• ABOUT
  • RESOURCES
    • Network Map
    • Rate Centers & LATAs
    • CPNI
    • Docs
    • Product Roadmap
  • SOLUTIONS.
    • E911
    • CNAM
    • PBX
    • Colocation
    • Call Routing LRN
    • LATA and Switch Data
    • Global Phone numbers
  • PRICING
  • DEVELOPERS
  • COVERAGE
  • CONTACT
  • LOGIN
  • SIGNUP
  • EN
    • EN
    • ID
    • RO
    • ES
    • PT
    • RU
    • IT

UNLIMITED

$0.10
Please enter a contact.
Dialing... Incoming call Answering, connectivity checks in progress... Call connected

1

2

ABC

3

DEF

4

GHI

5

JKL

6

MNO

7

PQRS

8

TUV

9

WXYZ

*

0

+

#

ERROR: This service requires JavaScript. Please enable JavaScript in your web browser settings. ERROR: This service requires WebRTC. Please try Mozilla Firefox or Google Chrome, using the latest version is strongly recommended. ERROR: JsCommunicator configuration not found ERROR: Failed to initialize user agent ERROR: SIP Registration failure ERROR: failed to start call, check that microphone/webcam are connected, check browser security settings, peer may not support compatible codecs
WebSocket link: Connected Disconnected
SIP registration: Registered Not Registered
September 04, 2019

Configuring call termination via Cisco gateways

This article explains how you should configure your Cisco CME (or CUBE) gateway for sending calls to our system

Step-by-step guide

First of all, obtain your SIP Username and Password from MultiTEL.

    You can create a pair of SIP Username/Password in our Trunks section:
    https://multitel.net/trunks/accounts


    You can also add your IP address here:
    https://multitel.net/trunks/ips

    1. Login to your Cisco voice gateway and use a configuration similar to the one below (modify where needed)
      Once finished, test your calls from the Cisco CLI console using the following command:

      csim start <enter_here_your_phone_number>

      For example, if your phone number was 18886868581, this CLI command would show like this

      csim start 18886868581

      This command would originate a phone call from your Cisco voice gateway through MultiTEL's termination service and would be used for you to test your config.

      In addition to that, you can always use the following two commands:

      show call active voice brief
      show call history voice brief

      These would tell you which dial-peers were attempted (search for the pid: keyword) and why calls failed.
    #This section will vary depending on your configuration.
    #We do suggest setting the variable min-se which controls session timeout to 900 seconds.

    voice rtp send-recv
    voice service voip
    !
      allow-connections sip to sip
    sip
        min-se 900
    !
    voice class codec 711
     codec preference 1 g711ulaw
     codec preference 2 g711alaw
    !
    voice class codec 729
    codec preference 1 g729r8 bytes 40
    codec preference 2 g729br8 bytes 40
    !
    voice class codec 700
    codec preference 1 g711ulaw
    codec preference 2 g711alaw
    codec preference 3 g729r8 bytes 40
    codec preference 4 g729br8 bytes 40
    !
    ! We support E164 format only. US/CA numbers have to be dialed in full 11 digit format.
    ! Use this template if you wish to convert from 7 or 10 to 11 digits.
    ! Replace NPA with your local area code when using 7 digit numbers
    !
    ! translation rule definition below
    !
    voice translation-rule 1
      rule 1 /^9\(1[2-9]..[2-9]......\)$/ /\1/     ! send out if using 9 dial-out prefix and 11 digit format ; strip 9 
      rule 2 /^9\([2-9]......\)$/ /1NPA\1/         ! send out if using 9 dial-out prefix and 10 digit format ; strip 9 and 1 country code
      rule 3 /^\(.......\)$/ /1NPA\1/              ! 7 digit format, add 1 and the "NPA" and send out (don't forget to replace NPA with your local prefix)
    rule 4 /^011\(.*)$/ /1\1/ ! strip 011, send call out
    rule 5 /^00\(.*)$/ /1\1/ ! strip 00 , send call out

    !
    ! translation profile using the translation rules above

    !

    voice translation-profile multitel
    translate called 1
    !
    ! Outbound dial-peer definition
    !

    dial-peer voice 1 voip
    permission term
    description Termination towards MultiTEL translation-profile outgoing multitel destination-pattern .T ! or replace with 1[2-9]..[2-9]......T if only sending US calls session protocol sipv2 session target dns:sbc-us.multitel.net ! -- replace here with the SBC which is closest to you
     session transport udp  ! or "tcp", depending how your account is configured on MultiTEL
    voice-class codec 711 dtmf-relay rtp-nte no vad
    !
    ! Inbound dial-peer definition
    !
    dial-peer voice 2 voip
    permission orig
    description Origination from MultiTEL
    huntstop
    session transport udp ! or "tcp", depending how your account is configured on MultiTEL
    incoming called-number .T
    dtmf-relay rtp-nte
    voice-class codec 711
    fax-relay ecm disable
    fax-rate 9600
    fax protocol t38 ls-redundancy 1 hs-redundancy 1 fallback pass-through g711ulaw
    no vad
    !

    !
    ! IMPORTANT: please make sure you DO HAVE the below config for security purposes
    ! this will allow calls to your gateway only from authorized IP addresses
    voice source-group myawesomecompany
    access-list 99
    disconnect-cause call-reject
    ! add here inbound translation profile if you need to
    !
    ! access-list definition - list here your IP addresses
    ! default action is deny
    !
    access-list 99 permit 192.168.1.0 0.0.0.255
    ! You may have to add MultiTEL's origination IP addresses here to allow inbound DID/origination calls from MultiTEL
    ! Please consult the article below for complete (and usually up to date) list of IP addresses we use to send DID/origination calls
    ! https://multitel.net/docs/view/161/knowledgebase/
    ! Sample entries below:
    access-list 99 permit X.Y.Z.0 0.0.0.255
    access-list 99 permit A.B.C.0 0.0.0.255




    Similar Articles

    February 09, 2015

    Required documents for certain countries

    READ MORE
    February 09, 2015

    STUN Servers

    READ MORE
    March 05, 2015

    What is Toll Fraud

    READ MORE
    April 14, 2015

    Linksys LRT224 recommended settings for VoIP

    READ MORE
    April 15, 2015

    Recommended settings for SonicWall products for SIP protocol

    READ MORE
    • COMPANY
    • PRICING
    • DEVELOPERS
    • CONTACT
    • SOLUTIONS
    • E911
    • CNAM
    • PBX
    • Call Routing LRN
    • LATA and Switch Data
    • Global Phone numbers
    • COVERAGE
    • International Numbers
    • Portability
    • Voice and SMS
    ANYWHERE, ANYTIME

    Hassle-free cost-effective
    communications & collaboration solutions

    SIGNUP NOW
    MultiTEL BBB Business Review
    SecurityMetrics PCI validation certification logo

    2025. MultiTEL LLC. All rights reserved

    • Terms of Service
    • Privacy Policy
    • Support
    • Contact