[cisco-voip] Ringback problem when calling to PSTN

[Matthew Berry]

Oh, sure. I see what you're saying then.  I was reading the thread wrong.

Thanks!

Matthew Berry, CCIE #26721 (Voice)
Sr. Unified Communications Engineer, CDW
+1.763.592.5987  |  protocol.by/matthewberry

On Dec 16, 2011, at 7:44 AM, Bill Riley wrote:

It does not unless you have the option enabled he specified.

Try changing service parameter -> callmanager -> Send H225 User Info
Message -> use ANN for ringback


From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at puck.nether.net> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Matthew Berry
Sent: Friday, December 16, 2011 7:35 AM
To: Roger Wiklund
Cc: cisco-voip
Subject: Re: [cisco-voip] Ringback problem when calling to PSTN

I don't believe the annunciator is what plays ringback.  That should be a function of the PSTN gateway.

Have you tried a SIP trunk between the gateway and CUCM?  It'd eliminate the interop between H.323 and SIP.

Thanks!

Matthew Berry, CCIE #26721 (Voice)
Sr. Unified Communications Engineer, CDW
+1.763.592.5987  |  protocol.by/matthewberry

On Dec 16, 2011, at 6:37 AM, Roger Wiklund wrote:


Ensure you have a MRGL with ANN on your H323 GW and phones.

Try changing service parameter -> callmanager -> Send H225 User Info
Message -> use ANN for ringback

/Roger

On Fri, Dec 16, 2011 at 1:22 PM, Robert Hass <robhass at gmail.com<mailto:robhass at gmail.com>> wrote:

Hi

I have problem with ringback tone when doing calls from IP Phone
connected to CUCM to PSTN.
CUCM is configured to use our gateway (Cisco 2811) via H.323 and our
gateway is using SIP Trunk
do our carrier.

We don't hear ringback tone when calling to PSTN. But connectivity is
working fine.
We hear ringback tones when taking call from PSTN to CUCM.

Our gateway configuration:

voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 fax protocol pass-through g711alaw
 h323
 modem passthrough nse codec g711alaw
 sip
!
voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
!
voice class h323 1
 h225 timeout tcp establish 3
!
!
dial-peer voice 1 voip
 translation-profile incoming 1
 incoming called-number .
 codec g711alaw
 no vad
!
dial-peer voice 2 voip
 description SIP Trunk to carrier (CUCM->PSTN)
 translation-profile outgoing 2
 huntstop
 destination-pattern 0T
 progress_ind setup enable 3
 session protocol sipv2
 session target ipv4:x.x.x.x:5060
 dtmf-relay rtp-nte
 codec g711alaw
 ip qos dscp cs5 media
 no vad
!
dial-peer voice 3 voip
 description PSTN->CUCM
 huntstop
 destination-pattern 49xxxxxx...$
 progress_ind setup enable 3
 session target ipv4:192.168.36.2
 voice-class h323 1
 codec g711alaw
 ip qos dscp cs5 media
 no vad

Cisco IOS Software, 2800 Software (C2800NM-ADVIPSERVICESK9-M), Version
15.0(1)M4, RELEASE SOFTWARE (fc1)
CUCM is 8.6

Any hints what is bad configured ?

Thanks
Robert
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