H.323 GW - SIP Calls cannot go through

[Jefferson Islan Silva]

I have a CUCM 7.1.5 working properly, with a H.323 gateway. External calls needs to go through via SIP Provider.

I have installed a CUBE IOS on the H.323 gateway and SIP Calls to SIP provider cannot go through.

ON CUCM side, MTP, Inbound and Outbound fast start are enabled. SIP Calls are going through using G.729r8 codec.

I'm not to identify why the call cannot go through, seems that everything is configured properly, but cannot identify what's wrong.

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

h323

  emptycapability

sip

  bind control source-interface FastEthernet0/0

  bind media source-interface FastEthernet0/0

  registrar server expires max 3600 min 3600

  redirect contact order best-match

  localhost dns:sip.solutionsvoip.com.br

voice class codec 1

codec preference 1 g729r8

codec preference 3 g711alaw

codec preference 4 g711ulaw

!

voice class h323 1

  h225 timeout tcp establish 2

  call start fast

  no call preserve

interface FastEthernet0/0

ip address 192.168.1.5 255.255.255.0

speed 100

full-duplex

h323-gateway voip interface

h323-gateway voip bind srcaddr 192.168.1.5

sccp ccm group 1

bind interface FastEthernet0/0

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 2 register SP_TRANSCODER01

associate profile 1 register SP_MTP01

!

dspfarm profile 2 transcode universal 

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729br8

maximum sessions 1

associate application SCCP

!

dspfarm profile 1 mtp 

codec g729r8

maximum sessions software 100

associate application SCCP

!

dial-peer voice 3 voip

description Local Calls

translation-profile outgoing DIGITSTRIP-CUCM

destination-pattern 11[2-9].......

session protocol sipv2

session target sip-server

session transport udp

voice-class codec 1

voice-class h323 1

dtmf-relay rtp-nte

no vad

!

dial-peer voice 5 voip

incoming called-number .T

voice-class codec 1

voice-class h323 1

dtmf-relay h245-alphanumeric

no vad

sip-ua

credentials username 116618 password 7 12345 realm sip.solutionsvoip.com.br

authentication username 116618 password 7 12345 realm sip.solutionsvoip.com.br

retry invite 4

retry response 3

retry bye 3

retry cancel 2

retry register 10

timers register 100

registrar dns:sip.solutionsvoip.com.br expires 3600

sip-server dns:sip.solutionsvoip.com.br

host-registrar

I have attached the full sh run

What's wrong for this environment?

Thank you and I really appreciate any help

[Chris Deren]

What IOS version did you install?

If 15.1T + add the following command:

voice service voip

no ip address trusted authenticate

If that does not help, post "debug ccsip messages" for failed call.

HTH,

Chris

[Jefferson Islan Silva]

No, the IOS is 15.0.1

[Chris Deren]

Can you post the debug then?

Chris

[Jefferson Islan Silva]

Hi,

Please take a look on the traces

[heathrw]

Is that making calls or receiving calls?

Ask your service provider if they have configuration guidelines for CUBE

as they might have specific requirements and save some time.

Sent from Cisco Technical Support iPhone App

[Jefferson Islan Silva]

I'm making calls...

[Chris Deren]

You are not getting back anything from provider after your INVITE message, is your authentication working?  Can you ping their Proxy?  I suggest you contact your provider for assistance.

HTH,

Chris

[Jefferson Islan Silva]

Yes, the authentication seems to be working fine, because the router is registered normally when I type the command show sip-ua register status

This is the second SIP Provider and I cannot estabilish the connection

[heathrw]

Can you recieve calls?

[Jefferson Islan Silva]

I'm not trying to receive calls, because I only have plan to place external calls.

Anything else?

[Dennis Mink]

In your logs-new.zip, I am only seeing outbound SIP messages sent to your provider. Where is the rest, or are you not receiving anything?

[khayes1984]

So you do not have inbound VoIP dial-peers coming to CM. You need the following

dialpeer voice 2 voip

description Incoming to Cisco Unified CM Sub or Pub (Which ever)

destination target " DID"

session target ipv4: ip of CM

session protocol sipv2

.dtmf-relay rtp-nte or dtmf-relay h245-alphanumeric, or dtmf-relay sip-notify force rtp-nte <-- not sure on this one but check to make sure I'm correct

no vad

!

dial-peer voice 3 voip

description Incoming to Cisco CM Cluster

destination target .

session target sip-server

session protocol sipv2

dtmf-relay rtp-nte or dtmf-relay h245-alphanumeric, or dtmf-relay sip-notify force rtp-nte <-- not sure on this one but check to make sure I'm correct again.

no vad

!

check this commands to make sure they are laid out right, I'm half asleep so lol, also make sure you have voice translation rules to translate your inbound and outbound calls.

Sent from Cisco Technical Support iPad App

[Jefferson Islan Silva]

I'm not getting SIP INVITE reply.

I have checked and I'm getting SIIP INVITE timeout.

How I can fix that?

[Jefferson Islan Silva]

Anyone else?