Hi Guys,
I hope someone can help me in our issue with our Cube router.
We recently migrated from PRI to SIP. We just learned that MOH from the flash will not work on a SIP trunk in a cube environment. We want to enable multicast moh since unicast m...
Hi Guys,
Want to ask assistance from you guys again. We migrated from PSTN to SIP last year. Our set-up is like ITSP>>(sip)>>Cube>>>CCMv4>>(sccp)>>Unityv4. All calls inbound and outbound are working. However, when a call from mobile calls our Unity ...
Hi Guys!
Good day! I have an upcoming project wherein we will migrate PSTN centres to SIP. These centres are composed of 130 sites and i was advised by my boss that SIP trunk will be installed individually per site. No TEHO, no intersite calling bet...
Hi Guys!
Good day! Hope someone can help me with my dilemma involving an ISR router and a very old call manager. We are in the process of migrating several of our centers from PSTN to SIP. Most of these centers are using old Cisco call manager versi...
Hi Guys, Good day! Hope everyone is doing well at the moment. We are having some weird issue with our callmanager and unity connection. They both on version 8.x. Whats happening is that when a call for an extension is forwarded to voicemail (tick VM ...
Hi Rajan,
Thanks for your reply. I did not enable that option. Once "Send send-receive SDP in mid-call INVITE" is enabled, the first problem (putting caller on hold then resume = no audio) doesnt get resolved.
Also, it does work when its unicast mo...
Hi Dennis,
Thanks for your reply. The SIP debug is already attached on my first entry (mmoh-noaudio.txt).
One more thing that i noticed, the moment the call is transferred, i still see the multicast ip 239.1.1.1 on one of the call legs. So its possi...
Thanks a lot Les Pruden! This is what im looking for. Our's are VM running on ucs-c. I guess we can now proceed with the project. I appreciate your help sir!
