Mta 6328 2re Firmware Download

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Device Configuration Guides
InnoMedia MTA 6328Re (used by many service providers including SunRocket)

Note: To find the password and instructions to access the configuration settings, check with your service provider or search 'smithonvoip' in Google.

The standalone InnoMedia MTA 6328Re is compatible with any standard analog telephone set, the MTA 6328Re delivers voice quality and features equivalent or superior to those of PSTN. Its versatile and open system interfaces provide the flexibility to work with many different networks (HFC cable, ADSL, fiber, wireless) and broadband access devices. The MTA 6328Re allows users to share their broadband connection throughout their home network by either connecting a PC or a hub into the MTA downlink port. Its data rate limiting feature ensures voice quality during phone calls by automatically throttling down data throughput and reserving bandwidth for voice whenever a call is in progress.


  1. 1 Open your web browser and connect to your MTA (See Logging In on page 9 for more details). 2 Click VoIP and then Config File. 3 To save the current configuration, click the Download button.
  2. Download Visio Stencils for InnoMedia. Home; Product & Solutions. Visio Diagramming. MTA 6328-24 Visio Stencil. MTA 6328-2Re Visio Stencil. MTA 6328-4 Visio Stencil. MTA 6328-8 Visio Stencil. Back to Manufacturer List. Partnerships Hardware Partnerships.

Innomedia MTA 6328-2re Voip 2 Lines Adaptor Visit the Lenovo Store. Currently unavailable. We don't know when or if this item will be back in stock.

STEP 1
Turn off provisioning if you have not done this already. To do this, go to NETWORK -> Provisioning setting. Uncheck the 'Enable Provisioning' box and hit the 'Save and Reboot' button. After rebooting, you may not be able to login to the admin account since the reboot will default to index.htm. Hit CANCEL, and go to ยป192.168.251.1/Voice_adminPage.htm to login.

STEP 2
Login with the required password.

STEP 3
Go to VOIP -> SIP Proxy and enter these settings:

  1. SIP Proxy - sip.inphonex.com
  2. Uncheck 'Use Outbound Proxy' box
  3. SIP Local Signaling Port: 5065
  4. SIP Domain: sip.inphonex.com
  5. Registration Expiration time: 60
  6. Preferred Codecs List: G729a
  7. Choose 'NONE' for all others.
  8. Click 'Save' button.

STEP 4
Go to VOIP -> User Account.

  1. Choose Line 1 from dropdown
  2. User ID: Your 7 digit InPhonex Virtual Phone Number
  3. Password: SIP password that you set (by default this is the same password you use to login to the InPhonex On-line Control Panel
  4. User Name: Same as User ID above
  5. Authentication ID: Leave this field Blank
  6. Use Hot Phone Number: NO
  7. Hot Phone Number:
  8. Use T38 FAX: Yes
  9. Click 'Save' button

STEP 5
Reboot your cable/dsl modem, then reboot your device, and power-cycle your phone (if it is a powered phone). Check under the 'information' to see if line 1 is registered.

STEP 6
If your phone shows as registered, pick up the phone and make calls using InPhonex!

Note: Innomedia users must dial 7288888 to access the Voicemail System.

Hi,
I am fairly new to SS. I am trying to setup GV-Gizmo5-SS and am unable to get it working correctly on my innomedia ATA
however fring works just fine from the same Access point with the same SS account.
i am running
Image Version : V3.0.38
BBS Version : V7.2.8
Controller Code Version : V3.10.37 SIPMTA 6328-R Nov 16 2005 19:46:23Firmware
DSP Code Version : V2.4.11 11/14 22:09 2005
SIP Stack Version : V2.9.100
on my ATA,
these are the settings on Innomedia for sip
SIP_Settings> p
Current SIP Proxy Servers = 69.59.142.213:2060
Use Outbound Proxy = No
Current Local SIP Port = 5060
Response Code for Retry Registration =
Retry Registration Interval = 60 seconds
Current SIP Domain =
Current Exponential Backoff = 500 ms
Current Exponential Cap = 2000 ms
Current Non-INVITE retry = 4 times
Current INVITE msg retry = 4 times
Current REGISTER expiration = 600 seconds
Current Session Timer = 0 seconds
Current Bullet Interval = 0 seconds
Current Number of Codecs = 1
Current Codec List = G711(PCMU)
Digitmap Partial Match Timeout = 16
Digitmap Critical Timeout = 4
Cancel Call Waiting Invoke String = *70
Call Transfer Invoke String = *90
CID Block Invoke String = *67
CID Display Invoke String = *82
Call Park Invoke String = *98
Call Retrieve Invoke String = *99
Outside Line Access Number = 9
Use User-Agent Header = No
Set Jitter Buffer Adaptive = No
Use SIP INFO for DTMF = Yes
Re-registration Credential Enable = No
Current SIP PING Interval = 0 seconds
Current SIP PING Proxy Require Header =
Current SIP External IP address =
logs from ATA
********************SLAC_OP_CH_ACTIVE 0
proxy[0]=69.59.142.213:2060
UA#1 prx filter[0]=69.59.142.213:2060
ch2: xxxxxx1 Sign In Ok! (ticks:15227)
ch1: Sign In Ok! (ticks:16327)
rtime=20, pSM->FrameTime=10, pSM->RxPacketization=20, RxFramesPer=2
Mta ****pSM->VoiceFrameLength=80,pSM->CodecType=0
****pSM->TxFramesPerPacket=2,pSM->TxPacketization=20
is_fax_pending:bad ch:0,t38_fax_enable:0
IM_Set_NAT_Throughput
Reserved Bandwidth for voice up:85 down:85
ch = 0,Get the event 0x2
sync with DSP
IM_Set_NAT_Throughput
Reserved Bandwidth for voice up:0 down:0
fast path enabled
Mtaproxy[0]=69.59.142.213:2060

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UA#0 prx filter[0]=69.59.142.213:2060
ch1: xxxxxx2 Sign In Ok! (ticks:19527)
logs from SS console
Monitor 16:11:07:011: basetype=console, ipaddress=*, user=xxxxx, event=*, request=*, serveripaddress=*, server=*, regex=.*.
Registrar 16:11:45:541 sip1: Authentication required for xxxxx@sipsorcery.com from udp:122.181.155.28:5060.
Registrar 16:11:45:979 sip1: Binding update request for xxxxx@sipsorcery.com from udp:122.181.155.28:5060, expiry requested 180s granted 180s.
RegisterSuccess 16:11:46:026 sip1: Registration successful for xxxxx@sipsorcery.com from udp:122.181.155.28:5060 (proxy=udp:69.59.142.213:2060), expiry 180s.
DialPlan 16:11:49:213 sip1: New call from udp:122.181.155.28:5060 successfully authenticated by digest.
DialPlan 16:11:49:244 sip1: Using dialplan default for Out call to sip:18004322737@69.59.142.213:2060.

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NewCall 16:11:49:260 sip1: Executing script dial plan for call to 18004322737.
DialPlan 16:11:49:369 sip1: ** Call from 'xxxxxx xxxxxx' <sip:xxxxxx@69.59.142.213:2060>;tag=ACA8286B13C4-805487280 to 18004322737 **
DialPlan 16:11:49:369 sip1: Calling 18004322737 via Google Voice
DialPlan 16:11:49:385 sip1: SDP on GoogleVoiceCall call had public IP not mangled, RTP socket 172.168.40.107:10000.

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DialPlan 16:11:49:385 sip1: UAS call progressing with Ringing.
DialPlan 16:11:49:400 sip1: Logging into google.com for xxxxx@gmail.com.
DialPlan 16:11:49:432 sip1: Google Voice pre-login page loaded successfully.

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DialPlan 16:11:49:447 sip1: GALX key zNUI1TXOi2I successfully retrieved.
DialPlan 16:11:50:307 sip1: Google Voice home page loaded successfully.
DialPlan 16:11:50:322 sip1: Call key AOLAKgcU4yL1MDlhQShFYlr44cs= successfully retrieved for xxxxxx.xxxxxx@gmail.com, proceeding with callback.
DialPlan 16:11:50:338 sip1: SIP Proxy setting application server for next call to user xxxxx as udp:69.59.142.213:5070.
DialPlan 16:11:50:494 sip1: Google Voice Call to 18004322737 forwarding to 17471740497 successfully initiated, callback timeout=30s.
DialPlan 16:11:50:838 sip1: SIP Proxy directing incoming call for user xxxxxx to application server udp:69.59.142.213:5070.
DialPlan 16:11:50:854 sip1: Google Voice Call callback received.
DialPlan 16:11:50:854 sip1: Answering client call with a response status of 200.
DialPlan 16:11:50:994 sip1: Google Voice Call was successfully answered in 1.61s.
DialPlan 16:11:51:338 sip1: Dial plan execution completed with normal clearing.
DialPlan 16:12:02:916 sip1: Matching dialogue found for BYE to sip:69.59.142.213:5060 from udp:69.59.142.213:5060.
Registrar 16:12:17:775 sip1: Authentication required for xxxxxx@sipsorcery.com from udp:122.181.155.28:5060.
DialPlan 16:12:17:791 sip1: No dialogue matched for BYE to sip:69.59.142.213:2060.
Registrar 16:12:18:072 sip1: Binding update request for xxxxxx@sipsorcery.com from udp:122.181.155.28:5060, expiry requested 180s granted 180s.
RegisterSuccess 16:12:18:119 sip1: Registration successful for xxxxxx@sipsorcery.com from udp:122.181.155.28:5060 (proxy=udp:69.59.142.213:2060), expiry 180s.
calls drops randomly anywhere between 20sec to 1 minute.
I am also not able to use DTMF while the call is on...

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can any of you suggest what the problem is?
or my ATA has gone bad? its a Sunrocket ata with OEM firmware installed.

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thanks alot for all the help...