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2013
Views
45
Helpful
26
Replies

Translation rules and profile dial-peer

enkli
Level 1
Level 1

HI all,

I have got sip number from numbers from SIP provider so far the translation rules profiles and dial peers are as below:

voice translation-rule 1
rule 1 /022799232/ /501/
!
voice translation-rule 2
rule 1 /5../ /022799232/

and translation profiles

voice translation-profile Inbound
translate called 1
!
voice translation-profile Outbound
translate calling 2

 

Having dial peer as below:

 

dial-peer voice 100 voip
translation-profile incoming Inbound
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 022799232
dtmf-relay rtp-nte
!
dial-peer voice 201 voip
description National
translation-profile outgoing Outbound
destination-pattern 02.......
session protocol sipv2
session target sip-server
session transport udp
voice-class sip profiles 1
dtmf-relay rtp-nte sip-kpml
no vad

When I Call in dial peer 201 is selected and calls are not translated or rings two times and no more.

SO I am not clear how to workaround the situation.

Any idea on organizing?

So I have the same number for incoming but it goes to outgoing dial-peer.

I do not understand how.

Please any idea how to organize things.

26 Replies 26

I think so, newer done it myself, so can’t give you definite answer. Try it out and check the result.



Response Signature


You still have “session target sip-server” on your inbound dial peer. As this is used in the outbound direction it has no place or use on an inbound dial peer. Please remove it.



Response Signature


Thanks very much your comments are very valuable.

 

the thing is

having that router on NAT

Incoming calls are ok but on outgoing only one-way audio.

I have crreated a loopback interface with the public IP.

AI have read on voip service sip there is not necessasry put the bind interface.

So the questions is putting the loopback interface  (public ip) on outgoing dial peers would be the solution to tackle one-way audio?

Or you think better crreate voice tenants?

Best regrds and thanks

The common way to configure CUBE these days, is using tenants.

So put everything related to the SIP connection to the provider in a tenant and bind it to the dial-peers, which you use in/out to/from provider.

Also don't forget to bind the signalling and media to the interfaces (voice-class sip bind media/signalling source-interface ...).

For the CME part, put it under voice service voip and for the provider side, put it under the tenant or the corresponding dial-peers.

 

--- Please rate this post as "Helpful" or accept as a solution, if your question has been answered ---

So phisical interface (with private ip ) under viuce serivce voip

and loopback interface (with public ip in the dial peer)?

 

Thanks

Hi, 

 

The loopback address is one way to do it, but for me it's a little bit of a hack, especially if that public IP address is already defined on an upstream Internet gateway (if I have understood your setup correctly). It can certainly be done though. 

 

If you don't have a public IP on your CME, you may need to configure some SIP profiles to modify some of the SIP headers from your private to your public IP address. If your upstream Internet gateway has SIP-ALG or SIP Inspection enabled, then it's probably modifying these SIP headers already. If it's not, then review the Outbound Layer 7 Fixup section of this document for a config example:

 

https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In-Depth-Explanation-of-Cisco-IOS-and-IO.html

 

I always push the customers to a scenario, where we have a public IP directly on the CUBE. Everything else just makes troubles.

So, I would recommend using 2 physical interfaces. 1 towards internal network and 1 with public IP towards internet.

And on the interface with the public IP you should configure an access-list to only allow traffic (deny everything else) from and to the provider. Normally, that isn't so complicated, if you have the info from the provider, which IP's / ports / UDP or TCP he uses for signalling and media.

Yes Scott.

the public IP is unused (its from the block that I have received from ISP) and I have nated private to public. and the loopback has public IP  (not the gateway but that of nat rule.).

 

You mean better have the gateway IP?

Yes, it’s better to use the gateways public IP if it has one and stay clear of any address translation. It could cause more trouble then what it solves, especially if you don’t know what to do to handle it properly.



Response Signature


Hi,

 

Not sure if attaching your unused public IP to a loopback address would work. I suppose it would if you used the "IP unnumbered" command to your physical interface, but why would you do that if you could just configure the IP directly on the interface?

 

So essentially, it sounds like you have two choices:

 

1. Put the public IP directly onto a physical interface on your CME. The config will be much simpler and cleaner, but I appreciate it represents a small topology change on your side as you're bringing the public element of your network onto the CME. However as pointed out, ACLs applied inbound which restrict it to only your ITSP Signaling and Media endpoints will keep it secure. You may also need to evaluate if you can actually make that physical connection to your ISPs equipment. Is it on a separate connection or is it a different public IP range attached to your existing Internet service?

2. Keep the topology as it is, where your CME is behind your Firewall. Configure a static NAT on your Firewall translating the private IP to your new public IP. If you're Firewall is doing SIP Inspection, it should be rewriting the private IPs in the SIP headers to your public IP. If your Firewall doesn't do SIP Inspection, then you'll need SIP profiles configured on the CUBE to do that.

 

In short, do option 1 if you can, option 2 if you can't!

Scott Leport
Level 7
Level 7

Hi,

 

If your PSTN gateway and CME are one and the same, make the changes Roger suggested and that should sort your inbound calling issue.

Regarding your outbound calling, I see you have the session target sip-server command defined. What is the sip-server you have configured and where is it placed? sip-ua config, voice class tenant etc?

In addition to the URI config Roger suggested, which will be applied to your inbound dial-peer for incoming calls, it may also be advisable to configure a session server-group which reference your ITSP Signaling endpoints for your outbound calls and reference that in your outgoing PSTN facing dial-peers instead of the sip-server config. To expand on that, I mean this:

 

voice-class server-group 100

 description ***ITSP Signaling Servers**

  ipv4 <sbc1>

  ipv4 <sbc2>   <----- Add as many as required


dial-peer voice 201 voip
description National
translation-profile outgoing Outbound
destination-pattern 02.......
session protocol sipv2
session server-group 100
session transport udp
voice-class sip profiles 1
dtmf-relay rtp-nte sip-kpml
no vad

Yes it is defined on SIP-UA.

BUT the ITSP is offering some subnets and adding one by one is difficult als dns-pattern is not the same? any idea on this?

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