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Cisco 2811 Voice Gateway

Discussion in '3CX Phone System - General' started by snoop152, May 10, 2011.

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  1. 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.
     
  2. netswork

    netswork Active Member

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    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.
     
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  3. netswork

    netswork Active Member

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    Here is the link to the other discussion

    http://www.3cx.com/forums/3cx-and-cisco-router-20142.html#p101722
     
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  4. netswork

    netswork Active Member

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    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.
     
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  5. netswork

    netswork Active Member

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    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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