<div dir="ltr"><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">What's this dial-peer below?</p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">
<br></p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">dial-peer voice 3000 voip<u></u><u></u></p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">
destination-pattern 3...$ <<<<phone back at corporate<u></u><u></u></p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">session target ipv4:10.82.65.11<u></u><u></u></p>
<p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">codec g729 <b>---> is this r8 or some other flavor?</b><u></u><u></u></p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">
dtmf-relay h245-alphanumeric<u></u><u></u></p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">no vad</p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">
<br></p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">I'd try 2 things:</p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">
1. create </p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">voice class codec 1</p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">
codec preference 1 g729r8</p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif"> codec preference 2 g729br8</p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">
codec preference 3 g711ulaw</p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif"><br></p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">
and then assign it on the dial-peer as 'voice-class codec 1'. This way i'm offering multiple codecs for negotiation.</p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">
<br></p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">2. Convert the transcoder to universal mode and then try the call again. Perhaps the router is trying to xcode between 2 flavors of g729, and this can only be done by a universal transcoder. You'll have lesser 'max sessions' but it's worth a try. May tell us what codecs are being used on each leg.</p>
<p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif"><br></p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif"><br></p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">
Further, there are always debugs such as what Amit pointed out to.</p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif"><br></p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">
Thanks</p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">Sreekanth</p><p class="MsoNormal" style="margin:0in 0in 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">
<br></p></div><div class="gmail_extra"><br><br><div class="gmail_quote">On 25 April 2014 23:04, Jason Aarons (AM) <span dir="ltr"><<a href="mailto:jason.aarons@dimensiondata.com" target="_blank">jason.aarons@dimensiondata.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div lang="EN-US" link="#0563C1" vlink="#954F72"><div><p class="MsoNormal">CME 8.6 >> Existing 7965s work great at this site. We added some 9971s. If 9971s dial an extension back to Corporate CallManager across a h323 dial peer the call sets up. You answer at corporate and after about 10 seconds hear fast busy. The 9971 has g729r8 the dial-peer to callmanager has g729r8. The TCS message from callmanager offers g729 and g729annexA. In CCM I have unchecked'The Wait For Far End H.245 Terminal Capability Set.<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal"> Is the problem that the Callmanager TCS isn’t g729r8 ?<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">In debug h245asn1 on CME I see the incoming TCS show g729 and g729annexA<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">Based on the debugs below it appears a Transcoder is required for an audio call to callmanager? I’m not clear why CME can’t figure out it’s G729/G711 only call from the dial-peer. Why is it invoking a XCODE in the debug voip ccapi inout?<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">I can’t upgrade at this time and am looking for technical reason why this is failing <span style="font-family:Wingdings">J</span><u></u><u></u></p><p class="MsoNormal">
<u></u> <u></u></p><p class="MsoNormal">Show dial-peer voice 40001 <<<<<<<<< this is the 9971 virtual dial-peer<u></u><u></u></p><p class="MsoNormal"> voice-class codec = 1<u></u><u></u></p>
<p class="MsoNormal"> codec = g729r8, payload size = 20 bytes,<u></u><u></u></p><p class="MsoNormal"> video codec = None<u></u><u></u></p><p class="MsoNormal"> voice class codec = 1<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">show run | begin voice service voip<u></u><u></u></p><p class="MsoNormal">voice service voip<u></u><u></u></p><p class="MsoNormal"> allow-connections h323 to h323<u></u><u></u></p>
<p class="MsoNormal"> allow-connections h323 to sip<u></u><u></u></p><p class="MsoNormal"> allow-connections sip to h323<u></u><u></u></p><p class="MsoNormal"> allow-connections sip to sip<u></u><u></u></p><p class="MsoNormal">
h323<u></u><u></u></p><p class="MsoNormal"> sip<u></u><u></u></p><p class="MsoNormal"> bind control source-interface Vlan88<u></u><u></u></p><p class="MsoNormal"> bind media source-interface Vlan88<u></u><u></u></p><p class="MsoNormal">
registrar server expires max 1200 min 300<u></u><u></u></p><p class="MsoNormal"> !<u></u><u></u></p><p class="MsoNormal">dial-peer voice 3000 voip<u></u><u></u></p><p class="MsoNormal"> destination-pattern 3...$ <<<<phone back at corporate<u></u><u></u></p>
<p class="MsoNormal"> session target ipv4:10.82.65.11<u></u><u></u></p><p class="MsoNormal"> codec g729<u></u><u></u></p><p class="MsoNormal"> dtmf-relay h245-alphanumeric<u></u><u></u></p><p class="MsoNormal"> no vad<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">show dialplan number 3001 | begin Successful Calls<u></u><u></u></p><p class="MsoNormal"> Successful Calls = 13, Failed Calls = 28, Incomplete Calls = 0<u></u><u></u></p>
<p class="MsoNormal"> Accepted Calls = 0, Refused Calls = 0,<u></u><u></u></p><p class="MsoNormal"> Last Disconnect Cause is "2F ",<u></u><u></u></p><p class="MsoNormal"> Last Disconnect Text is "no resource (47)", <<<<<<<<failed call<u></u><u></u></p>
<p class="MsoNormal"> Last Setup Time = 7461971.<u></u><u></u></p><p class="MsoNormal"> Last Disconnect Time = 7463158.<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">Show sccp <<<<<<< Transcoder on router registered to CME<u></u><u></u></p>
<p class="MsoNormal">TCP Link Status: CONNECTED, Profile Identifier: 2<u></u><u></u></p><p class="MsoNormal">Reported Max Streams: 6, Reported Max OOS Streams: 0<u></u><u></u></p><p class="MsoNormal">Supported Codec: g729r8, Maximum Packetization Period: 60<u></u><u></u></p>
<p class="MsoNormal">Supported Codec: g711ulaw, Maximum Packetization Period: 30<u></u><u></u></p><p class="MsoNormal">Supported Codec: g711alaw, Maximum Packetization Period: 30<u></u><u></u></p><p class="MsoNormal">Supported Codec: g729ar8, Maximum Packetization Period: 60<u></u><u></u></p>
<p class="MsoNormal">Supported Codec: g729abr8, Maximum Packetization Period: 60<u></u><u></u></p><p class="MsoNormal">Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30<u></u><u></u></p><p class="MsoNormal">
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30<u></u><u></u></p><p class="MsoNormal">Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30<u></u><u></u></p><p class="MsoNormal">
<u></u> <u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">Debug voip ccapi inout<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal"> Destination Interface=0x0, Destination Call Id=-1, Source Call Id=3012,<u></u><u></u></p>
<p class="MsoNormal"> Caps(Codec=0xC, Fax Rate=0x2, Vad=0x2,<u></u><u></u></p><p class="MsoNormal"> Modem=0x0, Codec Bytes=20, Signal Type=2)<u></u><u></u></p><p class="MsoNormal">.Apr 25 17:01:08.098: //3012/069E7F2A8303/CCAPI/cc_api_caps_ind:<u></u><u></u></p>
<p class="MsoNormal"> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),<u></u><u></u></p><p class="MsoNormal"> Playout Max=1000(ms), Fax Nom=300(ms))<u></u><u></u></p><p class="MsoNormal">.Apr 25 17:01:08.098: //3011/069E7F2A8303/CCAPI/cc_api_caps_ack:<u></u><u></u></p>
<p class="MsoNormal"> Destination Interface=0x0, Destination Call Id=-1, Source Call Id=3011,<u></u><u></u></p><p class="MsoNormal"> Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),<u></u><u></u></p>
<p class="MsoNormal"> Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=1)<u></u><u></u></p><p class="MsoNormal">.Apr 25 17:01:08.098: //3012/069E7F2A8303/CCAPI/cc_api_call_connected:<u></u><u></u></p><p class="MsoNormal">
Interface=0x4A0B5790, Data Bitmask=0x1, Progress Indication=NULL(0),<u></u><u></u></p><p class="MsoNormal"> Connection Handle=0<u></u><u></u></p><p class="MsoNormal">.Apr 25 17:01:08.098: //3012/069E7F2A8303/CCAPI/cc_api_call_connected:<u></u><u></u></p>
<p class="MsoNormal"> Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)<u></u><u></u></p><p class="MsoNormal">.Apr 25 17:01:08.098: //3011/069E7F2A8303/CCAPI/ccConferenceCreate:<u></u><u></u></p><p class="MsoNormal">
(confID=0x4C0E49BC, callID1=0xBC3, gcid=69FB77A-CBD211E3-8306F106-9323311A, tag=0x0)<u></u><u></u></p><p class="MsoNormal">.Apr 25 17:01:08.098: //3012/069E7F2A8303/CCAPI/ccConferenceCreate:<u></u><u></u></p><p class="MsoNormal">
(confID=0x4C0E49BC, callID2=0xBC4, gcid=69FB77A-CBD211E3-8306F106-9323311A, tag=0x0)<u></u><u></u></p><p class="MsoNormal">.Apr 25 17:01:08.098: //3011/069E7F2A8303/CCAPI/ccConferenceCreate:<u></u><u></u></p><p class="MsoNormal">
Conference Id=0x4C0E49BC, Call Id1=3011, Call Id2=3012, Tag=0x0<u></u><u></u></p><p class="MsoNormal">.Apr 25 17:01:08.102: //3011/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">.Apr 25 17:01:08.102: cc_api_get_xcode_stream : 4702<u></u><u></u></p><p class="MsoNormal">.Apr 25 17:01:08.102: //3011/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:<u></u><u></u></p><p class="MsoNormal">
<u></u> <u></u></p><p class="MsoNormal">.Apr 25 17:01:08.102: cc_api_get_xcode_stream : 4702<u></u><u></u></p><p class="MsoNormal">.Apr 25 17:01:08.102: //3011/069E7F2A8303/CCAPI/cc_api_bridge_done:<u></u><u></u></p><p class="MsoNormal">
Conference Id=0x36, Source Interface=0x4ACC2474, Source Call Id=3011,<u></u><u></u></p><p class="MsoNormal"> Destination Call Id=3012, Disposition=0x0, Tag=0x0<u></u><u></u></p><p class="MsoNormal">.Apr 25 17:01:08.102: //3012/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:<u></u><u></u></p>
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