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New Member

CUCM 10.5 problem with call transfe

Good morning

 

I have a problem with call transfer with consultation from an intern to another.

SIP phones used 7841

 

This schenario

Branch A

CUCM 10.5 at DATACENTER

RTR with gre-over ipsec TUNNELS to the DC

Pstn Gateway with 3 ISDN-BRI

 

Branch B

RTR-gre-tunneled over ipsec to DC

trace attachement

 

Problem

Receiving a call from the PSTN in Branch A, I try to transfer it to my colleague in the same office and after having talked with him, when i try to transfered the call, both ipphone remain blocked.

 

By tracer known that the error received is

MSG_LABEL: 488 Not Acceptable Media

 

This doesn't happen if from Branch A transfer to  Branch B

 

Can someone explain me why the phone of the Branch A not accept the media streams?

 

Thanks a lot

8 REPLIES

Hi Lorenzo.Just to make sure,

Hi Lorenzo.

Just to make sure, if you make a direct call transfer, does it work?

Can please send a full trace from when pstn  call arrives to Phone A?

 

Please let me know

 

 

Regards

 

Carlo

Please rate all helpful posts "The more you help the more you learn"
New Member

Hi Carlo,with the direct

Hi Carlo,

with the direct transfer there are no problems even if I use a SCCP phone I don't find some problem.

Attachment trace complete

 

Best Regards

 

Lorenzo

Hi Lorenzo.Thanks for sending

Hi Lorenzo.

Thanks for sending me the info requested.

Which signaling protocol are you using between CUCM and VG?

If SIP, please activate a debug ccsip message during the call transfer and post it here.

Thanks

Regards

 

Carlo

Please rate all helpful posts "The more you help the more you learn"
New Member

Hi Carlothe VG uses H323

Hi Carlo

the VG uses H323 protocol to communication with the CUCM

Regards

 

Lorenzo

Hi Lorenzo.On your VG, add a

Hi Lorenzo.

On your VG, add a voice class codec with g711U/Alaw and g722-64 codec and add it to dialpeer pointing to CUCM.

 

Please try it and let me know.

 

 

HTH

 

Regards

 

Carlo

Please rate all helpful posts "The more you help the more you learn"
New Member

on the dial peer to the CUCM

Hi Carlo on the dial peer to the CUCM is already present the codec. I opened a tac and an engineer cisco asked me the sniffed traffic behind the phone. I will update the post as soon as I will get a response. Let me know if you have another idea to solve this problemi. Thank you Lorenzo

Hi Lorenzo.Can you please

Hi Lorenzo.

Can you please post full vg config in order to replicate it on my lab?

 

Thanks

 

Regards

Carlo

Please rate all helpful posts "The more you help the more you learn"
New Member

Hi Carlo,I open a case with

Hi Carlo,

I open a case with Cisco have not been able to re solve the problem, requiring every time packet capture behind the phones.
when I discovered that the problem came from the SIP inspection that was the firewall in the data center I decided to update the firewall multicontext with the latest version, this has solved the problem. I do not have enough data to declare that the version of ASA is affected by bugs, but it is very likely

thanks for your time

Lorenzo

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