Transfer call from CUE failed

[Anju Josua]

Hi all,

i'm having problem with transfering call from CUE to phone registered in CME

When i call from other branch to unity express, the call is transfered but as soon as it picked up the call dropped.

i'm assuming this is related to codec problem because when i use g711ulaw as default codec, this problem doesn't happen.

i already configure transcoder on CME, and already verified that transcoder is working.

i also attached my sh run configuration

any help will be appreciated

Thanks

Regards,

Anju Josua

[Roger Kallberg]

Not sure if I understand the described call flow fully, but CUE only support G711ulaw and ASFAIK it's not supported to use a voice-class codec under a voice register pool. The command is accepted, but will not work as you would think. For SIP CME phones the recommended praxis is to hard code the codec under each voice register pool.

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[Anju Josua]

HI roger kallberg

thanks for your reply.

so i must specify the codec betwaeen g729 or g711 under voice register pool?

if i specified it as g711ulaw, can call from the branch using g729 transfered to sip ip phone?

or i should use g729 instead?

Regards,

Anju Josua

[Roger Kallberg]

In either case you will need a transcoder to handle the G.729 to G.711 calls.

May I ask if there's any special reason for why you opted to use SIP CME? I've always thought that Cisco did a sort of piss poor work with that part of CME. If possible I would recommend you to use SCCP CME instead. Then you wouldn't have to set the codec on the phones, but you still would need to have transcoding resources for the branch office calls that forwards to CUE.

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Sent from Cisco Technical Support iPhone App

[Anju Josua]

Hi again,

It's other guys that set up CME, he said this cisco phone type doesn't support sccp.

i haven't try to hardcode the codec, i'll give the result soon.

thanks

Regards,

Anju Josua

[Roger Kallberg]

What kind of phones do you use?

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[Roger Kallberg]

Forget my last question, I saw in your sh run that it's 3905. Those phones unfortunately only support SIP.

As I checked your config again I saw that you have used the codec g711ulaw on all your voip dial peers. That would be the reason for why you don’t get a request for a transcoder for your calls from the branch office that are forwarded to CUE. Please change that to use g729, I took the liberty to do the needed changes to your dial peers.

dial-peer voice 1000 voip

session protocol sipv2

incoming called-number .

dtmf-relay sip-notify

codec g729r8

no vad

!

dial-peer voice 1800 voip

corlist outgoing pbx

destination-pattern 91..

session protocol sipv2

session target ipv4:192.168.255.16

dtmf-relay sip-notify

codec g729r8

no vad

!

dial-peer voice 301 voip

description *Connection to PBX*

translation-profile outgoing OUTGOING-to-PBXviaVOIP

destination-pattern 9

session protocol sipv2

session target ipv4:192.168.255.16

dtmf-relay rtp-nte sip-notify

codec g729r8

no vad

Note: I haven't changed your dial peer that is used for CUE. This is by purpose since that need to be g711ulaw. The difference in codecs used will trigger the use of a transcoder resource when needed.

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[Anju Josua]

Hi roger

i've tried your suggestion but still failed on sip phone.

i try to set up a shoftphone on cme, try to call and it's works.

i think somtehing miss for sip configuration

any ideas?

regards,

Anju Josua

[Bruno Rangel]

Try this:

voice service voip

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

If does not works, send the output of debugs (calling/called number)

debug ccsip messages

debug ccsip all

Cheers
Bruno Rangel

"Se você quiser alguém em quem confiar, confie em si mesmo. Quem acredita sempre alcança"
Renato Russo

[Anju Josua]

Hi bruno,

i'vet ried that 2 command, but still not working

i attach my debug

Regards,

Anju

[Bruno Rangel]

I can see Disconnect Cause SIP: 500, but not found any reason for that ... I think a TAC Case will help.

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Cheers
Bruno Rangel