01-05-2015 07:32 AM - edited 03-17-2019 01:29 AM
My network consists of a central hub with a CUCM publisher, and 9 remote sites, each with their own 2951 Voice router and SIP trunk. There is a SUbscriber at our DR site too.
We are in the process of changing our network provider, and some sites have IPVPN with the new guys and SIP with the old guys.
So via this setup I have noticed that the voice routers send calls to other voice routers when processing a call, and I'd like to stop this and make each voice router be able to work on it's own.
I thought it was to do with MTP points and MRGL, but as far as I can see, resources are not being shared, except for the Publishers.
So I don't understand why the routers are doing this.
Can anyone shed any light please?
Thanks for reading.
01-05-2015 07:43 AM
Hi
You'll have to tell us what you mean by 'I have noticed that the voice routers send calls to other voice routers'.
Aaron
01-05-2015 07:45 AM
Sure, let me gather an example
01-05-2015 07:55 AM
It's when "Media Termination Point is Required" is tickled on the trunk page in CUCM.
Then I do a "show call active voice compact" on the voice router and it shows call legs for a current call:
UKGGVR1#sh call active voice compact
<callID> A/O FAX T<sec> Codec type Peer Address IP R<ip>:<udp>
Total call-legs: 2
852 ANS T3 g711ulaw VOIP P55555555555 xx.xxx.xxx.xxx:53870
853 ORG T3 g711ulaw VOIP P55555555555 10.210.97.5:19512
Actual phone numbers have been replaced with 5s!
So 10.210.97.5 is a different sites voice router. This site (GG) is 10.214.97.5
But the MRGL for GG only contains GG_MRG and CallManager Software_PUB
And the GG_MRG contains CFB and MTP from the GG Voice router, and XCODE from the voice router at the Publisherr site.
01-05-2015 08:15 AM
Is this a call routing directly between the SIP gateways, rather than using a foreign site transcoder?
e.g.
Call to 5555555555 is sent to local GW by CUCM
Local SIP GW has a dial-peer matching 555555555555 pointing to 10.210.97.5 (remote site)
Aaron
01-05-2015 08:53 AM
Nope, it's an external call coming into the site for that site. The dial peer directs the incoming call to CUCM, which then sends it back to that site.
The 10.210.97.5 shouldn't be involved at all.
01-05-2015 10:04 AM
What IP address did you redact on your 'sh call active voice compact' output? Was that the SIP SP?
From your description, you seem to have your MRGs etc set up correctly. I would verify that they are correctly assigned to the DP and that the gateway, transcoders and conference bridges, and local phones on each site are correctly assigned to that DP. Also that there aren't any errant assignments of MRGs from the wrong site on any devices that might override that..
Transcoders run over SCCP, so if you run a 'show sccp connections' on the gateway that shouldn't be involved when a call is in progres, do you see active connections? If not, then it's routing rather than the MRGL/transcoding that's a problem.
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