[isepic]

Hiya Guys! (and Ladies!)
I have a SIpura 3000 - and I think I set it up right to call in using my pots line and hear the beeps dial my PIN and hear a dial tone.. now when I dial a 800 number it goes through! BUT, only 1/2 of the time.. so when I dial a non-800 (toll free) it never goes through.. the call manager pops up showing the outgoing call (just like if I would have lifted the handset and dialed) then it changes to "resource unavailable" - but when I dial 800-444-4444 it usually goes through (50% of the time goes through, 50% of the time gives that error)... the PW that the BV guy gave me I have set in the GW1 - he said that's for the Messenger/Softphone.. not sure if that is the same PW as my main account PW or not as that's set with the auto config.....or is that the account PW? Hmm... anyway, thanks for any help you can offer...
-Chris

[mberlant]

The credentials you enter for Line 1, PSTN Line and gw1 are all independent of each other, as far as the SPA is concerned. Since it is impossible for the phone connected to the Line 1 port to place a call via the service provider registered to the PSTN Line, many people have chosen to put those credentials into gw1, as well.

So, if I can decipher correctly, you have your main BV number programmed into Line 1 and you have your BV softphone account programmed into both gw1 and PSTN Line. Is this correct?

Now, when you call in on your PSTN line and try calling any number that fails, have a look in the Info page to see what phone number your SPA passed to BV. For this purpose you probably want to choose a number that doesn't have seven 4s in a row. (How about TellMe, 1 800 555-8355?)

Please tell us what you find. By the way, your main BV number and your softphone BV number each has its own SIP credentials.

[isepic]

Hey, (1) First off, thanks for taking the time to reply! Much appreciated!

>So, if I can decipher correctly, you have your main BV number >programmed into Line 1 and you have your BV softphone account >programmed into both gw1 and PSTN Line. Is this correct?

(2) Yes, this is correct, even though I have tried it both ways. Current settings is that I have my main BR account programmed into Line1 and the Softphone credentials they gave me (gibberish looking) programmed into the PSTN line and the GW1 line.

>Now, when you call in on your PSTN line and try calling any number that >fails, have a look in the Info page to see what phone number your SPA >passed to BV. For this purpose you probably want to choose a number >that doesn't have seven 4s in a row. (How about TellMe, 1 800 >555-8355?)

(3) Okay, I tried the Tell me and the MCI auto-ani both work ALL the time now. But, *ANY* other number doesn't. Those numbers that don't work, (such as the time number: POPCORN e.g. 7672676 (and other ones like my cell phone etc.) show up properly - the correct digits - I even tried 14087672676 to make sure it didn't need the 1+area code --- which is different - it doesn't say the error msg, it just is silence, and times out -- but shows the proper dialing out number on the info page (see below)

(4) THANK YOU AGAIN for helping! I just don't understand why it would pass the toll-free ones, but not the non-toll free (even local). Weird! Again, thanks for taking the time to read all of this!


Here is some text captures...(from the refreshed info page from the sipura 3000 during the out bound calling (dial in to pstn--- enter pin---new dial tone----dialing out as described in each ===section===)....

=============================
(*NOT WORKING* 767-2676 number - "we're sorry your call cannot be completed at this time"-Voice, and "Resource Unavailable"-Call manager display)
=============================
System Status
Current Time: 12/28/2004 16:51:48 Elapsed Time: 00:16:54
Broadcast Pkts Sent: 0 Broadcast Bytes Sent: 0
Broadcast Pkts Recv: 13 Broadcast Bytes Recv: 1747
Broadcast Pkts Dropped: 0 Broadcast Bytes Dropped: 0
RTP Packets Sent: 4246 RTP Bytes Sent: 679360
RTP Packets Recv: 4216 RTP Bytes Recv: 674560
SIP Messages Sent: 61 SIP Bytes Sent: 34590
SIP Messages Recv: 66 SIP Bytes Recv: 27380
External IP:

Line 1 Status
Hook State: On Registration State: Registered
Last Registration At: 12/28/2004 16:51:27 Next Registration In: 9 s
Message Waiting: No Call Back Active: No
Last Called Number: Last Caller Number:
Mapped SIP Port:
Call 1 State: Idle Call 2 State: Idle
Call 1 Tone: None Call 2 Tone: None
Call 1 Encoder: Call 2 Encoder:
Call 1 Decoder: Call 2 Decoder:
Call 1 FAX: Call 2 FAX:
Call 1 Type: Call 2 Type:
Call 1 Remote Hold: Call 2 Remote Hold:
Call 1 Callback: Call 2 Callback:
Call 1 Peer Name: Call 2 Peer Name:
Call 1 Peer Phone: Call 2 Peer Phone:
Call 1 Duration: Call 2 Duration:
Call 1 Packets Sent: Call 2 Packets Sent:
Call 1 Packets Recv: Call 2 Packets Recv:
Call 1 Bytes Sent: Call 2 Bytes Sent:
Call 1 Bytes Recv: Call 2 Bytes Recv:
Call 1 Decode Latency: Call 2 Decode Latency:
Call 1 Jitter: Call 2 Jitter:
Call 1 Round Trip Delay: Call 2 Round Trip Delay:
Call 1 Packets Lost: Call 2 Packets Lost:
Call 1 Packet Error: Call 2 Packet Error:
Call 1 Mapped RTP Port: Call 2 Mapped RTP Port:

PSTN Line Status
Hook State: Off Registration State: Not Registered
Last Registration At: Next Registration In:
Last Called VoIP Number: 18005558355 Last Called PSTN Number:
Last VoIP Caller: Last PSTN Caller: ,
Last PSTN Disconnect Reason: CPC Signal PSTN Activity Timer: 25420 (ms)
Mapped SIP Port: Call Type: VoIP Gateway Call
VoIP State: Calling PSTN State: PSTN Caller Accepted
VoIP Tone: PSTN Tone: None
VoIP Peer Name: PSTN Peer Name:
VoIP Peer Number: 7672676 PSTN Peer Number:
VoIP Call Encoder: G711u VoIP Call Decoder: G711u
VoIP Call FAX: No VoIP Call Remote Hold: No
VoIP Call Duration: VoIP Call Packets Sent: 0
VoIP Call Packets Recv: 0 VoIP Call Bytes Sent: 0
VoIP Call Bytes Recv: 0 VoIP Call Decode Latency: 0 ms
VoIP Call Jitter: 0 ms VoIP Call Round Trip Delay: 0 ms
VoIP Call Packets Lost: 0 VoIP Call Packet Error: 0
VoIP Call Mapped RTP Port: 16392 >> 0

=============================
(not working 1-408-767-2676 number - Silence (from headset), and times out and then re-order tone...
=============================
System Status
Current Time: 12/28/2004 16:55:04 Elapsed Time: 00:20:10
Broadcast Pkts Sent: 0 Broadcast Bytes Sent: 0
Broadcast Pkts Recv: 14 Broadcast Bytes Recv: 1807
Broadcast Pkts Dropped: 0 Broadcast Bytes Dropped: 0
RTP Packets Sent: 4460 RTP Bytes Sent: 713600
RTP Packets Recv: 4424 RTP Bytes Recv: 707840
SIP Messages Sent: 72 SIP Bytes Sent: 41709
SIP Messages Recv: 78 SIP Bytes Recv: 32411
External IP:

Line 1 Status
Hook State: On Registration State: Registered
Last Registration At: 12/28/2004 16:54:58 Next Registration In: 24 s
Message Waiting: No Call Back Active: No
Last Called Number: Last Caller Number:
Mapped SIP Port:
Call 1 State: Idle Call 2 State: Idle
Call 1 Tone: None Call 2 Tone: None
Call 1 Encoder: Call 2 Encoder:
Call 1 Decoder: Call 2 Decoder:
Call 1 FAX: Call 2 FAX:
Call 1 Type: Call 2 Type:
Call 1 Remote Hold: Call 2 Remote Hold:
Call 1 Callback: Call 2 Callback:
Call 1 Peer Name: Call 2 Peer Name:
Call 1 Peer Phone: Call 2 Peer Phone:
Call 1 Duration: Call 2 Duration:
Call 1 Packets Sent: Call 2 Packets Sent:
Call 1 Packets Recv: Call 2 Packets Recv:
Call 1 Bytes Sent: Call 2 Bytes Sent:
Call 1 Bytes Recv: Call 2 Bytes Recv:
Call 1 Decode Latency: Call 2 Decode Latency:
Call 1 Jitter: Call 2 Jitter:
Call 1 Round Trip Delay: Call 2 Round Trip Delay:
Call 1 Packets Lost: Call 2 Packets Lost:
Call 1 Packet Error: Call 2 Packet Error:
Call 1 Mapped RTP Port: Call 2 Mapped RTP Port:

PSTN Line Status
Hook State: Off Registration State: Not Registered
Last Registration At: Next Registration In:
Last Called VoIP Number: 7672676 Last Called PSTN Number:
Last VoIP Caller: Last PSTN Caller: ,
Last PSTN Disconnect Reason: CPC Signal PSTN Activity Timer: 5890 (ms)
Mapped SIP Port: Call Type: VoIP Gateway Call
VoIP State: Calling PSTN State: PSTN Caller Accepted
VoIP Tone: PSTN Tone: None
VoIP Peer Name: PSTN Peer Name:
VoIP Peer Number: 14087672676 PSTN Peer Number:
VoIP Call Encoder: G711u VoIP Call Decoder: G711u
VoIP Call FAX: No VoIP Call Remote Hold: No
VoIP Call Duration: VoIP Call Packets Sent: 0
VoIP Call Packets Recv: 0 VoIP Call Bytes Sent: 0
VoIP Call Bytes Recv: 0 VoIP Call Decode Latency: 0 ms
VoIP Call Jitter: 0 ms VoIP Call Round Trip Delay: 0 ms
VoIP Call Packets Lost: 0 VoIP Call Packet Error: 0
VoIP Call Mapped RTP Port: 16394 >> 0




=============================
( working 800 number - of the info page)
THIS ONE WORKS ALL THE TIME NOW!! Almost any 800 number!
=============================
System Status
Current Time: 12/28/2004 16:47:32 Elapsed Time: 00:12:38
Broadcast Pkts Sent: 0 Broadcast Bytes Sent: 0
Broadcast Pkts Recv: 9 Broadcast Bytes Recv: 1131
Broadcast Pkts Dropped: 0 Broadcast Bytes Dropped: 0
RTP Packets Sent: 2975 RTP Bytes Sent: 476000
RTP Packets Recv: 2945 RTP Bytes Recv: 471200
SIP Messages Sent: 43 SIP Bytes Sent: 24139
SIP Messages Recv: 48 SIP Bytes Recv: 21231
External IP:

Line 1 Status
Hook State: On Registration State: Registered
Last Registration At: 12/28/2004 16:47:27 Next Registration In: 25 s
Message Waiting: No Call Back Active: No
Last Called Number: Last Caller Number:
Mapped SIP Port:
Call 1 State: Idle Call 2 State: Idle
Call 1 Tone: None Call 2 Tone: None
Call 1 Encoder: Call 2 Encoder:
Call 1 Decoder: Call 2 Decoder:
Call 1 FAX: Call 2 FAX:
Call 1 Type: Call 2 Type:
Call 1 Remote Hold: Call 2 Remote Hold:
Call 1 Callback: Call 2 Callback:
Call 1 Peer Name: Call 2 Peer Name:
Call 1 Peer Phone: Call 2 Peer Phone:
Call 1 Duration: Call 2 Duration:
Call 1 Packets Sent: Call 2 Packets Sent:
Call 1 Packets Recv: Call 2 Packets Recv:
Call 1 Bytes Sent: Call 2 Bytes Sent:
Call 1 Bytes Recv: Call 2 Bytes Recv:
Call 1 Decode Latency: Call 2 Decode Latency:
Call 1 Jitter: Call 2 Jitter:
Call 1 Round Trip Delay: Call 2 Round Trip Delay:
Call 1 Packets Lost: Call 2 Packets Lost:
Call 1 Packet Error: Call 2 Packet Error:
Call 1 Mapped RTP Port: Call 2 Mapped RTP Port:

PSTN Line Status
Hook State: Off Registration State: Not Registered
Last Registration At: Next Registration In:
Last Called VoIP Number: 18005558355 Last Called PSTN Number:
Last VoIP Caller: Last PSTN Caller: ,
Last PSTN Disconnect Reason: CPC Signal PSTN Activity Timer: 29990 (ms)
Mapped SIP Port: Call Type: VoIP Gateway Call
VoIP State: Connected PSTN State: PSTN Caller Accepted
VoIP Tone: PSTN Tone: None
VoIP Peer Name: PSTN Peer Name:
VoIP Peer Number: 18005558355 PSTN Peer Number:
VoIP Call Encoder: G711u VoIP Call Decoder: G711u
VoIP Call FAX: No VoIP Call Remote Hold: No
VoIP Call Duration: 00:12:38 VoIP Call Packets Sent: 261
VoIP Call Packets Recv: 256 VoIP Call Bytes Sent: 41760
VoIP Call Bytes Recv: 40960 VoIP Call Decode Latency: 70 ms
VoIP Call Jitter: 2 ms VoIP Call Round Trip Delay: 0 ms
VoIP Call Packets Lost: 0 VoIP Call Packet Error: 0
VoIP Call Mapped RTP Port: 16384 >> 0

[isepic]

Update to above: with the local numbers, when I dial a 1-408 /vs./ just 408 - get these same results every time:

1-408-xxxxxxx - silence, then reorder after a while, and looks like a normal outgoing call on the call manager
408-xxxxxxx - "we're sorry ..." voice message, and "Resource unavailable" in call manager

Do you think its my dialing plan? Here is my dialing plan for both the line1 and PSTN
(*xx|#xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)

and I did also use the default...
(xx.)
but same results..

the above dialing plan works with my BV line one and was given to me (through their setup .cfg file) and I just cut and pasted it into the DP1 on the PSTN settings page...

and to top it all off.. no difference if I dial 1-800-555-8355 or 800-555-8355 (they both go through just fine with the same results) - and 611 is routed to them normally too---just as if I picked it up on the LINE1 handset and dialed!

also here are my versions..
Software Version: 2.0.8(GW) Hardware Version: 2.0.1(818a

[PhoneBoy]

That's a rather old version of firmware with several known bugs, please upgrade to the latest revision.

[isepic]

Upgraded - but you know what, Broadvoice says to "upgrade" to the .8 version, so I had it higher, then followed thieir instuctions, and unknowling downgraded to .8 ----- so, now that I'm a bit smarter with these things, I upgraded...
Software Version: 2.0.11(GWg) Hardware Version: 2.0.1(818a)

and it still has the exact same problems - I even did a factory reset before and after the upgrade, then used the voxilla wizards to make it again - and its the same results as above.

[isepic]

new info....

Product Information
Product Name: SPA3000 Serial Number: xxxx
Software Version: 2.0.11(GWg) Hardware Version: 2.0.1(818a)
MAC Address: xxxxxx Client Certificate: Installed

System Status
Current Time: 12/28/2003 23:04:48 Elapsed Time: 00:01:09
Broadcast Pkts Sent: 0 Broadcast Bytes Sent: 0
Broadcast Pkts Recv: 1 Broadcast Bytes Recv: 250
Broadcast Pkts Dropped: 0 Broadcast Bytes Dropped: 0
RTP Packets Sent: 164 RTP Bytes Sent: 26240
RTP Packets Recv: 207 RTP Bytes Recv: 33120
SIP Messages Sent: 15 SIP Bytes Sent: 8185
SIP Messages Recv: 16 SIP Bytes Recv: 7961
External IP:

Line 1 Status
Hook State: On Registration State: Registered
Last Registration At: 12/28/2003 23:04:35 Next Registration In: 15 s
Message Waiting: No Call Back Active: No
Last Called Number: 18004444444 Last Caller Number:
Mapped SIP Port:
Call 1 State: Idle Call 2 State: Idle
Call 1 Tone: None Call 2 Tone: None
Call 1 Encoder: Call 2 Encoder:
Call 1 Decoder: Call 2 Decoder:
Call 1 FAX: Call 2 FAX:
Call 1 Type: Call 2 Type:
Call 1 Remote Hold: Call 2 Remote Hold:
Call 1 Callback: Call 2 Callback:
Call 1 Peer Name: Call 2 Peer Name:
Call 1 Peer Phone: Call 2 Peer Phone:
Call 1 Duration: Call 2 Duration:
Call 1 Packets Sent: Call 2 Packets Sent:
Call 1 Packets Recv: Call 2 Packets Recv:
Call 1 Bytes Sent: Call 2 Bytes Sent:
Call 1 Bytes Recv: Call 2 Bytes Recv:
Call 1 Decode Latency: Call 2 Decode Latency:
Call 1 Jitter: Call 2 Jitter:
Call 1 Round Trip Delay: Call 2 Round Trip Delay:
Call 1 Packets Lost: Call 2 Packets Lost:
Call 1 Packet Error: Call 2 Packet Error:
Call 1 Mapped RTP Port: Call 2 Mapped RTP Port:

PSTN Line Status
Hook State: On Line Voltage: 127 (V)
Loop Current: 0.0 (mA) Registration State: Registered
Last Registration At: 12/28/2003 23:04:35 Next Registration In: 15 s
Last Called VoIP Number: Last Called PSTN Number:
Last VoIP Caller: Last PSTN Caller: ,
Last PSTN Disconnect Reason: PSTN Disconnect Tone PSTN Activity Timer: 30000 (ms)
Mapped SIP Port: Call Type:
VoIP State: Idle PSTN State: Ringing
VoIP Tone: PSTN Tone:
VoIP Peer Name: PSTN Peer Name:
VoIP Peer Number: PSTN Peer Number:
VoIP Call Encoder: VoIP Call Decoder:
VoIP Call FAX: VoIP Call Remote Hold:
VoIP Call Duration: VoIP Call Packets Sent:
VoIP Call Packets Recv: VoIP Call Bytes Sent:
VoIP Call Bytes Recv: VoIP Call Decode Latency:
VoIP Call Jitter: VoIP Call Round Trip Delay:
VoIP Call Packets Lost: VoIP Call Packet Error:
VoIP Call Mapped RTP Port:


System Configuration
Restricted Access Domains:
Enable Web Server: yesno Web Server Port:
Enable Web Admin Access: yesno Admin Passwd:
User Password:

Internet Connection Type
DHCP: yesno
Static IP: NetMask:
Gateway:

Optional Network Configuration
HostName: Domain:
Primary DNS: Secondary DNS:
DNS Server Order: ManualManual,DHCPDHCP,Manual DNS Query Mode: ParallelSequential
Syslog Server: Debug Server:
Debug Level: 0123 Primary NTP Server:
Secondary NTP Server:


SIP Parameters
Max Forward: Max Redirection:
Max Auth: SIP User Agent Name:
SIP Server Name: SIP Accept Language:
DTMF Relay MIME Type: Hook Flash MIME Type:
Remove Last Reg: yesno Use Compact Header: yesno

SIP Timer Values (sec)
SIP T1: SIP T2:
SIP T4: SIP Timer B:
SIP Timer F: SIP Timer H:
SIP Timer D: SIP Timer J:
INVITE Expires: ReINVITE Expires:
Reg Min Expires: Reg Max Expires:
Reg Retry Intvl: Reg Retry Long Intvl:

Response Status Code Handling
SIT1 RSC: SIT2 RSC:
SIT3 RSC: SIT4 RSC:
Try Backup RSC: Retry Reg RSC:

RTP Parameters
RTP Port Min: RTP Port Max:
RTP Packet Size: Max RTP ICMP Err:
RTCP Tx Interval:

SDP Payload Types
NSE Dynamic Payload: AVT Dynamic Payload:
INFOREQ Dynamic Payload: G726r16 Dynamic Payload:
G726r24 Dynamic Payload: G726r40 Dynamic Payload:
G729b Dynamic Payload: NSE Codec Name:
AVT Codec Name: G711u Codec Name:
G711a Codec Name: G726r16 Codec Name:
G726r24 Codec Name: G726r32 Codec Name:
G726r40 Codec Name: G729a Codec Name:
G729b Codec Name: G723 Codec Name:

NAT Support Parameters
Handle VIA received: yesno Handle VIA rport: yesno
Insert VIA received: yesno Insert VIA rport: yesno
Substitute VIA Addr: yesno Send Resp To Src Port: yesno
STUN Enable: yesno STUN Test Enable: yesno
STUN Server: EXT IP:
EXT RTP Port Min: NAT Keep Alive Intvl:


Configuration Profile
Provision Enable: yesno Resync On Reset: yesno
Resync Random Delay: Resync Periodic:
Resync Error Retry Delay: Forced Resync Delay:
Resync From SIP: yesno Resync After Upgrade Attempt: yesno
Resync Trigger 1:
Resync Trigger 2:
Resync Fails On FNF: yesno
Profile Rule:
Profile Rule B:
Profile Rule C:
Profile Rule D:
Log Resync Request Msg:
Log Resync Success Msg:
Log Resync Failure Msg:
Report Rule:

Firmware Upgrade
Upgrade Enable: yesno Upgrade Error Retry Delay:
Downgrade Rev Limit:
Upgrade Rule:
Log Upgrade Request Msg:
Log Upgrade Success Msg:
Log Upgrade Failure Msg:

General Purpose Parameters
GPP A:
GPP B:
GPP C:
GPP D:
GPP E:
GPP F:
GPP G:
GPP H:
GPP I:
GPP J:
GPP K:
GPP L:
GPP M:
GPP N:
GPP O:
GPP P:


Call Progress Tones
Dial Tone:
Second Dial Tone:
Outside Dial Tone:
Prompt Tone:
Busy Tone:
Reorder Tone:
Off Hook Warning Tone:
Ring Back Tone:
Confirm Tone:
SIT1 Tone:
SIT2 Tone:
SIT3 Tone:
SIT4 Tone:
MWI Dial Tone:
Cfwd Dial Tone:
Holding Tone:
Conference Tone:
Secure Call Indication Tone:
VoIP PIN Tone:
PSTN PIN Tone:

Distinctive Ring Patterns
Ring1 Cadence: Ring2 Cadence:
Ring3 Cadence: Ring4 Cadence:
Ring5 Cadence: Ring6 Cadence:
Ring7 Cadence: Ring8 Cadence:

Distinctive Call Waiting Tone Patterns
CWT1 Cadence: CWT2 Cadence:
CWT3 Cadence: CWT4 Cadence:
CWT5 Cadence: CWT6 Cadence:
CWT7 Cadence: CWT8 Cadence:

Distinctive Ring/CWT Pattern Names
Ring1 Name: Ring2 Name:
Ring3 Name: Ring4 Name:
Ring5 Name: Ring6 Name:
Ring7 Name: Ring8 Name:

Ring and Call Waiting Tone Spec
Ring Waveform: SinusoidTrapezoid Ring Frequency:
Ring Voltage: CWT Frequency:

Control Timer Values (sec)
Hook Flash Timer Min: Hook Flash Timer Max:
Callee On Hook Delay: Reorder Delay:
Call Back Expires: Call Back Retry Intvl:
Call Back Delay: VMWI Refresh Intvl:
Interdigit Long Timer: Interdigit Short Timer:
CPC Delay: CPC Duration:

Vertical Service Activation Codes
Call Return Code: Blind Transfer Code:
Call Back Act Code: Call Back Deact Code:
Cfwd All Act Code: Cfwd All Deact Code:
Cfwd Busy Act Code: Cfwd Busy Deact Code:
Cfwd No Ans Act Code: Cfwd No Ans Deact Code:
Cfwd Last Act Code: Cfwd Last Deact Code:
Block Last Act Code: Block Last Deact Code:
Accept Last Act Code: Accept Last Deact Code:
CW Act Code: CW Deact Code:
CW Per Call Act Code: CW Per Call Deact Code:
Block CID Act Code: Block CID Deact Code:
Block CID Per Call Act Code: Block CID Per Call Deact Code:
Block ANC Act Code: Block ANC Deact Code:
DND Act Code: DND Deact Code:
CID Act Code: CID Deact Code:
CWCID Act Code: CWCID Deact Code:
Dist Ring Act Code: Dist Ring Deact Code:
Speed Dial Act Code: Secure All Call Act Code:
Secure No Call Act Code: Secure One Call Act Code:
Secure One Call Deact Code: Conference Act Code:
Attn-Xfer Act Code:
Referral Services Codes:
Feature Dial Services Codes:

Outbound Call Codec Selection Codes
Prefer G711u Code: Force G711u Code:
Prefer G711a Code: Force G711a Code:
Prefer G723 Code: Force G723 Code:
Prefer G726r16 Code: Force G726r16 Code:
Prefer G726r24 Code: Force G726r24 Code:
Prefer G726r32 Code: Force G726r32 Code:
Prefer G726r40 Code: Force G726r40 Code:
Prefer G729a Code: Force G729a Code:

Miscellaneous
Set Local Date (mm/dd): Set Local Time (HH/mm):
Time Zone: GMT-12:00GMT-11:00GMT-10:00GMT-09:00GMT-08:00GMT-07:00GMT-06:00GMT-05:00GMT-04:00GMT-03:30GMT-03:00GMT-02:00GMT-01:00GMTGMT+01:00GMT+02:00GMT+03:00GMT+03:30GMT+04 :00GMT+05:00GMT+05:30GMT+05:45GMT+06:00GMT+06:30GM T+07:00GMT+08:00GMT+09:00GMT+09:30GMT+10:00GMT+11: 00GMT+12:00GMT+13:00 FXS Port Impedance: 600900600+2.16uF900+2.16uF270+750||150nF220+820||1 20nF220+820||115nF370+620||310nF
FXS Port Input Gain: FXS Port Output Gain:
DTMF Playback Level: DTMF Playback Length:
Detect ABCD: yesno Playback ABCD: yesno
Caller ID Method: Bellcore(N.Amer,China)DTMF(Finland,Sweden)DTMF(Den mark)ETSI DTMFETSI DTMF With PRETSI DTMF After RingETSI FSKETSI FSK With PR(UK) FXS Port Power Limit: 12345678



Line Enable: yesno

Streaming Audio Server (SAS)
SAS Enable: yesno SAS DLG Refresh Intvl:
SAS Inbound RTP Sink:

NAT Settings
NAT Mapping Enable: yesno NAT Keep Alive Enable: yesno
NAT Keep Alive Msg: NAT Keep Alive Dest:

Network Settings
SIP TOS/DiffServ Value: Network Jitter Level: lowmediumhighvery high
RTP TOS/DiffServ Value:

SIP Settings
SIP Port: SIP 100REL Enable: yesno
EXT SIP Port: Auth Resync-Reboot: yesno
SIP Proxy-Require: SIP Remote-Party-ID: yesno
SIP Debug Option: none1-line1-line excl. OPT1-line excl. NTFY1-line excl. REG1-line excl. OPT|NTFY|REGfullfull excl. OPTfull excl. NTFYfull excl. REGfull excl. OPT|NTFY|REG RTP Log Intvl:

Call Feature Settings
Blind Attn-Xfer Enable: yesno MOH Server:
Xfer When Hangup Conf: yesno

Proxy and Registration
Proxy: Use Outbound Proxy: yesno
Outbound Proxy: Use OB Proxy In Dialog: yesno
Register: yesno Make Call Without Reg: yesno
Register Expires: Ans Call Without Reg: yesno
Use DNS SRV: yesno DNS SRV Auto Prefix: yesno
Proxy Fallback Intvl:

Subscriber Information
Display Name: User ID:
Password: Use Auth ID: yesno
Auth ID:
Mini Certificate:
SRTP Private Key:

Supplementary Service Subscription
Call Waiting Serv: yesno Block CID Serv: yesno
Block ANC Serv: yesno Dist Ring Serv: yesno
Cfwd All Serv: yesno Cfwd Busy Serv: yesno
Cfwd No Ans Serv: yesno Cfwd Sel Serv: yesno
Cfwd Last Serv: yesno Block Last Serv: yesno
Accept Last Serv: yesno DND Serv: yesno
CID Serv: yesno CWCID Serv: yesno
Call Return Serv: yesno Call Back Serv: yesno
Three Way Call Serv: yesno Three Way Conf Serv: yesno
Attn Transfer Serv: yesno Unattn Transfer Serv: yesno
MWI Serv: yesno VMWI Serv: yesno
Speed Dial Serv: yesno Secure Call Serv: yesno
Referral Serv: yesno Feature Dial Serv: yesno

Audio Configuration
Preferred Codec: G711uG711aG726-16G726-24G726-32G726-40G729aG723 Silence Supp Enable: yesno
Use Pref Codec Only: yesno Silence Threshold: highmediumlow
G729a Enable: yesno Echo Canc Enable: yesno
G723 Enable: yesno Echo Canc Adapt Enable: yesno
G726-16 Enable: yesno Echo Supp Enable: yesno
G726-24 Enable: yesno FAX CED Detect Enable: yesno
G726-32 Enable: yesno FAX CNG Detect Enable: yesno
G726-40 Enable: yesno FAX Passthru Codec: G711uG711a
DTMF Process INFO: yesno FAX Codec Symmetric: yesno
DTMF Process AVT: yesno FAX Passthru Method: NoneNSEReINVITE
DTMF Tx Method: InBandAVTINFOAutoInBand+INFOAVT+INFO FAX Process NSE: yesno
Hook Flash Tx Method: NoneAVTINFO Release Unused Codec: yesno
Symmetric RTP: yesno

Gateway Accounts
Gateway 1: GW1 NAT Mapping Enable: yesno
GW1 Auth ID: GW1 Password:
Gateway 2: GW2 NAT Mapping Enable: yesno
GW2 Auth ID: GW2 Password:
Gateway 3: GW3 NAT Mapping Enable: yesno
GW3 Auth ID: GW3 Password:
Gateway 4: GW4 NAT Mapping Enable: yesno
GW4 Auth ID: GW4 Password:

VoIP Fallback To PSTN
Auto PSTN Fallback: yesno

Dial Plan
Dial Plan:
Enable IP Dialing: yesno

FXS Port Polarity Configuration
Idle Polarity: ForwardReverse Caller Conn Polarity: ForwardReverse
Callee Conn Polarity: ForwardReverse



Line Enable: yesno

NAT Settings
NAT Mapping Enable: yesno NAT Keep Alive Enable: yesno
NAT Keep Alive Msg: NAT Keep Alive Dest:

Network Settings
SIP TOS/DiffServ Value: Network Jitter Level: lowmediumhighvery high
RTP TOS/DiffServ Value:

SIP Settings
SIP Port: SIP 100REL Enable: yesno
EXT SIP Port: Auth Resync-Reboot: yesno
SIP Proxy-Require: SIP Remote-Party-ID: yesno
SIP Debug Option: none1-line1-line excl. OPT1-line excl. NTFY1-line excl. REG1-line excl. OPT|NTFY|REGfullfull excl. OPTfull excl. NTFYfull excl. REGfull excl. OPT|NTFY|REG RTP Log Intvl:

Proxy and Registration
Proxy: Use Outbound Proxy: yesno
Outbound Proxy: Use OB Proxy In Dialog: yesno
Register: yesno Make Call Without Reg: yesno
Register Expires: Ans Call Without Reg: yesno
Use DNS SRV: yesno DNS SRV Auto Prefix: yesno
Proxy Fallback Intvl:

Subscriber Information
Display Name: User ID:
Password: Use Auth ID: yesno
Auth ID:
Mini Certificate:
SRTP Private Key:

Audio Configuration
Preferred Codec: G711uG711aG726-16G726-24G726-32G726-40G729aG723 Silence Supp Enable: yesno
Use Pref Codec Only: yesno Echo Canc Enable: yesno
G729a Enable: yesno Echo Canc Adapt Enable: yesno
G723 Enable: yesno Echo Supp Enable: yesno
G726-16 Enable: yesno FAX CED Detect Enable: yesno
G726-24 Enable: yesno FAX CNG Detect Enable: yesno
G726-32 Enable: yesno FAX Passthru Codec: G711uG711a
G726-40 Enable: yesno FAX Codec Symmetric: yesno
DTMF Process INFO: yesno FAX Passthru Method: NoneNSEReINVITE
DTMF Process AVT: yesno DTMF Tx Method: InBandAVTINFOAutoInBand+INFOAVT+INFO
Release Unused Codec: yesno FAX Process NSE: yesno
Symmetric RTP: yesno

Dial Plans
Dial Plan 1:
Dial Plan 2:
Dial Plan 3:
Dial Plan 4:
Dial Plan 5:
Dial Plan 6:
Dial Plan 7:
Dial Plan 8:

VoIP-To-PSTN Gateway Setup
VoIP-To-PSTN Gateway Enable: yesno VoIP Caller Auth Method: nonePINHTTP Digest
VoIP PIN Max Retry: One Stage Dialing: yesno
Line 1 VoIP Caller DP: none12345678 VoIP Caller Default DP: none12345678
Line 1 Fallback DP: none12345678
VoIP Caller ID Pattern:
VoIP Access List:
VoIP Caller 1 PIN: VoIP Caller 1 DP: none12345678
VoIP Caller 2 PIN: VoIP Caller 2 DP: none12345678
VoIP Caller 3 PIN: VoIP Caller 3 DP: none12345678
VoIP Caller 4 PIN: VoIP Caller 4 DP: none12345678
VoIP Caller 5 PIN: VoIP Caller 5 DP: none12345678
VoIP Caller 6 PIN: VoIP Caller 6 DP: none12345678
VoIP Caller 7 PIN: VoIP Caller 7 DP: none12345678
VoIP Caller 8 PIN: VoIP Caller 8 DP: none12345678

VoIP Users and Passwords (HTTP Authentication)
VoIP User 1 Auth ID: VoIP User 1 DP: none12345678
VoIP User 1 Password:
VoIP User 2 Auth ID: VoIP User 2 DP: none12345678
VoIP User 2 Password:
VoIP User 3 Auth ID: VoIP User 3 DP: none12345678
VoIP User 3 Password:
VoIP User 4 Auth ID: VoIP User 4 DP: none12345678
VoIP User 4 Password:
VoIP User 5 ID Auth ID: VoIP User 5 DP: none12345678
VoIP User 5 Password:
VoIP User 6 Auth ID: VoIP User 6 DP: none12345678
VoIP User 6 Password:
VoIP User 7 Auth ID: VoIP User 7 DP: none12345678
VoIP User 7 Password:
VoIP User 8 Auth ID: VoIP User 8 DP: none12345678
VoIP User 8 Password:

PSTN-To-VoIP Gateway Setup
PSTN-To-VoIP Gateway Enable: yesno PSTN Caller Auth Method: nonePIN
PSTN Ring Thru Line 1: yesno PSTN PIN Max Retry:
PSTN CID For VoIP CID: yesno PSTN CID Number Prefix:
PSTN Caller Default DP: 12345678 PSTN CID Name Prefix:
PSTN Caller ID Pattern:
PSTN Access List:
PSTN Caller 1 PIN: PSTN Caller 1 DP: 12345678
PSTN Caller 2 PIN: PSTN Caller 2 DP: 12345678
PSTN Caller 3 PIN: PSTN Caller 3 DP: 12345678
PSTN Caller 4 PIN: PSTN Caller 4 DP: 12345678
PSTN Caller 5 PIN: PSTN Caller 5 DP: 12345678
PSTN Caller 6 PIN: PSTN Caller 6 DP: 12345678
PSTN Caller 7 PIN: PSTN Caller 7 DP: 12345678
PSTN Caller 8 PIN: PSTN Caller 8 DP: 12345678

FXO Timer Values (sec)
VoIP Answer Delay: VoIP PIN Digit Timeout:
PSTN Answer Delay: PSTN PIN Digit Timeout:
PSTN-To-VoIP Call Max Dur: PSTN Ring Thru Delay:
VoIP-To-PSTN Call Max Dur: PSTN Ring Thru CWT Delay:
VoIP DLG Refresh Intvl: PSTN Ring Timeout:
PSTN Dialing Delay: PSTN Dial Digit Len:

PSTN Disconnect Detection
Detect CPC: yesno Detect Polarity Reversal: yesno
Detect PSTN Long Silence: yesno Detect VoIP Long Silence: yesno
PSTN Long Silence Duration: VoIP Long Silence Duration:
PSTN Silence Threshold: very highhighmediumlowvery low Min CPC Duration:
Detect Disconnect Tone: yesno
Disconnect Tone:

International Control
FXO Port Impedance: 600900270+750||150nF220+820||120nF370+620||310nF32 0+1050||230nF370+820||110nF275+780||115nF120+820|| 110nF350+1000||210nF0+900||30nF600+2.16uF900+1uF90 0+2.16uF600+1uFGlobal Ring Frequency Min:
SPA To PSTN Gain: Ring Frequency Max:
PSTN To SPA Gain: Ring Validation Time: 100 ms150 ms200 ms256 ms384 ms512 ms640 ms1024 ms
Tip/Ring Voltage Adjust: 3.1 V3.2 V3.35 V3.5 V Ring Indication Delay: 0256 ms512 ms768 ms1024 ms1280 ms1536 ms1792 ms
Operational Loop Current Min: 10 mA12 mA14 mA16 mA Ring Timeout: 0128 ms256 ms384 ms512 ms640 ms768 ms896 ms1024 ms1152 ms1280 ms1408 ms1536 ms1664 ms1792 ms1920 ms
On-Hook Speed: Less than 0.5 ms3 ms (ETSI)26 ms (Australia) Ring Threshold: 13.5-16.5 Vrms19.35-23.65 Vrms40.5-49.5 Vrms
Current Limiting Enable: yesno Ringer Impedance: High (Normal)Synthesized (Poland,S.Africa,Slovenia)
Line-In-Use Voltage:


Call Forward Settings
Cfwd All Dest: Cfwd Busy Dest:
Cfwd No Ans Dest: Cfwd No Ans Delay:

Selective Call Forward Settings
Cfwd Sel1 Caller: Cfwd Sel1 Dest:
Cfwd Sel2 Caller: Cfwd Sel2 Dest:
Cfwd Sel3 Caller: Cfwd Sel3 Dest:
Cfwd Sel4 Caller: Cfwd Sel4 Dest:
Cfwd Sel5 Caller: Cfwd Sel5 Dest:
Cfwd Sel6 Caller: Cfwd Sel6 Dest:
Cfwd Sel7 Caller: Cfwd Sel7 Dest:
Cfwd Sel8 Caller: Cfwd Sel8 Dest:
Cfwd Last Caller: Cfwd Last Dest:
Block Last Caller: Accept Last Caller:

Speed Dial Settings
Speed Dial 2: Speed Dial 3:
Speed Dial 4: Speed Dial 5:
Speed Dial 6: Speed Dial 7:
Speed Dial 8: Speed Dial 9:

Supplementary Service Settings
CW Setting: yesno Block CID Setting: yesno
Block ANC Setting: yesno DND Setting: yesno
CID Setting: yesno CWCID Setting: yesno
Dist Ring Setting: yesno Secure Call Setting: yesno
Message Waiting: yesno

Distinctive Ring Settings
Ring1 Caller: Ring2 Caller:
Ring3 Caller: Ring4 Caller:
Ring5 Caller: Ring6 Caller:
Ring7 Caller: Ring8 Caller:

Ring Settings
Default Ring: 12345678 Default CWT: 12345678
Hold Reminder Ring: 12345678none Call Back Ring: 12345678
Cfwd Ring Splash Len: Cblk Ring Splash Len:
VMWI Ring Splash Len: VMWI Ring Policy: New VM AvailableNew VM Becomes AvailableNew VM Arrives
Ring On No New VM: yesno


PSTN-To-VoIP Selective Call Forward Settings
Cfwd Sel1 Caller: Cfwd Sel1 Dest:
Cfwd Sel2 Caller: Cfwd Sel2 Dest:
Cfwd Sel3 Caller: Cfwd Sel3 Dest:
Cfwd Sel4 Caller: Cfwd Sel4 Dest:
Cfwd Sel5 Caller: Cfwd Sel5 Dest:
Cfwd Sel6 Caller: Cfwd Sel6 Dest:
Cfwd Sel7 Caller: Cfwd Sel7 Dest:
Cfwd Sel8 Caller: Cfwd Sel8 Dest:

PSTN-To-VoIP Speed Dial Settings
Speed Dial 2: Speed Dial 3:
Speed Dial 4: Speed Dial 5:
Speed Dial 6: Speed Dial 7:
Speed Dial 8: Speed Dial 9:

PSTN Ring Thru Line 1 Distinctive Ring Settings
Ring1 Caller: Ring2 Caller:
Ring3 Caller: Ring4 Caller:
Ring5 Caller: Ring6 Caller:
Ring7 Caller: Ring8 Caller:

PSTN Ring Thru Line 1 Ring Settings
Default Ring: 12345678Follow Line 1

[PhoneBoy]

Are both Line 1 and PSTN registered with BroadVoice? That's going to be a problem.

[isepic]

Well, yes, kinda - I have one with my "normal" ID, and the other one with my SOftphone ID they gave me a while back.
I can't use both? They never said that when I called... hmm so you saying this can't be done unless I pay them for two lines? and also pay my pots
provider too??? That's the whole reason I got the SIP3000 so I can use my dsl line Im paying $5 for a month, AND my VOIP line - and call into
it using my pots, and dial out VOIP saving LD from like my cell, or office or friends home - i dont plan to use both at the same time (bv main line and
dial into pots and transfer out of voip --- I dont know the PW for the main line, so I had to have it provision, and set it, then turn off the provision
(they did give me the pw to log on as Admin to the box)..

Gosh I really appreciate all the help on here!!!

[isepic]

Okay this is now fixed! From another thread - http://voxilla.com/forum-viewtopic-t-1761.html they suggested to change some DTMF settings, check out
the thread if you're having a similar issue! All the calls go through now just by changing these two DTMF settings (and perhaps the firmware too --- which
I wouldn't doubt --- its a bummer that Broadvoice when ding the initial provisioning, wilL DOWNGRADE your firmware to x.8 when its at x.11 :-(

Thanks to EVERYONE who helped! I appreciate you taking the time!!!

Happy New Year!