New to Cisco, what's a Call Leg? What's a dial-peer? and How can I setup an FXO to a Cicso CM 7?

[AvayaRedDude]

All,

I am new to Cisco (hence my handle, I come from experience of the TDM world) and I have no interest of paying Cisco for a useless CCNA cert. I'd like to ask the following questions:

I am interested in putting an FXO trunk on a Cisco Router (yes it's also a voice router w/ VICs) and route that to a full fledged Cisco CallManager R7.

From what I read, there are two things that I believe I have to a dial peer and call legs. What are they, how are they similar to the Avaya world and also can someone explain how can I set this up? The Web is polluted with white papers, and what-could-you-do-with-these instead of just explaining to me how to set it up. Also please don't refer me to other sites you lazy bums

Thanks in Advance!

 

[gnrslash4life]

What is the routing running, MGCP, H.323, or SIP?

Certifications:
A+
Network+
CCENT
CCNA Voice

 

[AvayaRedDude]

H323

 

[mikeyb123]

This is a really good article for you - taken from this site:

http://www.voip-info.org/wiki/view/Asterisk+cisco+FXO

Another Take on Caller ID

I may have inadvertently backed into a solution on this issue. I previously had figured out how to make the VIC-2FXO (not VIC-2FXO-M1) work with CallManager 4.1. The issue there was that CLID didn't work with MGCP, but research lead me to a solution that did work for that card with H.323. So, when I converted these ports to SIP for Asterisk, I found that they passed caller ID just fine! Here is the relevant configuration:

voice-port 1/0/0
output attenuation -1
no comfort-noise
connection plar opx 7145551212 - this is the number configured in the call mangler
description 71455551212
caller-id enable

dial-peer voice 1 pots
incoming called-number .
destination-pattern .
port 1/0/0

dial-peer voice 100 voip
preference 1
incoming called-number .
destination-pattern 7145551212
progress_ind setup enable 3
session protocol sipv2
session target ipv4:x.x.x.x:5060
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
no vad

sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:x.x.x.x

So the session target and sip server are the ipaddress of your call mangler - in this case it's a SIP connection not a H323, but coming from a TDM background you should be able to work the rest out..

helped me sort out a call mangler - no doubt this post will be pulled apart by the cisco kiddies ;-0 but it worked for me


It's not getting any smarter out there. You have to come to terms with stupidity, and make it work for you.

 

[AvayaRedDude]

^
I'd suggest against SIP in this environment, I really need this as reliable as possible, and SIP won't cut it.

I also have VIC-2FXO-M1, is this incompatible for my setup?

You dodged my question, what is a call leg, what's a dial-peer? how can I add an FXO trunk, add a line group, (got the idea of the ring group) Cisco doesn't use TAC (trunk access), so how can I emulate that type of setup with my CUCM?

Through answers are gratefully appreciated.

 

[AvayaRedDude]

I don't know why I am responding to that "answer" because that guy looked like he drunk typed (as that response came at 4am (ET) this morning to me).

 

[mikeyb123]

Call Leg and Dial Peers.

Suggest you look here

http://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/12164-dialpeer-call-leg.html

It's got a pretty picture that you can look at.

drunk typing? its 14:51 here in the UK so it's not even time for a drink.

Good Luck



It's not getting any smarter out there. You have to come to terms with stupidity, and make it work for you.

 

[gnrslash4life]

Mikeyb123 explained it pretty well actually. But to hopefully help here are H.323 dial peers.

dial-peer voice 1 pots
description Incoming POTS Calls
incoming called-number .
direct-inward-dial


dial-peer voice 2 pots
description For OutBound POTS Calls
preference 1
destination-pattern 9.T
progress_ind setup enable 3
progress_ind progress enable 8
port 0/0/0:23


dial-peer voice 101 voip
description For Incoming POTS Calls
preference 1
destination-pattern 123.......
progress_ind setup enable 3
progress_ind progress enable 8
progress_ind connect enable 8
session target ipv4:1.2.3.4
voice-class codec 1
voice-class h323 1
dtmf-relay h245-alphanumeric
fax-relay ecm disable
fax-relay sg3-to-g3
fax rate 14400
no vad

Similar to SIP dial peers.

Also, your FXO card will be used for things like POTS lines or overhead paging. Things that do not require dial tone created for them. FXS ports provide dial tone. FXO do not.

Certifications:
A+
Network+
CCENT
CCNA Voice

 

[AvayaRedDude]

I do know what an FXO is supposed to be used for but the "dial-peer" term is still making me confused. What does that term supposed to mean?

(I've used Avayas for so long I forgot how bad, the IT world is with vague words and engineer-speak.)

 

[mikeyb123]

definition of a dial peer

A dial peer, also known as an addressable call endpoint, is a device that can originate or receive a call in a telephone network. In voice over IP (VoIP), addressable call endpoints can be categorized as either voice-network dial peers or POTS (plain old telephone service) dial peers. Voice-network dial peers include VoIP-capable computers, routers, and gateways within a network. POTS dial peers include traditional telephone network devices such as phone sets, cell phones, and fax machines.


The term dial peer is sometimes used in reference to a program that matches a specific dialed sequence of digits to an addressable call endpoint. According to this definition, there is one dial peer for each call leg (connection between two addressable call endpoints).

It's not getting any smarter out there. You have to come to terms with stupidity, and make it work for you.