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Problem with CUCM6 SIP trunks to CUBE

csg-admin
Level 1
Level 1

Hi all,

We have a setup where there are two CUCM6 publishers with a couple of phones registered to each cluster. Each CUCM has a SIP trunk configured to the same 2821 running a CUBE IOS. The configuration on the CUBE is:-

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

!

dial-peer voice 100 voip

description ** Outbound DP **

destination-pattern 9.T

session protocol sipv2

session target ipv4:10.212.1.101

dtmf-relay sip-notify rtp-nte

codec g711ulaw

no vad

!

dial-peer voice 200 voip

description ** Inbound DP **

session protocol sipv2

session target ipv4:10.212.1.103

incoming called-number 92051

dtmf-relay sip-notify rtp-nte

codec g711ulaw

no vad

All the phones on each cluster are G711. The problem is when a call is made from phone 123456 to 92051 the call rings through the SIP trunk across the dial peers, out through the other SIP and phone 2051 rings. However, there is only one way audio and when you examine each phone only 2051 has the G711 codec set as the receiver codec. On the other phone 123456 nothing is set for the sender or receiver codecs.

I cannot find out where the problem is. The device pools on the two CUCMs are identical so each phone has identical DPs, locations and regions, yet there is a mismatch somewhere.

Any ideas?

10 Replies 10

Tommer Catlin
VIP Alumni
VIP Alumni

Try adding in voice service voip

SIP

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

sip

Tommer Catlin
VIP Alumni
VIP Alumni

One other question, why would not just send the SIP request right to the CUCM servers? CUCM and CUCM BE will listen to SIP 5060 and 5061 by default. Send the SIP call directly to the server. I would think the CUCM would be able to pick this up and route it.

Thanks for the reply.

I'm trying to emulate a setup that we will be deploying elsewhere and wanted to get it working in the lab first.

I've added the SIP command to the config but it hasn't made a difference. The phones can ring so signalling is working. Then one phone will show the receiver codec as G711u and nothing else. The other phone does not show any codecs at all. So we get one way audio and then after about 20 seconds the call is disconnected.

OK have made some progress. Upgraded the cube to version c2800nm-adventerprisek9_ivs-mz.124-19b.bin.

Now I can successfully make voice calls across the CUBE with 2 way audio. However they only last for 18 seconds and then the call drops.

Any ideas?

Thanks,

Getting nowhere with this. The call is consistently dropped after 18 seconds.

Here is the setup

CUCM6A --->SIP TRUNK --->CUBE ---> H323 Dial Peer --->SIP TRUNK --->CUCM6B

Servers A and B are on different clusters.

Attached is a debug ccsip all from the cube.

Help.

Hi,

I have just simulated your scenarion in my lab. I got it work ok. However here is my suggestion as this is what I used in my lab...

Try this scenario..

CUCM6A --->h323 trunk --->CUBE ---> sip Dial Peer --->SIP TRUNK --->CUCM6B

CUCM6B --->h323 trunk --->CUBE ---> sip Dial Peer --->SIP TRUNK --->CUCM6B.

Summary:

CCM6A uses an h.323 trunk (ICT non-gatekeepeer trunk) to CUBE. From CUBE, sip dial-peers are configured to route the call to CCM6B. CCM6B is configured to accep the calls over a sip trunk.

CUBE config is as shown below:

Router#sh run

Building configuration...

Current configuration : 1473 bytes

!

version 12.4

service timestamps debug datetime msec

service timestamps log datetime msec

no service password-encryption

!

hostname Router

!

boot-start-marker

boot-end-marker

!

!

no aaa new-model

!

resource policy

!

memory-size iomem 5

ip cef

!

!

!

!

!

!

!

!

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

h323

sip

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

interface FastEthernet0/0

ip address 172.16.5.44 255.255.255.0

duplex auto

speed auto

h323-gateway voip interface

!

interface FastEthernet0/1

no ip address

shutdown

duplex auto

speed auto

!

!

!

ip http server

no ip http secure-server

!

!

!

!

!

!

!

control-plane

!

!

!

!

!

!

!

dial-peer voice 102 voip

destination-pattern 30..

session protocol sipv2

session target ipv4:172.16.5.5

dtmf-relay rtp-nte digit-drop h245-alphanumeric

codec g711ulaw

!

dial-peer voice 108 voip

session protocol sipv2

incoming called-number 40..

dtmf-relay rtp-nte

codec g711ulaw

!

dial-peer voice 109 voip

session protocol sipv2

incoming called-number 30..

dtmf-relay rtp-nte

codec g711ulaw

!

dial-peer voice 103 voip

destination-pattern 40..

session protocol sipv2

session target ipv4:172.16.5.6

dtmf-relay rtp-nte digit-drop h245-alphanumeric

codec g711ulaw

!

!

!

!

!

gatekeeper

shutdown

!

!

line con 0

line aux 0

line vty 0 4

login

!

!

end

Router#

NB: DN 40.. is my CCM6B

DN 30.. is my CCM6A.

I configured two trunks each on both ccm..one is an h.323 trunk for outgoing calls...the other is a Sip trunk for incoming calls...

Make sure that both trunks point to your CUBE gateway

NB: you have to use software MTP on both trunks or on atleast one of the trunks..

So ensure the trunk has a MRG assigned to it and check the MTP required box..

Hope this helps.

For troubleshooting Incoming SIP calls..

enable debug ccsip messages

debug ccsip events

debug ccsip calls

Please rate all useful posts

Thank you very much for taking the time to help.

I have tried your configuration and it works perfectly!

I guess that what I was trying to achieve CUCM6A --> SIP --> CUBE --> VOIP DP --> SIP --> CUCM6B

is just not possible.

Many thanks.

Hi, looking into aokanlawon's configuration, I have a question.

For the scenario is

CUCM6A --->h323 trunk --->CUBE ---> sip Dial Peer --->SIP TRUNK --->CUCM6B

You are using 108 as incoming and 103 as outgoing dialpeer, right?

I don't know why you are using an ICT but the 108 dialpeer still have "session protocol sipv2" in it.

Could you please explain? Thanks.

yes, 108 is incoming and 103 is outgoing..

without the session protocol sipv2 in the incoming, the h.323 trunk coming from callmanager does not get "converted to sip" which is what we use for outgoing...hence if that is not there, the call will proceed with h.323 all the way and the incoming trunk on the CUCM6B is a SIP trunk..

NB: You can also use a sip to sip trunk all the way or use an h.323 to h.323 trunk all the way.

Please rate all useful posts

not necessarily. I have used H323 from CUCM to the router, then SIP to another end point and it was fine. I did have a problem with the router IOS version though. I had to upgrade to latest and greatest to get SIP to work correctly on the router. The router can handle the H323 to SIP fine, it's the IOS version that can not handle it.

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