[cisco-voip] sip trunk problem

[Baris Gulten]

Hi everybody,

I have ccm 6.1.1, also 2851 router. 

I define sip trunk on 2851 router. I have trouble when i decide make a call.
Lets hold my hand J

 

 

Here is the debug result

vgw#sh debug

CCSIP SPI: SIP Call Events tracing is enabled   (filter is OFF)

CCSIP SPI: SIP error debug tracing is enabled   (filter is OFF

-------------

#

*Feb  2 20:40:30.779: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued
event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP

*Feb  2 20:40:30.783: //109/803045A3FD12/SIP/Event/sipSPICreateRpid:
Received Octet3A=0x83 -> Setting ;screen=yes ;privacy=off

*Feb  2 20:40:33.295: //109/803045A3FD12/SIP/Error/sipSPIDoAudioNegotiation:
Media negotiation failed for m-line 1

*Feb  2 20:40:33.295: //109/803045A3FD12/SIP/Error/sipSPIDoMediaNegotiation:


no valid fax or audio streams

*Feb  2 20:40:33.295:
//109/803045A3FD12/SIP/Error/ccsip_api_call_cut_progress: MediaNegotiation
Failure - Send Cancel

*Feb  2 20:40:33.295: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued
event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT

 

 

Here is the config below (i did before cme router with this config)

!

voice rtp send-recv

!

voice service voip 

 allow-connections h323 to h323

 allow-connections h323 to sip

 allow-connections sip to h323

 allow-connections sip to sip

 supplementary-service h450.12

 h323

  session transport tcp calls-per-connection 200

 sip

  bind control source-interface GigabitEthernet0/1

  bind media source-interface GigabitEthernet0/1

  registrar server expires max 3600 min 3600

  no call service stop

!

voice class codec 1

 codec preference 1 g729r8

 codec preference 2 g729br8

 codec preference 3 g723r63

 codec preference 4 g723r53

 codec preference 5 g726r24

 codec preference 6 g726r16

 codec preference 7 g726r32

 codec preference 8 g723ar53

 codec preference 9 g723ar63

!  

interface GigabitEthernet0/1

 ip address xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx (defined real ip by sip
provider)

 duplex auto

 speed auto

!

 

!

dial-peer voice 100 voip

 preference 1

 voice-class codec 1

 destination-pattern 053T

 session protocol sipv2

 session target ipv4:xxx.xxx.xxx.xxx  (sip server ip)

 dtmf-relay rtp-nte

 clid network-number xxxxxxxxx

!

 

 

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