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CUCME to ASTERISK

Hello everbody,

I need a help with CUCME config.

I have the following situation:

I use Cisco 2911 with few IP Cisco 6921 SCCP Phone. Communication between these phones is OK, but I need to connect my Cisco 2911 to VoIP provider using SIP trunk to enable communication to the whole world from Cisco 6921 SCCP phone.

I have these paramters from VoIP provider:

  • •- IP: 178.22.117.58/30 (this is WAN IP of my 2911 router)
  • •- GW: 178.22.117.57 (GW to the provider)
  • •- DNS: 178.22.112.22 a 178.22.118.10
  • •- domain: sbcpeer.sipserver.cz port 5060 (195.122.207.108)
  • •- RTP stream gateway: 195.122.201.48/28
  • •- supported codecs: g711alaw (PCMA)
  • •- packets timeout ?: 20ms
  • •- DTMF: INBAND
  • •- authentication: username=sprintel1 secret=slu6ebn8. Trunk will
  • •- IP: 178.22.117.58/30
  • •- GW: 178.22.117.57
  • •- DNS: 178.22.112.22 a 178.22.118.10
  • •- domain: sbcpeer.sipserver.cz port 5060 (195.122.207.108)
  • •- RTP stream gateway: 195.122.201.48/28
  • •- supported codecs: g711alaw (PCMA), T.38 (na vyžádání)
  • •- pakets timeouts ?: 20ms
  • •- DTMF: INBAND
  • •- authenticate : username=sprintel1 secret=slu6ebn8. Trunk will be one way authenticate (for our outgoing calls). This is an Asterisk's paramter "insecure=invite". Our reservation dial pattern is 588 008 3xy.

I have done some settings.

When I dial from PSTN to Cisco 2911, I can see the incoming call but I don't hear any ringing voice in my PSTN Phone but the Cisco phone is ringing. After picking up the phone we hear nothing. And that's all.

I can't dial from Cisco IP phone to PSTN, busy tone have occured.

I really don't known where I have to use the domain: sbcpeer.sipserver.cz port 5060 (195.122.207.108) and where RTP stream gateway.

I supposed that I have problem with codecs transcoding.

Thanks to anyone who will help me.

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CUCME to ASTERISK

Best approach is that you give the job to a reputable consultant, or UC certified Cisco partner.

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