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Highlighted
Beginner

Re: CUBE-C2951 remote destination and wireless phones not working

Thanks a lot for your help. I solved the problem today. 

 

Here is my solution:

voice class sip-profiles 1
 request INVITE sip-header To modify "(<.*:)(.*@)" "<sip:2268156t7@" 
 request ANY sip-header From modify "(<.*:)(.*@)" "<sip:2268156t7@" 
 request REGISTER sip-header From modify "(<.*:)(.*@)" "<sip:2268156t7@" 
 request INVITE sip-header P-Preferred-Identity modify "<sip:0(.*)@(.*)>" "<sip:49\1@\2>" 
dial-peer voice 1 voip
 description *Outbound to Sipgate*
 translation-profile outgoing OUTGOING
 destination-pattern 0T
 session protocol sipv2
 session target ipv4:217.10.68.150
 session transport udp
 voice-class codec 1  
 voice-class sip asserted-id ppi
 voice-class sip profiles 1
 voice-class sip bind media source-interface GigabitEthernet0/0
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 2 voip
 description *Incoming from Sipgate*
 translation-profile incoming INCOMING
 preference 1
 service session
 session protocol sipv2
 session target ipv4:217.10.68.150
 session transport udp
 incoming called-number .T
 voice-class codec 1  
 voice-class sip asserted-id ppi
 voice-class sip profiles 1
 voice-class sip bind media source-interface GigabitEthernet0/0
 dtmf-relay rtp-nte
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad
!
dial-peer voice 3 voip
 description *Outbound to Sipgate*
 translation-profile outgoing OUTGOING
 destination-pattern +T
 session protocol sipv2
 session target ipv4:217.10.68.150
 session transport udp
 voice-class codec 1  
 voice-class sip asserted-id ppi
 voice-class sip profiles 1
 voice-class sip bind media source-interface GigabitEthernet0/0
 dtmf-relay rtp-nte
 no vad

I need to insert my outbound interface to all dialpeers which are faced to my ITSP. The key was the tip from Tony that my ITSP is sending his answer to my internal ip 192.168.83.254. 

 

Thank you very much again for the fast help. 

 

Best regards from Germany

Christoph

Highlighted
VIP Mentor

Re: CUBE-C2951 remote destination and wireless phones not working

I am glad you got to the bottom of it but I find the solution interesting. If your sip bind statements are wrong that would affect all calls not just SNR or diverted calls...The other interesting thing is that the first INVITE sent for your call got a 407 request, so your ITSP was able to reach you and then when you send INVITE with the authentication credentials, they were not able to respond. Either the information provided is incorrect, the symptoms described is in accurate or something else is going on...Anyway have fun in Germany

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