You can use python. I used python for something similar to deploy macros and in-room control ... View more

What do you mean by having a SIP trunk between two sites? Do you have two CUCM clsuter connected by SIP ICT? or do you have two CME gateways connected by SIP trunk? ... View more

Thank you Omar for this. This is exactly my gut feeling as well. Its too cumbersome and the task of maintaining it can be demanding ... View more

You might want to check to see if there is a firewall blocking sip registration on port 5060 using TCP ... View more

Hi, We dont see the outbound dial-peer as Mohammed mentioned. But here are a couple of things you can do.. 1. Apply the correct sip bind statements to your dial-peers eg   ++ CUCM facing dial-peer ++ dial-peer voice 2000 v voice-class sip bind control source-interface interface Loopback0  voice-class sip bind media source-interface interface Loopback0   ++ ITSP dial-peer ++ dial-peer voice yyyy voip voice-class sip bind control source-interface interface GigabitEthernet0/0/1 [assuming gig0/0/1 = interface that connects to ITSP] voice-class sip bind media source-interface interface GigabitEthernet0/0/1 [assuming gig0/0/1  = interface that connects to ITSP]   Second thing is turn off TCP as the transport protocol on the dial-peer facing your ITSP. Try with default UDP first because some ITSP wont allow TCP unless of course you have been told to use TCP   ... View more

Hi, You should be able to use the CME gateway as h323 gateway in cucm. You need to ensure that you configure the correct h323 bind statements to the correct interface.. eg interface loop x h323-gateway voip interface h323-gateway voip bind srcaddr x.x.x.x ( where x.x.x.x is the ip of the interface and its the ip you will use on CUCM)     You should then configure an inbound dial-peer that will be used to route calls from CUCM to the gateway.. dial-peer voice xxx voip incoming called-number [xxxxxxx] dtmf-relay h245-alpha codec .....[specify your codec or use voice class command to select codecs)   Rather than using h323 to cucm, you could also use sip. Just create a sip trunk on CUCM to the gateway and create a sip inbound dial-peer from CUCM dial-peer voice xxxx voip session protocol sip incoming called-number [xxxxxxx] dtmf-relay rtp-nte codec .....[specify your codec or use voice class command to select codecs)   ... View more

Hi, The best thing to look at is the cucm audit logs. That should tell you who did a reset of the pin. ... View more

Hi Dennis,   No I dont believe I got any further with this ... View more

Hi, With one-way audio the key things to verify is that there is connectivity between the media path used for the rtp streams. From your ITSP to the CUBE interface facing the ITSP, you need to check the IP address that CUBE is asking the ITSP to send media to. Unfortunately because you have hidden this, its difficult for me to tell. Also if you have multiple interfaces on your CUBE, then you need to ensure that you have configured your sip bind statements correctly so that signalling and media originates from the right interface. ... View more

Mark, The flow for call hold/transfer with CUCM is this 1. CUCM sends re-invite with inactive SDP to indicate a break in media path. 2. CUCM sends a DO to insert MOH or to resume media stream NB: CUCM expects a send-recv offer with SDP in responde to the DO. If CUCM gets an inactive offer SDP in the 200 OK, the media path remains inactive state and the call drops So when you say you do not see the re-invite with inactive SDP then something has gone amiss. We need to look at the logs to see why. CUCM must indicate a break in media path using a=inactive ... View more