Outside user call Call Center on for example 345567. After welcome ton he has possibility to chose 1, 2, 3, 4 and 5 option. How can I forward DTMF tone (as above) from PSTN over ISDN PRI (IOS MGCP Gateway) to CUCM then to SIP trunk to third party Call Canter?
Sip Trunk use "No Preference".A GW has Type of DTMF Relay "Current GW Config", and MGCP has dtmf-relay voip codec all mode out-of-band
Here are the DTMF options that generally work the best for each protocol:
mgcp dtmf voip codec all mode out-of-band
rtp-nte (RFC 2833)
For MGCP-SIP you have a few options, in order of preference:
1. out-of-band (NTFY) to OOB SIP (notify, kpml)
This is probably your easiest option providing that your 3rd party SIP gateway supports SIP-NOTIFY or KPML. The chances of this are probably less than 5%.
2. MGCP out-of-band to RFC 2833 on SIP side.
This will require an MTP on the SIP trunk to add the in-band DTMF into the stream. Chances are your SIP GW supports 2833.
3. "mgcp dtmf codec all mode nte-gw". This will produce RFC 2833 packets and it will go all the way through to the gateway. The reason this is a lower preference is because MGCP has an annoying habit of sending NTFY messages even when the method is set to NTE and duplicate digits will still be created under certain circumstances. To be honest, this is usually a mis configuration but I've never set it up in the lab to figure out the correct configuration.
This is all assuming your gateway supports either RFC 2833 or KPML/NOTIFY. If it only supports in-band (actual tones) it requires a CUBE with C5510 DSPs acting as transcoders.
My Gw is C2801 with VWIC2-1MFT-T1/E1 (PRI ISDN) configured as MGCP GW connected to CUCM 220.127.116.110-16.MGCP is configured as "mgcp dtmf-relay voip codec all mode out-of-band". Also CUCM has SIP trunk configured as "DTMF Signaling Method:RFC 2833".From Ip phone user can use DTMF ton to chose option. From outside user got welcome message but when press option the welcome message still plays.
Also on the SIP trunk is checked "Media Termination Point Required".
Did you ever get this resolved or anyone have any input? I have the identical problem with MGCP configured PSTN connection and a SIP Connection going internally directly from Call Manager 6.1 to an IVR system. DTMF works internally but call go out to the PSTN if I press a DTMF digit the RTP stops. I have not had a chance to change the MGCP yet as I am waiting until after hours to do so but thought I would ask if there was an update....
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