CUCM 10.5.2 - Problem with transfer in specific site

[wilsonsant]

Hi Guys,

My Customer have the solution with CUCM 10.5.2 and is verifying the follow problem: in specific site the User A that is the "PSTN" call for the office and User B pickup the Call. When the  B try transfer call to User C and User C pickup the call, this stay in wait music. Transfer internal worked fine.

Any have idea about this?

Thanks,

Wilson

[Ratheesh Kumar · Accepted solution]

Hi there

Is the gateway configured as SIP/H323/MGCP. Is it happening to all the phones at that site ? Any hardware MTPs registered. If you could share the debugs ccsip messages or all and CCM traces we could start troubleshooting.

Alternatively, if you could enable the MTP required on the SIP trunk (if SIP) and test the transfer.

Lets share how it goes.

Hope this helps !!!

Cheers
Rath!

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[wilsonsant]

Hi Cisco Rath!

Thanks a lot for Your attentation and Your help. Answering Your questions

- This site is working SIP protocol

- The problem is occurring with all IP Phones (30 IP Phones)

- The MTP is registered

BRSPCUR-GW-VOZ-01#
BRSPCUR-GW-VOZ-01#sh sccp
SCCP Admin State: UP
Gateway Local Interface: GigabitEthernet0/0
        IPv4 Address: 10.66.22.1
        Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 10.66.184.1, Port Number: 2000
                Priority: N/A, Version: 7.0, Identifier: 2
                Trustpoint: N/A
Call Manager: 10.66.184.2, Port Number: 2000
                Priority: N/A, Version: 7.0, Identifier: 1
                Trustpoint: N/A

Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 10.66.184.2, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 2
Reported Max Streams: 6, Reported Max OOS Streams: 0
Supported Codec: g729br8, Maximum Packetization Period: 60
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

Conferencing Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 10.66.184.2, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1
Reported Max Streams: 24, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: g729br8, 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
TLS : ENABLED

MTP Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 10.66.184.2, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 3
Reported Max Streams: 100, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
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
TLS : ENABLED

BRSPCUR-GW-VOZ-01#

- About debug ccsip I will verify with my co-worker can available, because, my profile not allow run debug

- About CUCM traces, what traces that You is mentioning?

- I enable the MTP in SIP Trunk, now I will requested the User to do test then I answer for You if worked or no

Thanks,

Regards,

Wilson

[CoDeC]

hi,

have you checked the "rerouting calling search space" of the SIP Trunk?

Regards,

[wilsonsant]

Hi Alberto Sanz,

Thanks a lot for Your contact. This option is none

Thanks,

Regards,

Wilson

[CoDeC]

Hi,
Try to configure with a CSS with the phone's partition.

[wilsonsant]

Hi Cisco Rath!

After enable Media Termination Point on SIP Trunk conference and transfer worked. I have the follow question: are there any impact keep this parameters enable?

Thanks,

Wilson

[Ratheesh Kumar]

Hi there

In your case, It seems to be an failure with MTP /transcoder allocation during the transfer. Typically you dont need to turn MTP on you trunk unless a DTMF mismatch, EO, supplementary features etc.

Alternatively, you can go to the SIP trunk's SIP profile and set the "Early Offer support for voice and video calls" to Mandatory (insert MTP if needed) and uncheck the MTP Required from the SIP Trunk.

With this what basically happens is the CUCM inserts a MTP if the media characteristics cant be determined.

Hope this helps!

Cheers
Rath!

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