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PRI EI CARD

Learnercisco
Level 1
Level 1

Hi champs,

i am trying the add the NIM-2MFT-T1/E1 card in the CUCM (12.5) and the CARD is installed in 4451X. I configured SIP Trunk between CUCM and VG4451. i can only add the FXO/FXS Card in the CUCM which is showing in the Module 0

is there any configuration required on the VG4451 to add this card in the CUCM. 

Thanks in advance. 

 

25 Replies 25

Hi Roger thanks for the support

I verified the connectivity and it was due to firewall. Now I  can recieve the call from outside to the phone but no voice. 

Secondly for the outide call from cucm to isdn Provider do I need to configure router pattern that will show the full DID numbers (assigned to my extension ) to the outside phone or mobile.

Thanks for your support

The audio part is setup directly between the receiving endpoint and the gateway, so if you have a firewall between your gateway and the rest of your network you’ll need to allow this traffic as well. May I ask what reason you have to put a firewall in between your gateway? With ISDN there are no way that you can get any intrusion attacks that can penetrate your network.

On your other question you’ll need to check with your service provider on how they want you to send calling number. As it is ISDN you’ll need to know how many digits they want and if they require a specific type and plan on the ISDN side. Often you can check what the service provider sends inbound to you and use the same outbound, but that’s not always the case. Best is to check with your provider what requirements they have.



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Hi Roger,

Thanks for the reply

Now there is no firewall in between Gateway and CUCM. Still i have once audio issue on receiver side (Phone 7821). for the outside, example the DID range 33559xxx, will send back first 4 digit along with our extension number (33559100) back to the provider. 

thanks for your support. 

What did you mean by this then if there is no firewall, “I verified the connectivity and it was due to firewall. Now I  can recieve the call from outside to the phone but no voice”?



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For reference this is how your service provider sends inbound calls to you.

 

Calling Party Number i = 0x2183, '97434105' 
                Plan:ISDN, Type:National 
        Called Party Number i = 0xC1, '5023' 
                Plan:ISDN, Type:Subscriber(local)

 



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Hi Roger,

Firewall we placed just because there was a sip line in the past. So when you suggested the connectivity issue between VG and CUCM, i completely isolated the Telephone LAN from Firewall. The one way audio issue is not because of firewall. i see in the community that some people face the same problem in PRI EI calling. 

The number you mentioned is the ISDN log the one which is our (real extension) sending by the Service Provider.  

Thanks in advance.

 

Issues with no or one way audio is about 100% related to network issues. I see that you have or at least had specific IP routing statements in your configuration. Add a default route instead and remove these and then test if you have connectivity between your gateway and the phone that is receiving the call.

Not sure if I understand what your comment about the numbers are about. Would you please mind to elaborate?



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Hi Rogers,

The one way audio issue is not related to the network. I think its related to codec issue from provider or our side. person is unable hear me on the extension but i can hear his voice.  there is layer 3 switch between call manager and Voice gateway. Do we need to add any RTP command on this device. can you suggest any debug commands to rectify this Audio issue. thanks for your support. 

 

Hi Roger,

Thanks for your support 

you are right that its a network issue, i put the ip routing in some switches and now i can hear the voice from two ends now. I appreciate you knowledge and skills in the platform.

 

Hi Rogers

For the call from CUCM to PSTN. I want to confirm if I configure only route patterns on cucm for national and International calls, do I need to do  configuration on voice gateway to match the incoming number dial to the PSTN

Thanks in advance. 

 

 

If you think about it for a moment what do you self think? A dial peer match what you define and if you have a need to match anything else outbound you need to configure that. In general with proper configuration you should not have a need to have specific configuration for the inbound direction.



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