Problem with SIP trunk between CUBE and SIP Provider

[Maxim Abramov]

Hello Everyone,

Currently we are using CME as a main telephony solution and all phones lives on it. With this everything works fine and we can make inbound and outbound calls through SIP trunk. But we want to switch from cme to cucm 8. I am going to keep sip configuration on cube/cme and use it as a voice gate. Between cucm and cube I am using h.323:

cucm<-H.323->cube<-SIP->Provider.

With this configuration I can`nt make calls at all from sip trunk every time call downgrade to isdn.

In sip logs (ccsip all, ccapi onout) I can that in call which failed there is not SDP. I was thinking it could be problem with early-offer, but I tried to apply it globaly in voice sevice voip -> sip and no joy.

Please find attached logs, one example with call from cucm and another one with call from cme. Config file in separated file.

Cheers,

Maxim Maxim

[Ayodeji Okanlawon]

Maxim,

First of all you need to make adjustments to your config. If you have a h323 trunk from CUCM to the CUBE, you have to define a dial-peer to reflect this. Your calls come from CUCM to the CUBE on dial-peer 101. which is a sip dial-peer.

Second, there is no benefit doing h323----SIP. It creates more harm and its more difficult to troubleshoot. A homogenous solution like sip---sip is much better and easier to troubleshoot.

To do this you need to do the ff

1. Set up a sip trunk from cucm to CUBE

2. You already have a sip dial-peer for inbound calls from cucm so you can leave this but for granularity, it is better to have seperate inbound and outbound dial-peer so I suggest you seperate them. So consider seperating them.

The third issue I see is that on the outbound invite to your sip provider, I do not see any SDP advertised. If you dont send any capabilites to your provider, there is no way they will know what codec etc to use. Bear in mind that CUCM by default does not do early offer, so your best bet is to configure it on the CUBE

So configure early offer as follows..

voice service voip

sip

early-offer forced

Once you have made the changes do a test call and send only the ff debug

debug ccsip messages.

If you do not change the inbound leg from cucm to sip, you will need to send debug voip ccapi inout also.

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[Maxim Abramov]

Thank you aokanlawon for respond.

At this moment I want to stick with h.323 and fix this problem if it possible.

What do you mean by saying to adjust config ? Do you want me to create separate incoming dial-peer for h.323 ?

I`ve got a configured sip trunk from cucm to cube, but I could not manage to make it work so far. But to be hounest with you I did`nt try siriously))

Anyway, I did notice that I am not sending SDP when I am calling from cucm and I did try to put this early-offer comand globaly. It didn`t help. But if you want I can do it again and I will attach new bunch of logs, is "ccsip all"  "and ccapi inout" do you want to take a look at "h.323" debug as well ?

Bets Regards,

Maxim

[Ayodeji Okanlawon · Accepted solution]

OK. But trust me I have done a few of this and h323---SIP is not the way to go. May I ask why you want to stick with h323?

Have you configured the CUBE as a h323 gateway in CUCM? or how have you configured the CUCM to send calls over h323 to CUBE?

However if you want to do this..then

1. You need to create a new dial-peer

dial-peer voice 102 voip

incoming called number .

codec g711a (if you want to use g711)

dtmf-relay h245-alphanumeric

no vad

and adjust this one..

dial-peer voice 101 voip

description SIP dial peer

translation-profile incoming INBOUND_SIP

translation-profile outgoing OUTBOUND_SIP

preference 1

max-conn 10

destination-pattern 9T

rtp payload-type cisco-rtp-dtmf-relay 101

rtp payload-type nte 102

session protocol sipv2

session target sip-server

incoming called-number (this needs to  match the your DDI coming from your sip provider)

dtmf-relay rtp-nte

codec g711alaw

So the first dial-peer will be the inbound h323 leg...This you dont have at the moment...You may also need to apply your xlation rule here depending on how you want it

The second dial-peer wil server two purposes

1. Outbound sip leg to your ITSP

2. Inbound sip leg from your ITSP to your DDI..so you need to configre inoming called number XXXXXX (where XXXX is your DDI)

Please do this and send the debugs

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

[Maxim Abramov]

Ok,

I will do it later, thanks !

Cheers,

Maxim

Edit:

Fogget to say that I configured h.323 gateway on cucm.

There are two log files one is bad outgoing number and the second one is bad incoming number.

Thanks in advance.

Edit:

In this logs I am using h.323 dial-peer 2598.

[Maxim Abramov]

I have just fixed this problem !

I changes h.323 configuration from cucm side, I checked "Enable Outbound FastStart".

Now I can make outbound calls, but inbound still dosn`t work. I think it is problem related with number translation rules or dial-peer mutching logic. I am going to review this area at some point.

Thank you for your help ! You gave me a clue !

I am going to rate you 5 stars.