No speech path over SIP Trunk between CS1K and IPO

[nortavaya]

Hello team,

We have configured a SIP Trunk between CS1K and IPO, all works fine

We can call from each site to another without any issue

Only one problem, in the CS1K side we have two sites (main and GR redundancy), users of the main site can call IPO users with no issue, however users from GR site (CS1K) can call IPO users but with no speech path

What can be the possibles causes of this issue ?

Can we suspect their network (Firewall, NAT...etc) ?

Thank you in advance

 

[nortavaya]

Many thanks I will keep you informed about the update

 

[nortavaya]

Hi bignose21, I think to one other thing than can causing this issue, may be MAC address of the phone was blocked on the Switch security list ! (customer side)

 

[nortavaya]

Hi bignose21, I enabled the traces on the IP Phones, but on the Wireshark, I see only some traces with ARP or STP, so no more informations that can help in our investigation

Customer confirm us that there is no restriction (Firewall, ACL..etc) on their side, however there some phones works fine when calling through SIP Trunk and some others got issue with speech path

Do you have an idea what can be the issue ?

Thank you

 

[bignose21]

I have had it before where half the phones had no dial tone. On investigation found one of the 2 Sig servers was not patched.

 

[bignose21]

If you run an enhanced trace in LD 80 against the phone does it look to be setting up a 2 way call

 

[nortavaya]

Thank you bignose21, yes in our case we have two Sig Server, the first is located on the main site and the second on the GR site, and all phones on the GR site are having this issue of speech path

On the phone IP address configuration, we already set S1 as Primary SS and S2 as secondary SS

Do you know how I can printout the IP phones registered on the First SS and those registered on the other SS ?

Thank you

 

[bignose21]

there should be no phones at the GR site as that should be in an inactive state. Ways to see how many sets are registered:
LD 117
=> stat ss
or
=> ecnt zone
or for details of sets
=> locrpt all

In the sig server see the details of the registered sets
> isetShow

Correct phone configuration is for the Primary site S1 & S2 are the primary sig server node IP. For the GR site the S1 is the GR site Node IP and the S2 is the primary site Node IP. This is because the correct registration process is the GR site phone goes to its local sig server and registers the callserver is in an inactive state and forwards a registration message out to the NCS this is either on NRS or Session Manager depending what you use, this forwards the request to Primary site and the set registers there.