07-23-2015 03:21 AM - edited 03-17-2019 03:44 AM
Hello all,
I have my CME router with two different link:
the first is VOIP and the second is POTS
I need to allow international outgoing calls to be routed through the POTS link
so i did these commands:
dial-peer voice 5 pots
corlist outgoing call-international
description Appel international via TT
preference 1
destination-pattern 000T
port 0/0/0:15
I have also configured the international calls to be routed through the SIP TRunk by using these commands
dial-peer voice 2200 voip
corlist outgoing call-international
description **International Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 000T
session protocol sipv2
session target ipv4:192.168.2.51
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
The problem is that calls always goes through the SIP trunk although i have specified the higher preference value for the POTS
Is there any missconfiguration
Please help me it is urgent!!!
Solved! Go to Solution.
07-23-2015 03:42 AM
Add "preference 2" under the VoIP dial peer. Please note that When you set the preference order, the lower the preference number, the higher the priority. The highest priority is given to the dial peer with preference order 0 and it is the default value.
Manish
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07-23-2015 12:14 PM
Hi,
Please share your config and the output of 'debug isdn q931'.
What is '000' for? Does it mean '0' as access code and '00' to match international dialing or any other interpretation.
Do you want to use the SIP trunk as a failover to PRI?
07-23-2015 03:42 AM
Add "preference 2" under the VoIP dial peer. Please note that When you set the preference order, the lower the preference number, the higher the priority. The highest priority is given to the dial peer with preference order 0 and it is the default value.
Manish
- Do rate helpful posts
07-23-2015 03:44 AM
Ok i will check this and i will come back to you
Thank you for your response
07-23-2015 03:48 AM
But i have made a simple test, i deleted the Sip tRunk configuration and i tested. The international calls fail
so i guess there is a problem with the POTS configuration
Have you any idea?
07-23-2015 03:53 AM
My response was based on issue with dial-peer selection. That configuration was incorrect, you need to correct it as per my last response. This call failing through POTS is a separate issue from dial peer selection order and requires debugs from the gateway for a test call.
Manish
07-23-2015 03:56 AM
Thank you a lot
i'm not familiar with CME. Know after making changes you have recommanded me i got the message that the dialed number is incorrect from my POST operator so it 's a good thing for the moment.
Thank you for your further help :)
07-23-2015 11:39 AM
Hello Marwa,
As Manish correctly said you can re-route your call based on preference level assigned on dial-peer however now seems like you have a problem with pots dial-peer even you have deleted the voip dial-peer ie SIP still calls fails not traversing over pots dial-peer.
Can you make one test call shut voip dial-peer and collect below debugs
# debug voice ccapi inout
# debug isdn q931
P.S : there would be a difference while sending over these dial-peer in pots explicit match will be strips however in voip it won't
Br,
Nadeem
07-23-2015 12:14 PM
Hi,
Please share your config and the output of 'debug isdn q931'.
What is '000' for? Does it mean '0' as access code and '00' to match international dialing or any other interpretation.
Do you want to use the SIP trunk as a failover to PRI?
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