I'm having issues with inbound calls from a SIP ISP to a PABX connected endpoint failing due to DTMF RTP-NTE when DTMF dynamic payload type 97 is advertised in the INVITE. As such, i'm trying to configure asymmetric payload for DTMF interworking so that DTMF dynamic payload type 101 is used for the call leg from the CUBE to CUCM instead of 97 as received from the ISP with no luck. Dynamic PT 97 is still being sent to CUCM.
SIP ISP === [CUBE enterprise] ===== [CUCM] =====[MGCP GW] ==< QSIG E1>=== [PABX]-------- [phone]
Inbound SIP PSTN calls to with dynamic payload type 101 (a=rtpmap:101 telephone-event/8000) are successful
Inbound SIP PSTN calls to with dynamic payload type 97(a=rtpmap:97 telephone-event/8000) are successful
No media resources configured for transcoding on CUCM (by design)
No MTP in use (by design)
Transcoding configured on CUBE
SIP Trunk on CUCM to CUBE DTMF signalling method: no preference
voice service voip ip address trusted list <omitted> address-hiding mode border-element license capacity 100 allow-connections sip to sip fax protocol none sip bind control source-interface Loopback0 bind media source-interface Loopback0 asserted-id pai outbound-proxy <omitted> asymmetric payload full midcall-signaling passthru privacy-policy passthru sip-profiles 10
dspfarm profile 10 transcode codec g729r8 codec g729br8 codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 28 associate application CUBE ! dial-peer voice 1001 voip description ** Outgoing via CUCM ** huntstop preference 1 destination-pattern <omitted> session protocol sipv2 session target <omitted> voice-class codec 10 no voice-class sip outbound-proxy rtp payload-type nte 101 voice-class sip asymmetric payload dtmf dtmf-relay rtp-nte fax-relay ecm disable fax rate disable no vad
! dial-peer voice 2000 voip description ** Incoming from SIP PSTN ** translation-profile incoming PSTN-Inbound-Calls rtp payload-type nse 99 session protocol sipv2 incoming uri via 2000 voice-class codec 10 voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte digit-drop fax protocol none ! dial-peer voice 2001 voip description ** Outgoing via SIP PSTN ** translation-profile outgoing PSTN-Outbound-Calls session protocol sipv2 session target sip-server destination e164-pattern-map 2000 voice-class codec 10 offer-all voice-class sip profiles 2000 dtmf-relay rtp-nte digit-drop fax protocol none
... View more
Hi guys, I've set up Cisco Jabber for Windows to use Extend and Connect to control a remote PBX endpoint. I've configured the required CTI-RD device, remote destinations, associated the users to the line and added the devices to end-user controlled device. The extend and connect part is working flawlessly without any issues. I'm able to receive inbound calls on the remote PBX endpoint and control the call (hold, resume, transfer etc.) using the Jabber call window that pops up. However, I'm unable to make any outbound calls via the Jabber client when in extend and Connect mode. Reading the Extend and Connect guide, I need to configure Dial Via Office (DVO) Reverse. So when the user initiates a Dial-Via-Office reverse call, CUCM calls and connect to the Extend and Connect device (CTI-RD). CUCM then calls and connects to the number the user dialled and finally connects the two call legs. After attempting to configure DVO-R for Jabber for Windows in Extend and Connect mode following the CUCM feature services guide, i'm unable to get any outbound calls working. From RTMT, i am receiving the following Termination Cause Code: (27) Destination out of order. What i also notice is that there is no calling number for that trace either. I would've thought that the calling party would've been the Enterprise Feature Access (EFA) number. Has anyone got this working or can provide some guidance? Thanks.
... View more