One way Communication issue since changed IP addressing on voice vlans
We have changed IP addressing of our VoIP phones from Public Range to Private 10.x.x.x. Our Main Site IP addresses change took place before I started so don't know if some configuration change was made on CUCM 6.1.3 to make it work. I changed the IP addressing of Voip Phones on one of the Remote site connected to main site (where Publisher / Subscriber resides). At the remote site I enabled the DHCP scope to allocate IP addresses to IP phones. I created the voice vlan and configured the correct voice vlan to switchport IP phone is connected. Phone gets the IP from DHCP scope and also gets correct parameters such as gateway,option 150 etc. Phone gets registered to Subscriber call manager. I have checked the IP cache and I see the session to port 2000.When
I call from our main site to phone assigned to new VLAN ,Remote site phone can hear us but we can't hear them and issue remains the same when call is initiated from remote site to phone's on other VLAN's or sites.
At remote site I installed another phone and assisgned the switchport to new voice vlan then both new phones can communicate to each other but same issue in communication to phones on other vlans / locations
I have checked all the access-lists and FWSM but couldn't see any issues ....ANY IDEA what's causing one way communication issue cos previous team changed the IP addresses on our main site but no such one way communication issue. Interesting thing is that I picked one of the free subnet from Main site IP address planning structure and implemented at remote site and had no issues with communication at all
Re: One way Communication issue since changed IP addressing on v
Thanks for help - I have managed to resolve the issue - Test phones we used at the Main site which had issue hearing the caller from remote site had some access-list applied to it's Vlan interface (SVI) and we also found assymetric routing which was causing outbound packet to go out the other path hence another access-list had to be mofied for the time being untill some route maps can be defined to make sure the correct packet flow path.
Now, one issue is left - PSTN user from outside when calls remote site phone - Remote site answers the phone can hear the caller but PSTN caller can't hear the remote site user. Previously we were using public address - Now we are using Private IP's - I will start looking in to the access-lists but is there any NAT issue to be taken care of.
Looking forward to your response and thanks again for your impressive answer earlier..(I had no clue about ? button - learning new technology the hard way)
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