said by tlpintpe :

Is there are 255 character limit per dial plan?

Notes:  - The dial plan length limit for <Dial Plan 1> through <Dial Plan 8>
is 511 characters. This is less than that for the <Dial Plan> under [Line 1],
which is 2047 characters. 

said by DracoFelis :

I'm wondering if it is your browser doing the truncation on the fields, as my current "Dial Plan" is already well over 255 chars. And from the Sipura SPA-3000 manual, we have this comment:

Notes: - The dial plan length limit for through is 511 characters. This is less than that for the under [Line 1], which is 2047 characters.

said by tlpintpe :

I am using Firefox on Linux, but I'll give it a go with Konqourer and see if that helps.

said by DracoFelis :

I'm wondering if it is your browser doing the truncation on the fields, as my current "Dial Plan" is already well over 255 chars. And from the Sipura SPA-3000 manual, we have this comment:

said by tlpintpe :

I think it was me trying to just cut and paste from your entries (dialing toll free numbers via FWD) that was the culprit. There is a carrage return in there, and when I edited the dial plan in a text editor, and removed the carrage return, then reentered the dial plan, all worked as it should.

1 747 xxx xxxx <:@proxy01.sipphone.com>

1 222 xxx xxxx <:@proxy01.sipphone.com>

said by p2pvoice :

Will this "trick" (inspired by your SIPPhone post) work?


I don't know where to put UN/PW?

911 S0 <:@GW1>

said by p2pvoice :

DracoFelis:

I have SPA1001 with two virtual lines. Line 1 is set up for "home" and Line 2 is set up for "business" with a different ringtone. Each line is an extension on my Asterisk server. So, by using a single phone, I can tell from the ringtone, whether an incoming call is a "home" call or a "business" call. I use primarily Line 1 for outgoing calls using Asterisk's outbound routing.

Here is the "trick: I need to implement:

My business number is an IPKall number, which "points" to my "business" extension on my Asterisk server.

My home number is "ported" to an ITSP and it "points" my "home" extension.

This ITSP is fully 911 compliant; however, I use this ITSP for incoming calls only (don't use this ITSP for outgoing calls). Since this ITSP is fully 911 compliant, I would like only the 911 call to go through this ITSP instead of the other ITSPs configured in my Asterisk's outbound routing.

Will this "trick" (inspired by your SIPPhone post) work?

911

I don't know where to put UN/PW?

BTW, my Asterisk server is registered to My911ITSP.com

Thanks

(S1 <:18005551212>)

said by rizzo2dial :

Another flaw is that this implementation doesn't allow LINE2 to be used as a backup VoIP provider for other outgoing calls (unless you add more to the LINE2 dialing plan and can dial your other # within 1 second).

I'm hopeful a better solution can be reached; however, I haven't been successful in otherwise getting a LINE1 dialing plan instruction to dial out on LINE2.

<911:15555551234>S0

said by DracoFelis :

I know you can reach a better solution with an SPA-3000. Because with an SPA-3000, you just put your 911 provider on one of the 4 "gateway" slots, and then just tell your dial plan to use the "gateway" for 911 dialing.

said by DracoFelis :

Edit: If you do have an SPA-3000, check out a previous "trick" I posted in this (long) thread, on how to use the four SPA-3000 "gateway" slots to store (and use) the login credentials of additional outbound only VoIP providers!

said by DracoFelis :

BTW:
Don't forget that if you are willing to give up E911, and do simple "speed dialing" (without the enhanced location info sent to your emergency center), you can replace a dialed 911 with a quick "speed dial" to whatever is a proper "emergency number" for your area, simply by playing with your "dial plan". And that "trick" (already posted in a previous message in this thread) works in most LinkSys/Sipura adapters, simply by modifying your "dial plan"!

<911:2#>S0

said by DracoFelis :

For example, if you check with your local emergency services, and find that a local emergency number is "1-555-555-1234" (that is a bogus number, use a number appropriate for your area), you can add this little pattern to the front of your dial plan (again, modified to use your real "emergency number", not the bogus 1-555-555-1234) to trap for a user dialing 911, and instead "speed dial" your local emergency number:

said by rizzo2dial :

Speaking of SPEED DIALING, the USER1 tab "Speed Dial [2-9]" entires actually support dialing out on the opposite line of the SPA-1001. For example, if the "Line 2 Select" key is left as the default value of "#" and a "SPEED DIAL x" entry on LINE1 gets configured as "#18005551212", it will dial "18005551212" on LINE2 (w/ a LINE2 dial-tone being heard for a split-second).

Unfortunately this same behavior doesn't occur when configuring it in a DIAL PLAN speed dial (as suggested in the quote below). It also doesn't work when convertijng a DIAL PLAN speed dial entry into a USER TAB speed dial entry:
Converts DIAL PLAN speed dial "911" to USER TAB "Speed Dial 2." While dialing "2#" directly allows dialing out on LINE2 (if configured to do so), dialing "911" (where it gets converted to "2#") doesn't work.

<911:2>S0|

said by rizzo2dial :

Here's how to configure LINE1 of the SPA-1001 to dial "911" on LINE2:

1) In the "Phone" tab, set the "Line 2 Select Code" to be the default value of "#"

2) In the "User 1" TAB, configure a "Speed Dial [2-9]" entry of your choice as "#911" (for dialing "911" on LINE2).

3) Assuming you configured entry "Speed Dial 2" in the previous step, insert into the beginning of the LINE1 Dialing plan the following:
If you configured a different "Speed Dial [2-9]" entry, replace the ":2" above with ":x"
(where x is the configured "Speed Dial [2-9]" entry #).

<12:2>S0

said by rizzo2dial :

Here's how to configure LINE1 of the SPA-1001 to dial "911" on LINE2:

1) In the "Phone" tab, set the "Line 2 Select Code" to be the default value of "#"

2) In the "User 1" TAB, configure a "Speed Dial [2-9]" entry of your choice as "#911" (for dialing "911" on LINE2).

3) Assuming you configured entry "Speed Dial 2" in the previous step, insert into the beginning of the LINE1 Dialing plan the following:
If you configured a different "Speed Dial [2-9]" entry, replace the ":2" above with ":x"
(where x is the configured "Speed Dial [2-9]" entry #).

When dialing 911, you'll hear a LINE2 dialtone for a split second followed by the true 911 call on LINE2.

Even if you dial any digits after 911 on LINE1 (i.e. 9111, 91111, 911111, etc.), they'll be ignored (which is precisely what you want for 911 purposes).

With this solution, you retain full LINE2 functionality should you want/need to manually dial out on that line.

Rizzo

<911:2>S0|<2#:>S0|

<911:2>S0|<2#:#>S0|

<911:2>S0|<2#:xxxxxxxxxxx>S0|
Replace 'xxxxxxxxxxx' w/ the phone number of your choice.

said by igi :

I have a question about latency though in your Line 2/Line 1 forwarding: have you checked if this increases latency/delay when you call another FWD user?

said by DracoFelis :

4) If I'm correct in what is going on, that would also explain how to directly call a Sipura "peer to peer" without any "service provider" (even FWD) at all! Anyone with a Sipura want to help test this theory out? Can another user on the internet directly "ring our phone" by calling the same URI that we have in our "forwarding" trick? I bet the answer is "yes", but I would like to test this theory out first, before saying for sure that it will work!

SIPphone server response:
Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux))

said by DKreil:

Is there any form of documentation of call plan syntax around or did you learn all that from the published examples & trial and error? I have paged through the Users' guide but it seems more like a list of features available for provisioning by VoIP providers.

said by DKreil:

I have tried to learn from your examples but I cannot get the call forwarding to work on my SPA-2100.

said by DKreil:

- localhost & port
4832525@127.0.0.1:5061

said by DKreil:

- SIP account at provider (sipgate, Austria):
4832525@sipgate.at

said by DKreil:

- external IP & port
4832525@voice.kreil.net:5061

said by DKreil:

Is it likely that it's a difference between the SPA2100 and the 3000? Is it worth going for the 3000 instead?

said by DKreil:

Lastly, I have tried to follow the discussion about using Gateway Accounts on the forum but am not quite sure of the latest: Is there now a way one can use Gateway Accounts or information in the calling plan to switch providers which
- require a password
- have a different user ID?

1[2-9]xx[2-9]xxxxxxS0 <:@GW1> 

said by DracoFelis :

For "dial plan" help, pay very careful attention to the "Appendix 1" (the last 3 or 4 pages of the PDF containing the user guide).

said by DracoFelis :

This one should work if BOTH of the following are true:

1) Anyone off the net can call your phone by calling the SIP URI "4832525@sipgate.at", even if/when they don't have an account with "sipgate.at". If this isn't the case, the forwarding will fail, because you have just forwarded to a destination that doesn't accept the call.

said by DracoFelis :

2) The VoIP provider you are registered with on the line you are forwarding from supports the "SIP reinvite" (call redirect) message. If not, call forwarding will never work from that line, as the Sipura doesn't appear to actually do the forwarding (instead it tells the calling SIP party/device to call somewhere else, and not all SIP providers/devices will follow that redirect message)!

said by DracoFelis :

And now (drum roll), how to forward all inbound calls to the OTHER line:
This is VERY USEFUL, because it either lets you have a TWO VoIP accounts that both "ring" the same phone, OR lets you use one account for all incoming, and a 2nd account for all outgoing (by putting the "phone" on the line with the outgoing VoIP service, and then forwarding all incoming calls on that other VoIP line to that one)!

said by JakesOnline :

I tried forwarding to line1id@127.0.0.1:5060, line1@myDynDNSname.com:5060 and line1id@myinternetip:5060.

i'm pretty sure i followed every step.

ports are forwarded properly on the nat as well.

Any ideas?

said by aup2 :

When using the Sipura (e.g., 2100) to do P2P calling without registering with a Proxy, does the unit use the settings for the particular line from which the call is made, even though the line is not registered?