Configuring the Synapse SB67070 SIP Gateway from AT&T for Clearfly SIP Trunking. January 2013
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1 Configuring the Synapse SB67070 SIP Gateway from AT&T for Clearfly SIP Trunking January 2013
2 1 Introduction This guide was created to assist Synapse partners with configuring the Synapse SB67070 SIP Gateway with Clearfly s SIP trunking services. It provides example data for the required fields. The actual account settings are provided by Clearfly when service is activated. Questions about software and hardware installation or other PBX configuration issues should be directed to AT&T Synapse technical support at or refer to the AT&T Synapse Administration Guide on our website 2 System and Network Components Required for SIP Trunking Service An AT&T Synapse SB67070 SIP Gateway. Internet connection with at least 100kb/s of download and upload bandwidth per line of service. Synapse firmware version 3.0 or higher installed on all Synapse devices. A Clearfly SIP trunk account. Note: AT&T recommends all hardware be updated to the latest software version available, and all information be gathered before programming. 3 Limitations and Known Issues SIP trunk faxing is not supported. We recommend if you require fax support to use an analog line or other efax services. For complete fax setup with analog lines, please refer to the AT&T Synapse Administration Guide found on our website. Static registration is not supported at this time. During an incoming call if a Synapse phone user hangs up before the originating caller hangs up, the line will still be active and in use. Until Clearfly receives a disconnect signal from the originating caller or 5 minutes pass without the originating caller hanging up the line will be active and in use. In most cases the originating caller will hang up within seconds of the Synapse users. Clearfly also blocks any kind of robocalling where the far end may not hang up. For other limitations of the AT&T Synapse SB67070 SIP Gateway, please refer to the Synapse Administration Guide on our website. 2
3 4 Clearfly Account Setup When setting up an account with Clearfly, please request a registered or dynamic account when speaking to a Clearfly representative. This type of account will allow to you register at any location and with any IP address and requires a username and password, which Clearfly will generate for you. After the account has been set up with the number of lines and DIDs needed, Clearfly will send you the account information, which will look something like this: Clearfly SIP Server/Domain: sc.cfly.co Username: Password: Transport: Codec: md15quam UDP/5060 G.711u RTP Payload Type: 101 DTMF Relay: RFC2833 Digits Transferred: Default is 10. DIDs: The SIP Server/Domain, Username, Password, and DIDs are the only settings we will need to set up the Clearfly account in the AT&T Synapse SB67070 SIP Gateway. The other information is not needed because the gateway already knows which settings to use, and there are no fields to enter this data. 4.1 Configuring the SIP Account To configure the SIP Account: 1. Log in to a Synapse device on the network. 2. Navigate to the SIP Settings page. 3. Enter the Basic Settings: a. Enable the account and fill out all fields except Primary Number. b. Enter a name for this SIP account in the Account Name field. The account name is not provided by Clearfly; you just make one up. 3
4 c. Enter the max number of lines you ordered with your account in the Max Calls field. d. Enter the caller ID name you wish you transmit in the Display Name field. e. Enter the username Clearfly gave you in the User Name and Auth User Name field. Both fields need to have the Clearfly Username in them. f. Enter the password Clearfly gave you in the Auth User Password field. 4. Leave the Registration Settings at the default values. 5. Under Server Settings, enter the SIP Server/Domain that Clearfly gave you into the SIP Server Address or URL field. 6. Leave the Codec Configuration at default. 4
5 7. Click Apply at the bottom of the page to save the settings and activate the account. The page will refresh and the Registration Status should change to Registered. 4.2 DID Configuration 1. Navigate to the Direct Inward Dial page. 2. Enter your primary number into the System Pilot Number field and add the range of DIDs that Clearfly provided to the Add Direct Inward Dial Range fields. 3. Click Add next to the DID range for each separate range, and then click Apply at the bottom of the page to save all the settings. 4. Next, navigate to the DID Assignments page to point all DIDs to whatever locations in the system you wish. For more information about DID setup, please refer to the AT&T Synapse Administration Guide on our website. 5
6 5 Troubleshooting Symptom Registration Status does not change to Registered. Suggested Action Check to make sure you can ping the SIP server from a PC on the same network as the phones. If you can ping the server, but are still not registering, please verify your account information. If the account information is correct you may need to set up port forwarding for SIP and RTP. SIP runs on port 5060 and should be forwarded to the AT&T Synapse SB67070 SIP Gateway. RTP runs on ports , which should also be forwarded to the AT&T Synapse SB67070 SIP Gateway. If you are not sure how to port forward these ports on the router, please check the router s manual or go to If the ports are forwarded and you are still not registering, please contact Clearfly at to verify your account information and to verify they can reach the AT&T Synapse SB67070 SIP Gateway. You can also contact AT&T Synapse Technical Support at for further assistance. The Desksets display All lines are busy. One way audio issues. DIDs are not routing properly or phones are not receiving calls. All the lines are in use at the time you are attempting to call out, or the gateway is not registered. If you know the lines are not in use, log in to the Synapse WebUI and check the status of the SIP account. If it is not registered, check to make sure you have Internet access by going to a web page on your PC. If you have Internet access and the account still will not register, please contact Clearfly at to check for outages. If there are no outages or service issues, contact AT&T Synapse Technical Support at for further assistance. If you are experiencing one way audio on all calls, please port forward ports 5060 and to the AT&T Synapse SB67070 SIP Gateway. If you are not sure how to port forward these ports on the router, please check the routers manual or go to If you are still experiencing one way audio after port forward, please contact Clearfly at for further assistance. If you are experiencing any issues with the DIDs routing properly, please verify that the DIDs are programmed without any dots or dashes in the numbers and that they are pointing to the correct location. If you are still experiencing DID issues, please contact Clearfly at to check for any problems with the phone numbers. 6
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Note: these functions are available if service provider supports them.
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nexvortex SIP Trunking
nexvortex SIP Trunking January 2015 510 SPRING STREET HERNDON VA 20170 +1 855.639.8888 Copyright nexvortex 2014 This document is the exclusive property of nexvortex, Inc. and no part may be disclosed,