I found this weekend that the default gateway for one of our UCCX servers is pointing to an incorrect gateway. The SIP router it's pointing to works, but won't provide the redundancy that the core switches will provide. Found this was incorrect this weekend when the SIP router was rebooted.   Any issues updating the default gateway? Was just worried that the licenses would have to be updated.   Thank you in advance. ... View more

I'm very new to VoIP so go easy on me.   Here is our current dial-peer configuration. We are looking to just have any call from our SIP provider to be passed directly to CUCM. We are an organization with a lot of numbers all over several states. Maintaining a e164 list isn't a major issue but it will be quite a list and just the fear that a number is missed and then calls don't process.   Is there a better way to just pass through these incoming invites?   dial-peer voice 11 voip  description Calls to CUCM (Pub)  session protocol sipv2  session target ipv4:10.19.254.10  destination e164-pattern-map 200  voice-class codec 1 offer-all  voice-class sip asserted-id pai  dtmf-relay rtp-nte  no vad ! dial-peer voice 12 voip  description Calls to CUCM (Sub)  preference 1  session protocol sipv2  session target ipv4:192.168.30.202  destination e164-pattern-map 200  voice-class codec 1 offer-all  voice-class sip asserted-id pai  dtmf-relay rtp-nte  no vad ! dial-peer voice 20 voip  description Calls from NexVortex  session protocol sipv2  session target sip-server  incoming called e164-pattern-map 200  voice-class codec 1 offer-all  voice-class sip asserted-id pai  voice-class sip profiles 1  dtmf-relay rtp-nte  no vad ! dial-peer voice 31 voip  description Calls to NexVortex  translation-profile outgoing OUTGOING-SIP  destination-pattern .T  session protocol sipv2  session target dns:nexvortex.com  voice-class codec 1 offer-all  voice-class sip asymmetric payload full  voice-class sip asserted-id pai  voice-class sip call-route url  voice-class sip profiles 1  dtmf-relay rtp-nte  no vad ... View more

We are having an issue with UCCX that when agents are logged into the queues at 7:50ish AM, and then the queues open at 8:00 AM, that the calls are staying in the queues. If an agent logs out of the queue and then back in at 8:00 AM, he/she starts receiving calls.   Anyone seen anything like this?   UCCX  11.6.1.10000-51 ... View more

I have built out a new pair of 9500-40A switches with stackwise-virtual and they appeared to be running quite well with the exception that I couldn't get a neighbor for EIGRP. I just upgraded to 16.6.3 and now the stackwise-virtual command and it's configuration is gone. Anyone else having this issue? ... View more

Hello guys/gals,   We are in the process of reviewing our AT&T SIP IP Flex circuits due to the time that it's taking them to complete phone number ports and any other changes. Unfortunately, this is taking months, not weeks.   I have reviewed a SIP provider, NexVortex, and have been impressed with the one site that we've already stood up with them. The amount of time to be assigned a phone number and even the porting of numbers has been very quick, days not weeks or months.   For our offsite locations, we currently are using a mix of some locations using AT&T IP flex so that the location would have their phone numbers exist in their building and some locations have their phone numbers that exist back on our corporate SIP IP Flex account.   Each works just fine, but the location based IP flex circuits are 1) expensive and 2) requires a cube router to be installed at each location.   We are considering moving to NexVortex's mSIP (managed SIP) option for our corporate site and all branches. We would install two new 4431 routers, one at our Corporate Office, and one at our DR site. These would be connected over DIA circuits with a vendor that they peer with so that we can ensure QoS and monitoring.   While this on paper sounds like a great deal and gives us the flexibility and agility that we need, I'm just asking if someone could point out any issues that we might encounter.   For reference, our CUCM cluster has one CUCM server at our Corporate Office and one at our DR site. ... View more

Please forgive me for this is only my 2nd attempt at SIP and VoIP, I'm a route/switch guy.   We have setup a new branch on a SIP trunk to our local cable internet provider and inbound and outbound calls are dropping at around 5 mins 45 seconds every time.   I have read over their documents over and over and still am having no luck. Hoping the group could quickly identify the issue and explain this is why it is.   This debug is from an inbound call that goes to an AA, and then with a digit press, goes to the internal extension. This call disconnected with my phone timer at 5 mins and 46 seconds with a busy tone. ... View more