Mta 6328 2re Firmware Download

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

The standalone MTA 6328-2Re with 2 voice ports is an ideal solution for broadband service providers looking to deploy new revenue-generating telephony services to their customers. Compatible with any standard analog telephone set, the MTA 6328-2Re delivers voice quality and features equivalent or superior to those of PSTN. 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.

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 19:46:23
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
****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
proxy[0]=69.59.142.213:2060
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.
NewCall 16:11:49:260 sip1: Executing script dial plan for call to 18004322737.
DialPlan 16:11:49:369 sip1: ** Call from 'xxxxxx xxxxxx' ;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.
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.
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.

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 19:46:23
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
****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
proxy[0]=69.59.142.213:2060
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.
NewCall 16:11:49:260 sip1: Executing script dial plan for call to 18004322737.
DialPlan 16:11:49:369 sip1: ** Call from 'xxxxxx xxxxxx' ;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.
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.
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..
can any of you suggest what the problem is?
or my ATA has gone bad? its a Sunrocket ata with OEM firmware installed.
thanks alot for all the help..

Mta 6328 2re Firmware Download Free

MTA 6328-2Re Specifications Next Generation VoIP CPE Devices For Broadband Service Providers

Features| Specifications |Datasheet

A.	Power
B.	RJ-45 port (uplink to broadband access device)
C.	RJ-45 (downlink to PC)
D.	RJ-11 port (connect to phone)

Product Specifications

Category	Specifications
Telephone Interface	2 FXS voice ports
Network Interface	10/100 Base-T RJ-45 Uplink and Downlink ports
Accessory	Ethernet Cable, AC/DC Power Adapter

Software Specifications

Mta 6328 2re Firmware Download Torrent

Category	Specifications
Protocols	SIP 2.0, MGCP 1.0, NCS 1.0
Speech Codec Capabilities	G.711 and one of the following: G.726 G.723.1; G.729A (Low bit rate codecs) Supports 3-way conferencing with compression
Quality of Service	IEEE 802.1p/q; IP TOS Tagging; Built-in Priority Switch; Data Bandwidth Control; Adaptive jitter buffer
Signal Processing	Echo cancellation Silence suppression T.38 Fax (or fall-back to G.711) Caller ID FSK signal regeneration Line reversal/Polarity reversal 16Khz metering pulse (MGCP only)
Certification	FCC part 15B; CE; UL
Tones	Ring back tone Busy tone Reorder tone Dial tone Off hook warning tone Message waiting tone (MWI)/Stutter tone Call waiting tone
DTMF Tone	DTMF tone detection and generation/RFC2833
Announcements	Play out any voice stream sent by Call Agent or SIP Proxy controlled announcement server Device IP announcement
OAM&P	Access components implemented: CLI, TFTP, HTTP 1.0, SNMP, Telnet, DHCP or DNS, HTTPS (available soon) Works with any SNMP (v.1, v.2c, v.3)-based EMS Offers web-based access as well as HTTP, Secured HTPP, or TFTP-based remote software downloads/upgrades Provisionable set feature codes
Other Features	Built-in DHCP server NAT capabilities for simultaneous Internet access for multiple PC's IP routing and port forwarding MAC cloning IP/Domain filtering STUN NAT traversal SIP TLS over TCP Multiple Line Profiles (SIP only) DSCP (Diffserv Codepoint) (SIP only) VLAN Tagging SIP INFO for DTMF/Flash Event SIP Notify for Flash Event SIP PING SIP PRACK RTCP-XR (SIP only)

Physical Specifications

Mta 6328 2re Firmware Download

Category	Specifications
Power Consumption	Idle: 12V/0.19A (2.28W) / Talking: 12V/0.28A (3.36W)
Power Supply	Output: DC 12V, 1A / Input: AC 120V, 60Hz, 200mA
Dimensions	1.18 in (H) x 4.60 in (W) x 5.12 in (D) / 30 mm (H) x 117 mm (W) x 130 mm (D)
Operating Temperature	32ºF to 104ºF (0ºC to 40ºC)
Storage Temperature	-4ºF to 158ºF (-20ºC to 70ºC)
Operating Humidity	10 to 90% RH
Storage Humidity	5 to 95% RH