008 04:33 p.m.<br><b>To:</b>
Stephen Collinson<br><b>Cc:</b> OSL CCIE Voice Lab Exam; cisco
voip<br><b>Subject:</b> Re: [cisco-voip] [OSL | CCIE_Voice] IPIPGW Sip to
Sip<br></font><br></div><div><div></div><div class="Wj3C7c">
<div></div>
<div dir="ltr">Right, the question is, how do you configure it
correctly?<br><br>What would cuz the audio to not cut thru and the call to
drop... I was suspecting codec, but it is G711 all the way thru (hard coded on
each dial peer)<br><br><br><br>Jonathan<br><br>
<div class="gmail_quote">On Wed, Aug 6, 2008 at 2:21 PM, Stephen Collinson <span dir="ltr"><<a href="mailto:scollinson@capewave.co.uk" target="_blank">scollinson@capewave.co.uk</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div vlink="blue" link="blue" lang="EN-US">
<div>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; color: navy; font-family: Arial;">SIP to SIP should
work fine, when configured correctly.</span></font></p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; color: navy; font-family: Arial;"></span></font> </p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; color: navy; font-family: Arial;">I was just trying to
give you a scenario where we may need to use it. Apologies if this was not
helpful</span></font></p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; color: navy; font-family: Arial;"></span></font> </p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; color: navy; font-family: Arial;"></span></font> </p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; color: navy; font-family: Arial;"></span></font> </p>
<div>
<div style="text-align: center;" align="center"><font size="3" face="Times New Roman"><span style="font-size: 12pt;">
<hr size="2" width="100%" align="center">
</span></font></div>
<p><b><font size="2" face="Tahoma"><span style="font-weight: bold; font-size: 10pt; font-family: Tahoma;">From:</span></font></b><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma;">
Jonathan Charles [mailto:<a href="mailto:jonvoip@gmail.com" target="_blank">jonvoip@gmail.com</a>] <br><b><span style="font-weight: bold;">Sent:</span></b> 06 August 2008 19:55<br><b><span style="font-weight: bold;">To:</span></b> Stephen Collinson<br>
<b><span style="font-weight: bold;">Cc:</span></b> cisco voip; OSL CCIE Voice Lab
Exam<br><b><span style="font-weight: bold;">Subject:</span></b> Re: [OSL |
CCIE_Voice] IPIPGW Sip to Sip</span></font></p></div>
<div>
<div></div>
<div>
<p><font size="3" face="Times New Roman"><span style="font-size: 12pt;"></span></font> </p>
<div>
<p style="margin-bottom: 12pt;"><font size="3" face="Times New Roman"><span style="font-size: 12pt;">Perhaps I wasn't clear...<br><br><br>There is no
CUE.<br><br>This is a SCCP phone on a CCME, and a SCCP phone on CCM with a SIP
trunk to an IPIPGW, and a SIP dial-peer to
CCME...<br><br><br>Jonathan</span></font></p>
<div>
<p><font size="3" face="Times New Roman"><span style="font-size: 12pt;">On Wed,
Aug 6, 2008 at 1:52 PM, Stephen Collinson <<a href="mailto:scollinson@capewave.co.uk" target="_blank">scollinson@capewave.co.uk</a>> wrote:</span></font></p>
<div vlink="purple" link="blue">
<div>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; color: navy; font-family: Arial;">Perhaps worth looking
at your config.</span></font></p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; color: navy; font-family: Arial;"></span></font> </p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; color: navy; font-family: Arial;">You will need sip to
sip, say to access CUE VM from a CCM SIP trunk.</span></font></p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; color: navy; font-family: Arial;"></span></font> </p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; color: navy; font-family: Arial;">Check all G711
etc.</span></font></p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; color: navy; font-family: Arial;"></span></font> </p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; color: navy; font-family: Arial;">Debug
CCSIP</span></font></p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; color: navy; font-family: Arial;"></span></font> </p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; color: navy; font-family: Arial;"></span></font> </p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; color: navy; font-family: Arial;"></span></font> </p>
<div>
<div style="text-align: center;" align="center"><font size="3" face="Times New Roman"><span style="font-size: 12pt;">
<hr size="2" width="100%" align="center">
</span></font></div>
<p><b><font size="2" face="Tahoma"><span style="font-weight: bold; font-size: 10pt; font-family: Tahoma;">From:</span></font></b><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma;"> <a href="mailto:ccie_voice-bounces@onlinestudylist.com" target="_blank">ccie_voice-bounces@onlinestudylist.com</a> [mailto:<a href="mailto:ccie_voice-bounces@onlinestudylist.com" target="_blank">ccie_voice-bounces@onlinestudylist.com</a>] <b><span style="font-weight: bold;">On Behalf Of </span></b>Jonathan Charles<br>
<b><span style="font-weight: bold;">Sent:</span></b> 06 August 2008
18:41</span></font></p>
<div>
<p><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma;"><br><b><span style="font-weight: bold;">To:</span></b> OSL CCIE Voice Lab Exam; <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
<b><span style="font-weight: bold;">Subject:</span></b> [OSL | CCIE_Voice] IPIPGW Sip to
Sip</span></font></p></div></div>
<p><font size="3" face="Times New Roman"><span style="font-size: 12pt;"></span></font> </p>
<div>
<p><font size="3" face="Times New Roman"><span style="font-size: 12pt;">So, I was
playing with an IPIPGW</span></font></p>
<div>
<div>
<p><font size="3" face="Times New Roman"><span style="font-size: 12pt;"><br><br>CCM on one side (SIP trunk) and CCME on the
other (SIP dial-peer)... call worked, but as soon as you answered it
dropped.<br><br>I changed the SIP dial-peer from the IPIPGW to H.323 (no
session protocol) and RTP cuts thru fine...<br><br>Am I misreading something,
is SIP to SIP not supported, or is my config
retarded?<br><br><br><br>Jonathan</span></font></p></div></div></div></div></div></div>
<p><font size="3" face="Times New Roman"><span style="font-size: 12pt;"></span></font> </p></div></div></div></div></div></blockquote></div><br></div></div></div></div>
</blockquote></div><br></div>