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Unread 02-04-2010, 04:19 AM   #1
tatjam
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Default advice on Sipura 2100 dial plan needed

I've recently been experimenting with dial plans on my Sipura 2100 and wanted to ask whether the strings I've come up with might be improved at all. I'm not so great with this kind of logic, so I assume they could use some improvement and/or simplification. I should point out that what I've got does seem to work fine, only I suspect there may be better ways of doing it. I'm in the US, by the way, which will become apparent as I progress.

Please note that this post refers to dial plans entered into the Sipura itself, and not to dial plans entered into my voxalot account. For the time being I'm sort of forced to do all this on the Sipura since, unless I directly connect to my voice provider (justvoip), I end up getting FUP exceeded charges.

So, here's what I've got:
Code:
(18[06-8][06-8]xxxxxxxS0|<:00>1[2-9]xx[2-9]xxxxxxS0|<911:0014141234567>|<:001414>[2-9]xxxxxxS0|00xx.<:@sip.justvoip.com;usr=my-usr;pwd=my-pwd>|[x*][x*].)
Explanation: all outgoing calls to the PSTN go through justvoip and, in accord with their stipulations, must be preceded by 00 (plus country code plus area code plus number). The first rule I created for toll free calls, which should go through voxalot, not through my VP. The second rule allows me to dial 1 plus 10-digit (area code and number) calls like on a PSTN line (it prepends 00 to the 1). The next rule allows me to reach the local non-emergency police dispatcher by dialing 911 (as close as I could come so far to anything like a real 911 service). The next rule prepends 001414 (my local area code plus justvoip's 001 prefix) to any 7-digit number dialed from my phone--in effect allowing me to make local calls in the same fashion as they're made from a PSTN phone. The next rule sends all calls with 001 prepended to them to my voice provider using their URI. The final rule is just the only rule I used to have in the Sipura. I have yet to experiment with removing and/or modifying it and would appreciate advice on it in particular.

Like I said, what I've got is mostly effective. There are cases, though, when I see a fairly lengthy delay before my call is terminated--I presume that's because the Sipura's built-in timer has kicked in and needs to time out (I think timers are set to 3 or 10 seconds depending on the case) before actually sending the call. It seems to me that improvements in my rules should therefore focus generally on minimizing or eliminating the delays.

So, any input on how my dialing rules might be improved? How about that last entry: do I really need that? I have yet to formulate a rule specifically for calling other voxalot users but I think that, were I to get rid of that final rule, it should probably be replaced by one designed to send calls to other voxalot users (like maybe xxxxxxS1?)

Help will be appreciated.

Thanks,
James
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Unread 02-07-2010, 08:28 AM   #2
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Your first match entry, 18[06-8][06-8]xxxxxxx, will match some numbers that are not toll-free. For instance it will match 806, 807 and 808 area codes. Instead, try using 8 separate match entries.
800[2-9]xxxxxx
866[2-9]xxxxxx
877[2-9]xxxxxx
888[2-9]xxxxxx
1800[2-9]xxxxxx
1866[2-9]xxxxxx
1877[2-9]xxxxxx
1888[2-9]xxxxxx

If you'd like to choose which termination provider carries your toll free call insert the following entries (and be sure to set 'Enable IP Dialing' = yes).
<#0,1:>[x*][x*].<:@voiper.ipkall.com:5060>|
<#1,:>[x*][x*].<:@tf.voipmich.com:5060>|
<#2,:>[x*][x*].<:@tollfree.sip-happens.com:5060>|
<#3,:>[x*][x*].<:@sip.tollfreegateway.com:5060>|
<#4,:>[x*][x*].<:@proxy.ideasip.com:5060>|
<#5,:>[x*][x*].<:@proxy01.sipphone.com:5060>|

About your match entry "00xx.<:@sip.justvoip.com;usr=my-usr;pwd=my-pwd" I did not know the SPA2100 can use the usr= and pwd= commands in the dial plan. Is that working for you? If so, what 'Software Version' is installed in your SPA2100?
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Unread 02-07-2010, 10:40 PM   #3
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Thanks for your reply, boatman. You're sure right about the fact that my toll-free dial plan could match non-toll-free numbers. So far it's worked fine because I've never tried to call anyone in the 806, 807, or 808 area codes.In fact, I'm not sure I ever would call anyone in those area codes. All the same I thank you for pointing out that potential problem and suggesting alternatives.

I'm still puzzling over the alternate toll-free entries you've provided. It seems evident that the ATA will route those calls through some alternate voice provider, correct? And do these providers terminate toll free calls to anyone, or must one set up an account of some sort before they will terminate toll-free calls? And, furthermore, why do you provide so many entries? Is it so that, in case one provider can't terminate the call, you can use an alternate one?

I can't say I really understand what those entries do. I'll have to consult the manual again to try and figure it out.

And my Sipura does seem work with the commands you asked about. I determined, at some point after entering that dial plan, that calls could not have been going through voxalot. But I've forgotten now how I tested that. In any case, the software version on this ATA is 3.2.5(d). Now I'll have to go off and try to remember what lead me to conclude that calls were, in fact, being routed directly to my voice provider and were not first going through voxalot.

James

PS Any suggestions for a dial plan for when I'm calling other voxalot users, i.e., when the phone number consists in just 6 digits?
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Unread 02-07-2010, 11:51 PM   #4
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Quote:
Originally Posted by tatjam View Post
And do these providers terminate toll free calls to anyone, or must one set up an account of some sort before they will terminate toll-free calls?
No account is needed.

Quote:
Originally Posted by tatjam View Post
And, furthermore, why do you provide so many entries? Is it so that, in case one provider can't terminate the call, you can use an alternate one?
In case the call cannot be completed for any reason you can try another provider. Additionally, some people like to use the provider with the lowest voice latency or best audio quality. The ping time to the provider's RTP connection address will give you some idea about the voice latency. Last time I checked the RTP connection addresses were as follows:

voiper.ipkall.com - 66.54.140.46
tf.voipmich.com - 69.41.0.51
tollfree.sip-happens.com - 65.98.233.182
sip.tollfreegateway.com - 204.8.45.222
proxy.ideasip.com - 64.2.142.87
proxy01.sipphone.com - 130.94.88.94

Quote:
Originally Posted by tatjam View Post
I can't say I really understand what those entries do. I'll have to consult the manual again to try and figure it out.
Press # followed by the number of your choice, then you will hear a kind of dial tone, dial the number and press # again.

Quote:
Originally Posted by tatjam View Post
I determined, at some point after entering that dial plan, that calls could not have been going through voxalot. But I've forgotten now how I tested that. In any case, the software version on this ATA is 3.2.5(d). Now I'll have to go off and try to remember what lead me to conclude that calls were, in fact, being routed directly to my voice provider and were not first going through voxalot.
Just reload the ATA's status page during the call and note 'Call 1 Peer Phone:'. If it shows only the number dialed then the call is going through the provider set in 'Proxy:' on line 1 or line 2 settings tab.

Quote:
Originally Posted by tatjam View Post
PS Any suggestions for a dial plan for when I'm calling other voxalot users, i.e., when the phone number consists in just 6 digits?
Maybe just press # after dialing the 6-digit number.
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Unread 02-08-2010, 12:30 AM   #5
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Quote:
Originally Posted by boatman View Post
No account is needed.
<snip>
Press # followed by the number of your choice, then you will hear a kind of dial tone, dial the number and press # again.
"Number of your choice," I gather, meaning 0, 1, 2, 3, 4, or 5? Just for additional clarification, x* means any digit or the star key, correct?
Quote:
Just reload the ATA's status page during the call and note 'Call 1 Peer Phone:'. If it shows only the number dialed then the call is going through the provider set in 'Proxy:' on line 1 or line 2 settings tab.
Just tried it and it shows 0012345678900. That's just the number dialed, if what you mean by "just the number dialed" is that only a number, i.e., without @sip.justvoip.com appended, will appear there. Is that what you meant? If so, then I may have been laboring under a delusion for the past couple of weeks (not a unique instance in my life to date).
Quote:
Maybe just press # after dialing the 6-digit number.
That's what I've been doing. I just thought there might be a more sophisticated way to do this.

Thanks,
James
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