Hi Cliff and Tech,
Thanks for your reply.
I am not sure if the xcode works properly and I made some show to check it.
The xcodes have been registered to ccm and cme successfully.
On cme site:
1. "show call active voice compact"-- It shows the g729 to ccm ip phone address
and g711 to cue ip address.
2. "show dspfarm session"-- It shows the g729 to cme loopback 0 and g711 to cme
loopback 0.
3: "show voip rtp connections"-- It shows 2 active RTP connections, one from
cme loopback 0 to ccm ip phone, the other from cme loopback 0 to cue.
On ccm site:
1. There is no xcode involved in the process because IP phone can choose g729
to cme.
2. If I tick the outbound fast connection and MTP, it shows xcode works from
the "performance monitor" interface when I made a call.
I guess it may be an IPIPGW h323 to sip promble.
When I use HDV in cme for xcode, it is a first generation dsp so it can only
xcode from g729 to g711,but it can not translate from h323 dtmf to sip dtmf.
The second generation dsp has an enhanced MTP funtion which can do xcode and
translate dtmf together.
Is that right? Someone can help me with this?
Thanks.
Jeffrey
________________________________
From: Cliff McGlamry <cl...@mcglamry.net>
To: ccie_voice@onlinestudylist.com
Sent: Sunday, April 5, 2009 12:49:10 AM
Subject: Re: [OSL | CCIE_Voice] ccm to cme voicemail
Dead air on a connected call means that the media
stream is not in place. This can be for several reasons:
1. Transcoder at HQ site (if you're using the
6608 blade) is not registered, not in the correct MRG, MRGL, or location(this
is
a hidden killer), or using the WRONG DEFAULT GATEWAY.
2. Transcoder at CME site not
registered. In addition to configuration of dspfarm profile, three
sdspfarm commands are required in telephony-service. Make sure the g729r8
codec has been set up for transcoding. Use the show sccp to confirm the
transcode resource is up, registered and operational.
3. SCCP is attempting to register to the
incorrect address. The sccp ccm command should point at the same ip
address being used as the source address in the telephony service section of
your config.
4. Incoming dial-peer not explicitly defined
correctly causing call to use default dial peer. Default dial peer cannot
invoke transcoder. Use the show voice call status to see which dial peers
are operational. This issue is the primary cause of this type problem,
although anything in this list can cause it.
5. MRGL not correctly assigned to the GK
Trunk.
Cliff
----- Original Message -----
From: jeffrey liujian
To: Cliff McGlamry ; Voice CCIE ; gughan...@yahoo.in ; rxlm...@gmail.com
Sent: Saturday, April 04, 2009 8:44 AM
Subject: Re: [OSL | CCIE_Voice] ccm to cme voicemail
Hi all,
There is some trouble.
ccm --> gk --> cme using g729, cme --> cue using g711.
when ccm phone makes a call to cme phone or cue number:
1. call can be connected, but no sound at all.
2. if I change the code (between ccm and cme) to g711, everything is fine.
Why does this happen?
Thanks.
Jeffrey
________________________________
From: Cliff McGlamry <cl...@mcglamry.net>
To: Gughan Gug <gughan...@yahoo.in>
Cc: ccie_voice@onlinestudylist.com
Sent: Saturday, March 21, 2009 5:19:05 PM
Subject: Re: [OSL | CCIE_Voice] ccm to cme voicemail
Oh, this one bites lots of folks.
In the telephony-service, make sure you have: call-forward pattern .T
If you're missing that, it won't work.
Cliff
----- Original Message -----
From: Gughan Gug
To: ccie_voice@onlinestudylist.com
Sent: Saturday, March 21, 2009 1:46 AM
Subject: [OSL | CCIE_Voice] ccm to cme voicemail
Hi,
CCM phones call to CME phones through gatekeeper . phone from ccm site can
call to cme voicemail but if the phone call to cme ip phone and get redirected
the calls get disconnected.
I have allowed the h323 to sip and sip to h323 coneectiion under voice service
voip,transcoder at cme
when a cme phone call the other one and at noan the calls get redirected
properly to vociemail.
Any help in this regard. anything i am missing here. since ccm phoes can call
directly to cme voice
mail using the voicemail number no issue, only when it gets redirectd the
call get disconnected.
Regards
Gug
________________________________
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