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Cisco 2811 Voice Gateway

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snoop152

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Hi all,

can anyone provide a working example for a connection between a 3cx system and a Cisco 2811 (IOS 124.22T) router with PRI ?

My config example:

dial-peer voice 20000 voip
description xxxx
destination-pattern xxxx..
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.60.100
dtmf-relay rtp-nte
no vad

sip-ua
credentials username 20000 password xxxxxxx realm 3CXPhoneSystem
authentication username 20000 password xxxxxx retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers expires 300000
registrar ipv4:192.168.60.100 expires 300
sip-server ipv4:192.168.60.100:5060

i have an outbound pots peer aswell

unfortunately, the router get´s blacklisted on the 3cx system...

Kind regards

s.l.
 
I am very interested in the same thing, there is another post about a similar issue with an E1. I hope someone chimes in to help out.
 
Here is the link to the other discussion

http://www.3cx.com/forums/3cx-and-cisco-router-20142.html#p101722
 
I found this post from a while back...let me know if this helps you. If it does I would love a copy of your config so I can test this out when my new router shows up.

I have a Cisco 2621XM router and have successfully configured it so that it acts as my VoIP gateway.

Hardware configuration is as follows:

NM-HD-2V with 1 2-port VIC2-2FXO card
IOS version: c2600-ipvoicek9-mz.124-9.T5.bin

Here's the necessary configuration to get things working on the Cisco router with 3CX PBX.

The code below tells the router what kind of VoIP service to use, which is set to sip

voice service voip
sip

Next, define the codecs you wish to you in order of preference.

voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8 bytes 40
codec preference 3 g723r63 bytes 96
codec preference 4 g726r16 bytes 80

Next configure your voice ports. This example is only for the first voice port. The last line is the most important as it tell the port what extension to dial on inbound call. The opx parameter tells the router that the extension is an off premise extension.

voice-port 1/0/0
no battery-reversal
timing hookflash-out 50
connection plar opx 999

Next program your dial-peers. Pick any arbitrary number for your dial-peers but they should make some sense to you. I've chosen 20000 for my voip dial-peer, which is used for inbound calls. If you have more than one voip dial-peer, set the preference order.

dial-peer voice 20000 voip
preference 1
destination-pattern 99.
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.253.5 --> address of 3CX server
dtmf-relay rtp-nte
no vad

Next program your POTS dial-peer.

dial-peer voice 10000 pots
preference 1 --> use if you have multiple POTS dial-peers
destination-pattern .T
port 1/0/0 --> matches your dial-peer with physcial port

Next program the SIP user agent parameters. The "authentication" parameter is used to authenticate the Cisco router against the gateway you programmed in the 3CX server.

sip-ua
authentication username 10000 password 091D1E594955
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers expires 300000
registrar ipv4:192.168.253.5 expires 300
sip-server ipv4:192.168.253.5:5060

The above should work for any Cisco router that supports VoIP.
 
I got some friends to send me some configs....let me know if this helps:

voice call send-alert
!
voice service voip
address-hiding
allow-connections sip to sip
redirect ip2ip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
h323
sip
header-passing error-passthru
asserted-id pai
midcall-signaling passthru
privacy-policy passthru
g729 annexb-all
!
!
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
voice translation-rule 1
rule 1 /^91\(.*\)/ /\1/
rule 2 /^9\(.*\)/ /\1/

voice translation-profile DIGITSTRIP_9
translate called 1

dial-peer voice 100 voip
description Inbound SIP Dial-Peer for XXXXXXXXXX DIDs to Publisher from ATT
preference 2
destination-pattern ..........
rtp payload-type nse 99
rtp payload-type nte 100
voice-class codec 1
voice-class sip profiles 1
session protocol sipv2
session target ipv4:10.x.x.x
dtmf-relay rtp-nte
fax-relay sg3-to-g3
!
dial-peer voice 101 voip
description Inbound SIP Dial-Peer for XXXXXXXXXX DIDs to Subscriber01 from A
preference 1
destination-pattern ..........
rtp payload-type nse 99
rtp payload-type nte 100
voice-class codec 1
voice-class sip profiles 1
session protocol sipv2
session target ipv4:10.x.x.x
dtmf-relay rtp-nte
fax-relay sg3-to-g3
!
dial-peer voice 200 voip
description Outbound SIP Dial-Peer for National Dialing to ATT Primary
translation-profile outgoing DIGITSTRIP_9
preference 1
destination-pattern 9[2-9]..[2-9]......
rtp payload-type nse 99
rtp payload-type nte 100
voice-class codec 1
voice-class sip privacy-policy passthru
voice-class sip early-offer forced
voice-class sip profiles 1
session protocol sipv2
session target ipv4:12.x.x.x
incoming called-number .
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400
!
dial-peer voice 201 voip
description Outbound SIP Dial-Peer for National Dialing to ATT Secondary
translation-profile outgoing DIGITSTRIP_9
preference 2
destination-pattern 9[2-9]..[2-9]......
rtp payload-type nse 99
rtp payload-type nte 100
voice-class codec 1
voice-class sip privacy-policy passthru
voice-class sip early-offer forced
voice-class sip profiles 1
session protocol sipv2
session target ipv4:12.x.x.x
incoming called-number .
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400
!
dial-peer voice 202 voip
description Outbound SIP Dial-Peer for LongDistance Dialing to ATT Primary
translation-profile outgoing DIGITSTRIP_9
preference 1
destination-pattern 91[2-9]..[2-9]......
rtp payload-type nse 99
rtp payload-type nte 100
voice-class codec 1
voice-class sip privacy-policy passthru
voice-class sip early-offer forced
voice-class sip profiles 1
session protocol sipv2
session target ipv4:12.x.x.x
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400
!
dial-peer voice 203 voip
description Outbound SIP Dial-Peer for LongDistance Dialing to ATT Secondary
translation-profile outgoing DIGITSTRIP_9
preference 2
destination-pattern 91[2-9]..[2-9]......
rtp payload-type nse 99
rtp payload-type nte 100
voice-class codec 1
voice-class sip privacy-policy passthru
voice-class sip early-offer forced
voice-class sip profiles 1
session protocol sipv2
session target ipv4:12.x.x.x
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400
!
dial-peer voice 204 voip
description Outbound SIP Dial-Peer for InterNational Dialing to ATT Primary
translation-profile outgoing DIGITSTRIP_9
preference 1
destination-pattern 9011T
rtp payload-type nse 99
rtp payload-type nte 100
voice-class codec 1
voice-class sip early-offer forced
voice-class sip profiles 1
session protocol sipv2
session target ipv4:12.x.x.x
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400
!
dial-peer voice 205 voip
description Outbound SIP Dial-Peer for InterNational Dialing to ATT Secondar
translation-profile outgoing DIGITSTRIP_9
preference 2
destination-pattern 9011T
rtp payload-type nse 99
rtp payload-type nte 100
voice-class codec 1
voice-class sip early-offer forced
voice-class sip profiles 1
session protocol sipv2
session target ipv4:12.x.x.x
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400
!
dial-peer voice 206 voip
description Outbound SIP Dial-Peer for Emergency Dialing to ATT Primary
translation-profile outgoing DIGITSTRIP_9
preference 1
destination-pattern 911
rtp payload-type nse 99
rtp payload-type nte 100
voice-class codec 1
voice-class sip early-offer forced
voice-class sip profiles 1
session protocol sipv2
session target ipv4:12.x.x.x
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400
!
dial-peer voice 207 voip
description Outbound SIP Dial-Peer for Emergency Dialing to ATT Secondary
translation-profile outgoing DIGITSTRIP_9
preference 2
destination-pattern 911
rtp payload-type nse 99
rtp payload-type nte 100
voice-class codec 1
voice-class sip early-offer forced
voice-class sip profiles 1
session protocol sipv2
session target ipv4:12.x.x.x
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400
!
dial-peer voice 208 voip
description Outbound SIP Dial-Peer for Information Dialing to ATT Primary
translation-profile outgoing DIGITSTRIP_9
preference 1
destination-pattern 9411
rtp payload-type nse 99
rtp payload-type nte 100
voice-class codec 1
voice-class sip early-offer forced
voice-class sip profiles 1
session protocol sipv2
session target ipv4:12.x.x.x
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400
!
dial-peer voice 209 voip
description Outbound SIP Dial-Peer for Emergency Information to ATT Secondar
translation-profile outgoing DIGITSTRIP_9
preference 2
destination-pattern 9411
rtp payload-type nse 99
rtp payload-type nte 100
voice-class codec 1
voice-class sip early-offer forced
voice-class sip profiles 1
session protocol sipv2
session target ipv4:12.x.x.x
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400
!
!
sip-ua
no remote-party-id
retry invite 2
 
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