Cisco On hold issue in sip trunk

[Eng Mohamed Saber]

Hello Dears,

I have an issue with my new CUBE setup, when a user makes an outbound call from a Cisco IP Phone and puts the call on hold all is going well and he can make an un-hold successfully 

But when the same Cisco IP phone receives an inbound call and tries to put it on hold, It gets stuck and can't un-hold again.

if he tries from the old voice gateway all is going well.

I checked the MRGL and all looks fine

The call flow is ( ITSP >> CUBE >>> CUCM >>> IP PHONE)

Also the transfer for inbound call has the same issue.

Is there any ideas 

[M02@rt37 · Accepted solution]

Hello @Eng Mohamed Saber 

When an IP phone attempts to un-hold or transfer an inbound call, the CUBE facilitates signaling exchanges between CUCM and the ITSP. Problems may arise if the CUBE mishandles SIP REFER or re-INVITE messages, which are often used in these scenarios. You should enable SIP debugging on the CUBE using debug ccsip messages to observe the signaling flow. Look for unexpected responses like 488 Not Acceptable Here or 500 Internal Server Error when attempting hold/un-hold or transfer operations. These errors could indicate a misconfiguration in how the CUBE processes mid-call signaling.

[Eng Mohamed Saber]

Hello,

Yes I already did this and all of a sudden after trying puting on hold the provider gave us "401 unauthorize "

the screenshot is below  

[Roger Kallberg]

As this is coming from your service provider have you asked them why they responds with this?

[Eng Mohamed Saber]

Hello Roger,

I think I got the issue , after checking the working outbound on hold flow and the inbound flow , I found that the reinvite for the working call has the sip authentication and the call has the issue not containing the sip authentication 

I checked the dial peers and found the sip tenant is exit only on the outbound dial peer 

But the incoming dial peer has no sip tenant 

So I will add the sip tenant to the incoming dial peer and hope it will work.

My final question, is the re invite use the same itsp sip inbound dial peer or it could convert to the outbound dial peer ???

I will make this change and update here if it solve the issue.

[M02@rt37]

@Eng Mohamed Saber 

A SIP re-INVITE typically uses the same dial peer that was matched during the initial call setup; for inbound calls, this would be the incoming dial peer. However, if there is a configuration mismatch—such as missing SIP tenant settings, codec mismatches, or signaling changes—the re-INVITE may fail or attempt to match a different dial peer. In your case, adding the sip tenant configuration to the incoming dial peer should resolve the issue, ensuring proper handling of the SIP signaling for hold/un-hold and transfers on inbound calls. Verify that both incoming and outgoing dial peers are consistently configured for codecs, SIP tenant, and transport protocols to avoid mismatches... This should allow re-INVITEs to function correctly without converting to the outbound dial peer.

[Eng Mohamed Saber]

Thanks a lot for your feedback, I will test it and update the case 

[Eng Mohamed Saber · Accepted solution]

Hello Dears,

The issue was solved after putting the Sip Tenant under the inbound Dial Peer.

Thanks alot