03-29-2018 01:09 PM - edited 03-17-2019 12:31 PM
I found this config in our forum :
interface loopback1----------------------------------------Interface pointing to SIP provider 1
ip address 10.10.10.1 255.255.255.0
interface loopback2-----------------------------------------Interface pointing to SIP provider 2
ip address 20.20.20.1 255.255.255.0
dial-peer voice 10 voip
description “Primary path to SIP SP-1”
destination-pattern 91[2-9]..[2-9]......
session protocol sipv2
session target ipv4:10.10.10.2
voice-class sip options-keepalive
voice-class sip bind control source-interface loopback1
voice-class sip bind media source-interface loopback1
dial-peer voice 20 voip
description “Secondary path to SIP SP-2”
destination-pattern 91[2-9]..[2-9]......
session protocol sipv2
session target ipv4:20.20.20.2
preference 2
voice-class sip options-keepalive
voice-class sip bind control source-interface loopback2
voice-class sip bind media source-interface loopback2
1 Can anyone explain how to connect to sip Service provider with loopback ip? We usually connect with public ip via mpls..
2 How to separate incoming calls from different SP and how inbound dial-peer will look like at this case ?
Thank you
03-29-2018 02:15 PM
04-03-2018 02:07 PM
Can old style dial-peer config and uri based dial-peer coexist/sit on the same CUBE and work ?
04-03-2018 02:29 PM
04-05-2018 08:02 AM
HI, can u please tell me what is the algorithm in dial-peers, for example if there is 3 dial-peers to the same destination and dest pattern but different preferences and all of them has translation profile to change outgoing digits?
I see on the trace call is hitting 1st dial-peer not changing digits then hitting next dial-peer not changing digits then next dial-peer... I thought changing should happen before to hit second dial-peer because of translation profile is inside of the 1st dial-peer ... But looks like there is different algorithm ?
04-05-2018 08:41 AM - edited 04-05-2018 08:41 AM
The higher the preference on the dial-peer, the least preferred it will be. So if you have 3 DP's with same destination pattern but with a preference of 1,2 and 3 respectively, the DP with preference 1 will be used always. Then the DP with pref 2 and the DP with pref 3 will be the least preferred.
04-05-2018 08:45 AM
right pref 1 is preferred than prefer 2 , i know that, My question was why translation profile with changing dialed number didn't work when call hit dial-peer 1 ?
I see cal is hitting 1 dial peer than 2nd than 3rd , Although i had translation profile to delete some digits from dialed number... Is there any logic I am missing ?
thank you
04-05-2018 08:52 AM
here is config :
voice translation-profile StripTESTLocalAccessCode
translate called 7
translate redirect-called 7
voice translation-rule 7 ## for outg calls to WCS
rule 1 /^7011\(.*\)/ /011\1/ type international international
rule 2 /^7\(1.*\)/ /\1/ type national national
rule 3 /^7\(.*\)/ /\1/ type subscriber subscriber
dial-peer voice 7 voip
description ** Outgoing Dial-Peer to WCS_NY from CUCM **
translation-profile outgoing StripTESTLocalAccessCode
preference 2
destination-pattern 7T
session protocol sipv2
session target ipv4:x.x.x.x
voice-class codec 100
voice-class sip profiles 2
voice-class sip options-keepalive up-interval 30
voice-class sip session refresh
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
ip qos dscp af41 signaling
no vad
04-05-2018 09:03 AM
04-05-2018 09:11 AM
Nipun you are the best men, u always trying to help. Thank you. I could fix this part right now, I saw its sending subcriber or national while there was something with unknown and decided to delete them, there was type national,international and subscriber on the translation rules , I deleted types from rules and now calls goes trough. Now its hitting dial-peer 7 then changing it and my cell phone ringing .
Only thing now i need to figure it out when call connects i have no audio , its probably because they have different ip for signaling and differnt for media. I have to find out is it routing issue...
04-06-2018 09:40 AM
Nipun, I captured pkacp files from wan and lan interfaces of cube and sii ITSP is sending trying after my invite and my wan is getting it but CUBE is not sending anything to CUCM then to LAN port, neither trying or something else. I have their ip as trusted , i don't understand ... On the wan port i have only policy map for qos allowing ports for signaling and media , so it should not block them.
Where also I can look on my configuration?
thank you
04-06-2018 11:05 AM
03-31-2018 03:16 AM
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