Outbound Ringback Issues over SIP trunk using CUCM 8.6 and H323 Gateway

[JZegers06]

Im using a SIP provider (voip.ms) to a H323 gateway (2801) and from there go to my CUCM 8.6 server.  Inbound calls work fine, i get ringback and outbound calls connect and work fine, i just hear no ringback on my outbound calls via my SIP provider.

Attached are my debug ccsip messages, debug ccsip calls and debug ccapi inout as well as my running config

Here is also a link to another thread for other stuff ive tried:

http://www.techexams.net/forums/ccnp-voice/78494-outbound-ringback-issues.html

If anyone could solve this, you would be my hero!!!!                  

[Ayodeji Okanlawon]

  Hi John,

By default, Cisco Unified Communications Manager  will signal the calling phone to play local ringback if SDP is not  received in the 180 or 183 response. If SDP is included in the 180 or  183 response, instead of playing ringback locally, Cisco Unified  Communications Manager will connect media, and the calling phone will  play whatever the called device is sending (such as ringback or busy  signal). If you do not receive ringback, the device to which you are  connecting may be including SDP in the 180 response, but it is not  sending any media before the 200OK response. In this case

The log you sent does not contain the debug ccsip messages, so I am unable to determine if you receive 183 with SDP. If your provider is sending 183 without SDP, then cucm should be teliing the phone to play ringback locally. However CUCM uses Annunciator to play ringback for sip trunks. So you need to ensure that you have ANNunciator in your MRG and assign this MRG to a MRGL and inturn assign the MRGL to your phones and cube gateway

If you are receiving 183 with SDP but still not getting ringback, then try you can use a feature called disable early media to force cucm to play ringback locally..(dont forget to enable Annunciator and assign to phones and cube gateway)

disable-early-media 180

To specify which call treatment, early media or local ringback, is provided for 180 responses with 180

responses with Session Description Protocol (SDP), use the disable-early-media 180 command in

sip-ua configuration mode. To enable early media cut-through for 180 messages with SDP, use the no

form of this command. disable-early-media 180

config t

sip-ua

disable-early-media 180

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[JZegers06]

I added the disable-early-media 180 with no luck.  attached is a copy of debug ccsip messages.  What do i need to do to enable Annunciator and assign it to phones/gateway?  Ive never done that before.

[JZegers06]

OK i have ANN_2 and ANN_3 under Announciator, i put them in a group then put them in a list.  On my gateway i applied the list and checked MTPR checkbox.  I selected the Media Resource Group list under my phones and device pool.  I also went to Service Parameters and set Send H225 User Info Message to use ANN for ringback and still ZERO luck.

Any more ideas or debugs you guys want?

[Ayodeji Okanlawon]

You dont need to check MTP box. Please uncheck otherwise you may create additional problems..

Can you do a test call and send me a

show voip rtp connection  (this will show us if the annunciator is been called upon to provide ringback)

NB: to see the annuciator in the voip rtp connection command to need to be refreshing the command. Dont just call and type the command once. Because the ann is called when its time to provide ringback.

also please send a full debug ccsip messages. The one you sent, the call did not complete

Do oyu know how to collect cucm traces? If you can send that will be great.

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[Ayodeji Okanlawon]

John,

Have you made any progress with this? Did you see my last post

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[JZegers06]

Havent had a chance to run through this yet.  Will tonight and post results that you requested.  Im not too familiar with CUCM traces

[shinajcm]

Hi All,

Even i have the same issue . There will be one ring then then a fax tone then it will give a message" We are sorry your call cannot be completed this time, please hang up and dial after sometime" this message will play twice in quick session of two times and then the call will disconnect. The call is reaching the gateway .

Working calls are receiving 183 message with SDP but the failing calls from this location does not have 183 instead it have 180 without any SDP message in it  and getting a 200 OK message soon.

Is it ok for us to involve ITSP.

Regards,

Shinaj cm 

[Ayodeji Okanlawon]

You should engage your ITSP and I hope it is not TATA. Provide traces to them and push them to look at why you are seeing this behaviour

[shinajcm]

Hi Ayodeji,

Customer is asking why it is service provider issue as all the numbers are going to the same provider and it is working.

this is occurring only for a set or block of DID.

Regards,

Shinaj cm

[JZegers06]

Here is the requested output.  Anytime you see the output of show voip rtp connection it is when the outbound call should be providing ringback, and the number i am calling is ringing on the other end.

In the mean time i have also tried this:

System > Service Parameters > Under Clusterwide Parameters (Device - H323) set Check Progress Indicator Before Establishing Media to True (False as default) and setting Send H323 User Info Message to H225 Info for Call Progress Tone, restarted callmanager service, reset the gateway, all with no luck.

[Ayodeji Okanlawon]

John,

Looking at your trace I dont see the communication with CUCM. I dont see your gateway receiving an invite from cucm or sending sip trying or 200 ok to cucm. If the call originated from cucm then this must happen.

Did you omit this traces?

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[JZegers06]

No those were from the router and nothing was omitted.  Were those suppose to come from CUCM traces?

[Ayodeji Okanlawon]

John,

I know why I dont see the trace, its beacise the leg to cucm is h323 leg, its not sip.

May I ask, why are you not using SIP all the way.? i.e. configure your sip trunk from cucm to your gateway. Have you tried this.

Let me know if you want to try with the sip trunk setup. Otherwise please do another test call and send

debug voip ccapi inout.

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[JZegers06]

I would try converting it to SIP the whole way.  How do i go about configuring that in CUCM and on the router>?