[cisco-voip] CME 8.6 and 9971 out H323 Trunk to CUCM 9.1 no audio
Sreekanth Narayanan
sknth.n at gmail.com
Sat Apr 26 14:26:05 EDT 2014
What's this dial-peer below?
dial-peer voice 3000 voip
destination-pattern 3...$ <<<<phone back at corporate
session target ipv4:10.82.65.11
codec g729 *---> is this r8 or some other flavor?*
dtmf-relay h245-alphanumeric
no vad
I'd try 2 things:
1. create
voice class codec 1
codec preference 1 g729r8
codec preference 2 g729br8
codec preference 3 g711ulaw
and then assign it on the dial-peer as 'voice-class codec 1'. This way i'm
offering multiple codecs for negotiation.
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.
Further, there are always debugs such as what Amit pointed out to.
Thanks
Sreekanth
On 25 April 2014 23:04, Jason Aarons (AM) <jason.aarons at dimensiondata.com>wrote:
> 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.
>
>
>
> Is the problem that the Callmanager TCS isn’t g729r8 ?
>
>
>
> In debug h245asn1 on CME I see the incoming TCS show g729 and g729annexA
>
>
>
> 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?
>
>
>
> I can’t upgrade at this time and am looking for technical reason why this
> is failing J
>
>
>
> Show dial-peer voice 40001 <<<<<<<<< this is the 9971 virtual dial-peer
>
> voice-class codec = 1
>
> codec = g729r8, payload size = 20 bytes,
>
> video codec = None
>
> voice class codec = 1
>
>
>
> show run | begin voice service voip
>
> voice service voip
>
> allow-connections h323 to h323
>
> allow-connections h323 to sip
>
> allow-connections sip to h323
>
> allow-connections sip to sip
>
> h323
>
> sip
>
> bind control source-interface Vlan88
>
> bind media source-interface Vlan88
>
> registrar server expires max 1200 min 300
>
> !
>
> dial-peer voice 3000 voip
>
> destination-pattern 3...$ <<<<phone back at corporate
>
> session target ipv4:10.82.65.11
>
> codec g729
>
> dtmf-relay h245-alphanumeric
>
> no vad
>
>
>
> show dialplan number 3001 | begin Successful Calls
>
> Successful Calls = 13, Failed Calls = 28, Incomplete Calls = 0
>
> Accepted Calls = 0, Refused Calls = 0,
>
> Last Disconnect Cause is "2F ",
>
> Last Disconnect Text is "no resource (47)", <<<<<<<<failed call
>
> Last Setup Time = 7461971.
>
> Last Disconnect Time = 7463158.
>
>
>
> Show sccp <<<<<<< Transcoder on router registered to CME
>
> TCP Link Status: CONNECTED, Profile Identifier: 2
>
> Reported Max Streams: 6, Reported Max OOS Streams: 0
>
> Supported Codec: g729r8, Maximum Packetization Period: 60
>
> Supported Codec: g711ulaw, Maximum Packetization Period: 30
>
> Supported Codec: g711alaw, Maximum Packetization Period: 30
>
> Supported Codec: g729ar8, Maximum Packetization Period: 60
>
> Supported Codec: g729abr8, Maximum Packetization Period: 60
>
> Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
>
> Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
>
> Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization
> Period: 30
>
>
>
>
>
> Debug voip ccapi inout
>
>
>
> Destination Interface=0x0, Destination Call Id=-1, Source Call Id=3012,
>
> Caps(Codec=0xC, Fax Rate=0x2, Vad=0x2,
>
> Modem=0x0, Codec Bytes=20, Signal Type=2)
>
> .Apr 25 17:01:08.098: //3012/069E7F2A8303/CCAPI/cc_api_caps_ind:
>
> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
>
> Playout Max=1000(ms), Fax Nom=300(ms))
>
> .Apr 25 17:01:08.098: //3011/069E7F2A8303/CCAPI/cc_api_caps_ack:
>
> Destination Interface=0x0, Destination Call Id=-1, Source Call Id=3011,
>
> Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
>
> Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=1)
>
> .Apr 25 17:01:08.098: //3012/069E7F2A8303/CCAPI/cc_api_call_connected:
>
> Interface=0x4A0B5790, Data Bitmask=0x1, Progress Indication=NULL(0),
>
> Connection Handle=0
>
> .Apr 25 17:01:08.098: //3012/069E7F2A8303/CCAPI/cc_api_call_connected:
>
> Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)
>
> .Apr 25 17:01:08.098: //3011/069E7F2A8303/CCAPI/ccConferenceCreate:
>
> (confID=0x4C0E49BC, callID1=0xBC3,
> gcid=69FB77A-CBD211E3-8306F106-9323311A, tag=0x0)
>
> .Apr 25 17:01:08.098: //3012/069E7F2A8303/CCAPI/ccConferenceCreate:
>
> (confID=0x4C0E49BC, callID2=0xBC4,
> gcid=69FB77A-CBD211E3-8306F106-9323311A, tag=0x0)
>
> .Apr 25 17:01:08.098: //3011/069E7F2A8303/CCAPI/ccConferenceCreate:
>
> Conference Id=0x4C0E49BC, Call Id1=3011, Call Id2=3012, Tag=0x0
>
> .Apr 25 17:01:08.102: //3011/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
>
>
>
> .Apr 25 17:01:08.102: cc_api_get_xcode_stream : 4702
>
> .Apr 25 17:01:08.102: //3011/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
>
>
>
> .Apr 25 17:01:08.102: cc_api_get_xcode_stream : 4702
>
> .Apr 25 17:01:08.102: //3011/069E7F2A8303/CCAPI/cc_api_bridge_done:
>
> Conference Id=0x36, Source Interface=0x4ACC2474, Source Call Id=3011,
>
> Destination Call Id=3012, Disposition=0x0, Tag=0x0
>
> .Apr 25 17:01:08.102: //3012/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20140426/42b4fd89/attachment.html>
More information about the cisco-voip
mailing list