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SIP Diversion Header

Velin Tsekov
Level 1
Level 1

Is it possible to exlude/disable SIP diversion header for an incomming call thru SIP trunk to UC560?

On CUCM 8.5 It's possible...

1 Accepted Solution

Accepted Solutions

Here is the CLI for disabling the diversion header:

voice class sip-profiles 1001

request INVITE sip-header Diversion REMOVE

For each of the inbound dial peers coming in from the SIP trunk provider, add voice-class sip profiles 1001.

Example:

dial-peer voice 3000 voip

description MainNumber

translation-profile incoming MainNumber_Called_1

session protocol sipv2

session target sip-server

incoming called-number 19725554000

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

voice-class sip profiles 1001

dtmf-relay rtp-nte

ip qos dscp cs5 media

ip qos dscp cs4 signaling

no vad

Note that all of the inbound dial peers have incoming called-number present and set to an external number, and they also have session target sipv2 and session target sip-server set.

View solution in original post

8 Replies 8

John Platts
Level 4
Level 4

Yes, it is actually possible to exclude or disable the SIP diversion header for an incoming call through the SIP trunk to the UC560. However, you will need to use CLI to configure this feature.

Cisco Unity Express requires the Diversion header to be present in a call that is forwarded to voicemail. If Cisco Unity Express detects a Diversion header, Cisco Unity Express allows the caller to leave a voicemail in the mailbox associated with the extension in the topmost Diversion header. If Cisco Unity Express does not detect any Diversion headers, it allows the caller to access his or her voicemail box. Cisco Unity Express will access the mailbox associated with the extension of the From header if such a mailbox exists, and if such a mailbox does not exist, will prompt for the extension.

OK, I will use the CLI...but what should i do? where should I disable the diversion header?

Here is the CLI for disabling the diversion header:

voice class sip-profiles 1001

request INVITE sip-header Diversion REMOVE

For each of the inbound dial peers coming in from the SIP trunk provider, add voice-class sip profiles 1001.

Example:

dial-peer voice 3000 voip

description MainNumber

translation-profile incoming MainNumber_Called_1

session protocol sipv2

session target sip-server

incoming called-number 19725554000

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

voice-class sip profiles 1001

dtmf-relay rtp-nte

ip qos dscp cs5 media

ip qos dscp cs4 signaling

no vad

Note that all of the inbound dial peers have incoming called-number present and set to an external number, and they also have session target sipv2 and session target sip-server set.

lgaughan
Level 1
Level 1

While completely possible as per John's instruction above, why do you want to remove the Diversion Headers?  What problem are they causing your inbound calls?

Laura

Dear Laura,

One customer of mine has SIP trunk configured in his deployment. He's using cisco UC560 as a call control element. His SIP provider provide them a capability to map their DID numbers (fixed number) to a mobile number. I mean if an internal phone X dials someone into the mobile network he's going to represent himself as a mobile number. When the mobile user want to call back he will dials the mobile number but the fixed number will ring. They received the call correctly, translate it correctly, the internal phone rings and the call is OK. But there is something

One customer of mine has SIP trunk configured in his deployment. The provider gives the customer a capability to map mobile numbers to fixed numbers. In example when internal phone X dials someone into the mobile network he is going to represent himself as a mobile number. When the mobile user want to call back he will dials the mobile number but the fixed number will ring. I received the call correctly, translate it correctly, the phone rings and the call is OK. But there is something irritatingli in the Callier ID:

In addition I am attaching you some debugs info.

Anyone?

It's all about an incomming diversion header, not outgoing.

Fixed with a couple of dial-peers.

Hi Velin,

I'm running into the same problem as you described, can you share what are "couple of dial-peers" that you created to solve this problem?

Thanks