04-22-2012 11:11 AM - edited 03-16-2019 10:45 AM
Hello guys!
I have a Cisco 2921 as a h323 voice gateway connected to a CUCM 8.5. When a PSTN call is placed and a directory number answers, if this call gets transefered the PSTN caller does not get the ringback tone but still get transfered. I already check the Service Parameters of Send User info to true and also on the voice gateway tried the progress_ind setup enable 3 on the outboind dial peer to CUCM with no success. What else I need to verify? Thanks in advanced!
Solved! Go to Solution.
04-23-2012 07:51 AM
OK, looks like you got Sip trunk as well, so best option here is to use the annuciator. As, in SIP you cannot send 180/183 for already established call. So make sure you have an annuniciator registered and included in the MRGL of h323 GW, and change the service paramater to have annunciator.
04-22-2012 12:03 PM
Hi,
On your H323 IOS gateway try adding
tone ringback alert-no-pi to the pots dial peer
E.g.
dial-peer voice 9 pots
desc *** OUTGOING CALLS TO PSTN ***
tone ringback alert-no-pi
destination-pattern 9T
port 0/0/0:15
!
Regards
Alex
04-22-2012 12:13 PM
Thanks for the reply. I configured as you adviced but still the PSTN is not getting ringback on transfers. Thanks again.
04-22-2012 04:52 PM
hello there,
Is this the call flow:-
PSTN----CIsco GW---h225----CCM----IP phone--transfer ---IP phone (in same cluster).
Some questions:-
++ Are you doing any external transfer?
++ I understand you cant hear the ringback, but did you hear MOH during the call transfer?
++ And you will have three options for user info in Service paramaeter, which one did you use ?
################################################################################################
Send H225 Uer info:- This parameter specifies which message Cisco CallManager sends for the ringback tone or tone on hold. Valid values follow: | |
This is a required field. | |
Default: User Info for Call Progress Tone ################################################################################################ |
++ If you choose the default, then you should see similar message during call transfer in the GW.
++ Can you place a test call and collect the following debugs:- (debug h225 asn1, debug h245 asn1, debug h225 q931 & debug voip ccapi inout with the calling details. )
H225.0 INCOMING PDU ::=
value H323_UserInformation ::=
{
h323-uu-pdu
{
h323-message-body information :
{
protocolIdentifier { 0 0 8 2250 0 5 }
callIdentifier
{
guid 'D66E5D8E60EE11E183C7E9D31167CDD6'H
}
}
h245Tunneling FALSE
}
}
++ You can try annuniciator option, but make sure the h323 GW MRGL has annuciator in it.
04-23-2012 07:47 AM
Thanks for the reply! Excdellent information and well detailed explanation. Let me narrow down the situation because it is more specific than I explained in the first post:
The situation is as follows: When a PSTN caller initiates the call and the Unity Connection auto-attendant answers, the call gets connected as it should be, but when the users press the desired extension, the Unity announces to the user "Please hols while I trnasfer your call" as it should be, then for a brief moment the PSTN caller hears the MoH while the Unity dials the desired extension, then when the PSTN call gets transfered to the desired extension and the desired extension starts to ring, the PSTN caller does not hear ringback until either the extension gets answer or the voicemail takes over.
I tested routing the iplot number directly to an extension and when the PSTN caller gets transfered, it gets MoH until the other person nsweres or the Voicemail takes over. So the issue is only when the Unity Connection auto-attendant answers and initiates the transfer of the PSTN incoming call to an internal directory number.
The call f;lows is as follows:
PSTN (SIP trunk) ----> Cisco Voice GW (h225) ----> CCM ---> Unityt Connection AA ---> IP Phone--transfer ---> IP phone
Im using the default User Info for Call Progress Tone on the service parameters in the CUCM.
Thanks again!
04-23-2012 07:51 AM
OK, looks like you got Sip trunk as well, so best option here is to use the annuciator. As, in SIP you cannot send 180/183 for already established call. So make sure you have an annuniciator registered and included in the MRGL of h323 GW, and change the service paramater to have annunciator.
04-23-2012 08:02 AM
Excellent, it is working beautifully. Thanks again for your help! Great explanations.
08-23-2017 11:43 AM
You're rigth! ANN in MRG of SIP Trunk!
11-09-2018 02:40 AM
Hello @marmugam et all. Please I am currently expeirencing same issue, same scenario except that I have UCCX in place of CUC. How do I confirm that annuniciator is registered and included in the MRGL of h323 GW?? Please help, I am relatively new to cisco gateways
11-15-2018 09:00 PM
I followed these steps and this did not fix my issue. I still have no ringback.
Any thought on what else could be tried?
Thanks in advance!
04-08-2019 12:08 PM
05-16-2019 01:17 AM
Hi
And how did you resolve that?
I have a SIP trunk to the Provider. The mrgl is selected and media termination point required is selected.All media resources are registered. But I have still no ringback while the transfer.
Any hint, where I can have a look?
Thanks a lot.
05-16-2019 01:49 AM
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