SIP trunk issues only on international calls

[fsousa]

Hi dear all,

We are configuring a SIP trunk to replace the PRI lines with same provider.

We configure the voice gateway and the CUCM, we can receive calls from outside(inbound call working), we had problems with ougoing calls, we spoke with provider to check on their side and now  we can call mobile numbers and we can call also land lines numbers all over the country, but the problem we are facing is that we can not call internationals numbers. We hear busy tone but the person we called, tell us that the phone rang but when he tries to answer the call last for 7 seconds and he can not hear anything then the call  is dropped.

When I am configuring SIP trunks, whenever I have problems with outgoing calls none of them work(mobile, land lines and internationals) but on this customer international calls are not working and the provider is telling me that when we try a international call the sdp does not contain codecs but I am using the same configuration on the dial-peer of the outgoing calls that are working(on the voice class codec).

Could you please help tshoot this problem?

Attached the configuration and the output of debug ccsip messages

Best regards

FSousa

[b.winter]

In the debug you attached, you didn't send any SDP in the INVITE to the provider. So, if you say, in the working calls you send SDP, then you're probably matching a different dial-peer.
Not on all dial-peers the command "voice-class sip early-offer forced" is configured.

Maybe you should check first, if you are matching the correct dial-peers...

[fsousa]

Hi @b.winter ,

Thank you for you camment, for calls that are working i am using diferent dial-peers, for the international call i am using the below dial.peer:

!
dial-peer voice 203 voip
description === CHAMADAS INTERNACIO ===
translation-profile outgoing OUT->SIP
destination-pattern 00T
session protocol sipv2
session target ipv4:10.131.0.210
session transport udp
voice-class codec 1
voice-class sip early-offer forced
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay sip-notify cisco-rtp sip-kpml rtp-nte

regards.

[b.winter]

Have you confirmed with a debug, that the call really uses this dial-peer. If yes, it should send SDP in the outgong INVITE. But your debug output says otherwise.

use the debugs, to check the dial-peer matching:
debug voice ccapi ind 1
debug voice ccapi ind 2
debug voice ccapi ind 74

[fsousa]

Hi dear find attached the debug you requested ?

best regards

Fsousa

[b.winter]

The debug output is not for me, it was for you.
I don't know, which dial-peer is the correct one, only you know that.

Compare the output for a national call and a international call. Like I said, maybe the calls use different dial-peers. If yes, check the differences between the dial-peers.

[fsousa]

But with useful commands you recommed we can confirm that the international calls are using the 203 dial-peer. Now I am going to compare the output for national calls and internationai calls and check the diference on dial-peers,  and give you feedback.

regards

[fsousa]

Hi @b.winter ,

i did compare the output for national calls and international, they were use different dial-peer as said before. Cheched the difference between them, did not see significant difference other than the dtmf. and dicided to use the same dial-peer (202) that is working for national calls for international calls as you can see on the output attached, only the national calls are working the international we had the same problem people from the other side said that the phone is ringing but when the try to answer it dropped.

!
dial-peer voice 202 voip
description === CHAMADAS  REDE MOVEIS ===
translation-profile outgoing OUT->SIP
preference 1
destination-pattern .T
session protocol sipv2
session target ipv4:10.131.0.210
session transport udp
voice-class codec 1
no voice-class sip early-offer forced
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte h245-alphanumeric sip-kpml sip-notify
no vad
!

any clue ?

Regards.

[b.winter]

On dial-peer 203, you don't have vad disabled.
The command "no vad" is not configured there. Maybe the Provider doesn't like it, if VAD is enabled.

Edit:
Please also please provide a file with the output for a national call and a second file with the output for the international calls.
debugs to use:
debug voice ccapi ind 1
debug voice ccapi ind 2
debug voice ccapi ind 74
debug ccsip messages

[fsousa]

hi @b.winter 

Now i am using the dial-peer 202 for national and international calls, i did shutdown the dial-peer 203 temporarily to test the international calls on an working dial-peer.

regards

[b.winter]

And? Is it working or not? Do not let everything be pulled out of your nose...
I cannot read your mind.

How about the debug outputs of a working and a non-working call?
How about the missing command "no vad" in the dial-peer 203? Have you configured it?

If you don't do what we suggest, how to you expect to get help?

[Scott Leport]

Based on that debug alone and the config you've shared, it would appear that you're matching the correct dial-peer for International calls, but as already mentioned check it against a known-working dial-peer and make some modifications on the dial-peer for International calls based on that. The config is raft with inconsistencies (such as early-offer enabled & disabled on selected dial-peers, vad enabled / disabled on certain dial-peers, codecs hardcoded and voice-class codecs configured, shutdown dial-peers etc etc) so it would be best to address this through some config standardization.

I would consider enabling early-offer at the global level and remove it from the selective dial-peers. It'll keep things simpler and eliminates the risk of forgetting to enable it on other dial-peers when it is required.

To enable globally:

voice service voip
Device (config-voi-serv) sip
Device (config-voi-sip) early-offer forced

Then consider removing the early-offer config from the dial-peers, config example below:

dial-peer voice 123 voip
 default voice-class sip early-offer forced
 

[fsousa]

Hi @Scott Leport ,

Thank you for replay, already tryed to removed unnecessary config as show on the file below. Tommorow i am going to aply your sugestion regarding early offer.

best regards.