01-29-2015 01:32 PM - edited 03-17-2019 01:46 AM
Greetings,
I have a question in regards to the best practice when it comes to creating inbound dial-peers for a CUBE that would be receiving SIP calls from various sources (CUCM, CME, two SIP providers). Would it be best to create a dial-peer to catch call ex.
dial-peer voice 1 voip
description ***INCOMING***
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 1
dtmf-relay rtp-nte
no vad
or create one for each incoming call leg?
dial-peer voice 1 voip
description ***INCOMING Call-leg 1 PSTN***
session protocol sipv2
session target sip-server
incoming called-number 21XXX
voice-class codec 1
dtmf-relay rtp-nte
no vad
dial-peer voice 2 voip
description ***INCOMING Call-leg 2 PSTN 2***
session protocol sipv2
session target sip-server
incoming called-number 85XXX
voice-class codec 1
dtmf-relay rtp-nte
no vad
dial-peer voice 3 voip
description ***INCOMING Call-leg 3 CME***
session protocol sipv2
session target sip-server
incoming called-number 31XXX
voice-class codec 1
dtmf-relay rtp-nte
no vad
dial-peer voice 4 voip
description ***INCOMING Call-leg 4 CUCM***
session protocol sipv2
session target sip-server
incoming called-number 1212233XXXX
voice-class codec 1
dtmf-relay rtp-nte
no vad
I've always done an incoming with a catch all with a codec profile that would always prefer G711, but have other's as a viable alternative. And if a codec needed to be statically defined, it would be done on the outbound dial-peer, along with bindings for their respective outgoing interfaces. Please advise.
01-29-2015 01:45 PM
I think the best practice would be what you have shown at the bottom. This allows you to separate the inbound and outbound peers in both directions. This may be necessary in order avoid matching the wrong dial peer since you are using VOIP in both directions. It also aids in troubleshooting, and if down the road you need to tweak a PSTN side dial peer but not a CUCM side dial peer it's easy to do.
01-29-2015 02:09 PM
So even for numbers that are meant for outbound calls to the PSTN, I'd be creating 2 dial-peers?
01-30-2015 07:12 AM
So every call needs an inbound dial-peer and an outbound dial-peer. Typically the dial-peers would be grouped like this:
1: Inbound dial-peer - Facing PSTN
2. Outbound dial-peer - Facing PSTN
3. Inbound dial-peer - Facing CUCM/CME
4. Outbound dial-peer - Facing CUCM/CME
So the call flow from the PSTN would be 1 -> 4
Outbound flow is 3 -> 2
Within each group you may have multiple dial peers due to multiple CUCMs, multiple dial strings that can't be summarized, etc.
02-01-2015 03:52 AM
If your matching incoming calls from 3 different system, its best that you match incoming calls by URI.
for example.
CUCM = X.X.X.X
CME = Y.Y.Y.Y
PSTN = Z.Z.Z.Z
voice class 101 sip
host ipv4:X.X.X.X
voice class 102 sip
host ipv4:Y.Y.Y.Y
voice class 103 sip
host ipv4:Z.Z.Z.Z
dial-peer voice 1 voip
description **From ITSP**
incoming uri via 103
Dial-peer voice 2 voip
description **FROM CUCM**
incoming uri via 101
Dial-peer voice 3 voip
description **FROM CME**
incoming uri via 102
Alternatively you can match CUCM and CME incoming calls on the same Dial-peer by both there ip address in a single voice class.
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