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devil24
Premium
join:2002-06-28
Houston, TX
| reply to DracoFelis
Re: [Equipment] Useful Sipura tricks...
said by DracoFelis  :I just got my SPA-3000 today, and (while the process was more painful than expected) I've already figured out how to have both my DialPad.com ($11.99/month) and my Teliax.com (pay as you go) accounts running on "Line 1" (and both accounts have different SIP passwords). Now if I could just find a reliable way to override the "User ID:" field (the "GWx Auth ID:" fields don't seem to behave as expected in this regard), I would be able to include my other SIP accounts that were done under a different userid (in addition to accounts with a different proxy and password)... [edit] Just got the solution to my SPA-3000 problem from the Voxilla forums. Instead of putting in my provider's "proxy_address" (i.e. 66.35.222.58 for DialPad.com) in for the value of "Gateway x:" and my userid for the value of "GWx Auth ID:", you put in "userid@proxy_address" (i.e. myaccount@66.35.222.58 for DialPad) for the "Gateway x:" field, and then also put in your "userid" for the "GWx Auth ID:" field. You apparently also need a "new enough" version of the firmware for this to work (my SPA-3000 has version "2.0.13(GWg)", which works fine with this "trick"). But once you get this "trick" to work, you can use the 4 "gateway" fields to totally override your userid/password/proxy settings of "Line 1" (of an SPA-3000). This allows you to easily have line 1 setup for 5 different VoIP providers (the default "Line 1" one being by-directional, and the 4 "gateway" ones being "outbound only"). And those 5 providers are on top of any providers (such as FWD) that you can call out to without "authorizing" (i.e. using the previous described "@proxy" trick to the dial-plan). BTW: At the moment I only have 3 providers, so I have some room for expansion. My "Line 1" default is currently setup for FWD (this is only so that I can receive inbound calls by FWD, otherwise I would have used the previous FWD outbound "trick"), my GW1 is setup for DialPad.com ($11.99/month "unlimited", and also the provider my dial plan selects for LD calls), and GW2 is my "pay as you go" Teliax account (default dialing for toll free numbers, and can be explicitly used by dialing "# 8 call_digits #"). Hey DracoFelis, great job, pal!
Now, can this 'multiple providers in one line' trick work on a SPA-2002??? if so, can you please post a more detailed example of how to set it up???.
I'm currently subscribed to 3 services, all running from my SPA-2002 and having to erase/edit any of the 2 active ones every time I want to use the other one is a PITA, so, if this is able to work on my device, it'll make things much much easier for me  .
Thanks in advance for your help. | |
DracoFelis
Premium
join:2003-06-15
1 edit | said by devil24 :Now, can this 'multiple providers in one line' trick work on a SPA-2002??? if so, can you please post a more detailed example of how to set it up???. I'm currently subscribed to 3 services, all running from my SPA-2002 and having to erase/edit any of the 2 active ones every time I want to use the other one is a PITA, I don't have an SPA-2002, but as far as I know the SPA-2002 has pretty much the same features as the SPA-2000 (which I do have).
So I think the answer is that the SPA-2002 can do about as much as the SPA-2000 can. Specifically, that would mean that providers that let you call their subscribers without authorization (such as FWD or SIPphone) can be used with appropriate "Dial Plan" tricks (shown above). But as far as I'm aware, you need an SPA-3000 to use any of the "gateway" tricks that allow multiple providers that require a username/password on the same "phone".
BTW: It was the "gateway" (multiple providers WITH authorization, on the same "phone") features, that were a strong reason why I recently purchased my SPA-3000 (from voxilla.com), despite the fact that I already owned a properly functioning SPA-2000. The SPA-3000 really does give you more options than many of the other Sipura models. OTOH it's not as if my investment in my SPA-2000 is totally wasted either. I can still use the SPA-2000 if/when I ever want to setup a "remote extension", and I will probably also make use of it if/when I ever setup an * box. And I may even experiment (at some point) with hooking the SPA-2000's ports into the "Line port" of the newer SPA-3000 (doing so should "in theory" allow for even more providers on the same "phone", than you can do with just the SPA-3000 by itself).
[EDIT]: As to your example, here goes. Just keep in mind that the "gateway" features are SPA-3000 specific (and won't work on say an SPA-2000)!
You first setup your primary "Line 1" provider normally (including "registration" with their SIP proxy). Choose your primary provider carefully, as that is the only provider your SPA-3000 will allow inbound calls from (unless you do clever tricks with forwarding and/or hooking up another ATA to the "Line port"). In my case, I use FWD as my "primary", as all my pay providers are currently not supplying inbound DIDs (and so FWD is the only provider I currently need to "ring" the phone).
For the other 4 outbound providers, you setup the "Line 1" (again SPA-3000 only) providers as follows:
The "Gateway x:" field gets "userid@proxy" (NOTE: It's non-obvious from the docs, but you really need the "userid @ the_proxy_address" in the gateway field, not just the proxy address!). The "GWx Auth ID:" ID gets your userid (yes, you need it in this field by itself, and part of the gateway field info). The "GWx NAT Mapping Enable:" field gets whatever your desired NAT setting is (in my case "Yes"). And the "GWx Password:" field obviously gets the password for that provider.
Once you have the Gateway fields (there are 4 of them, allowing for up to 4 additional "authorized" outbound providers) filled in, the only other step is to modify the "Dial Plan" to pick when you want to use that "gateway". For example, I have DialPad.com in my "gateway 1", so I use the following to send most LD calls to DialPad:
1[2-9]xx[2-9]xxxxxxS0 <:@GW1>
1 800 [2-9]xxxxxx S0 <:@GW2> | 1 888 [2-9]xxxxxx S0 <:@GW2> | 1 877 [2-9]xxxxxx S0 <:@GW2>
* | 1 866 [2-9]xxxxxx S0 <:@GW2>
(*) WARNING 1 long line(s) split
BTW: While I haven't played with it yet, the "Line port" is considered "gateway 0". So if you wanted to send 911 calls to the "Line port" (for example, if you hooked up the line port to a POTS line), you could probably use the following in your "Dial Plan":
NOTE: I think the 911 example is correct, but as I've already mentioned, I haven't tested it yet! | |
devil24
Premium
join:2002-06-28
Houston, TX | Thanks a lot for your reply . | |
balta
@netvisao.pt
| reply to DracoFelis
Hi.Just received my SPA-3000 today and need somehelp to configure my scenario.
I've two providers  ne [Provider A] gives me a PSTN number, the other [Provider B] doesn't. Nonetheless provider B has better rates, but unfortunatelly seems to require an outbound proxy.
The problem I get at this stage is that GTWn dial plan aliases do not allow oubound proxies (only the proxy) part, so I can't set Provider A on line 1, and and Provider B as a dialplan gateway.
On the other hand I don't seem to be able to forward all VOIP2 calls to the FXS phones. Any help ?
Thanks | |
MichiganTelephone
@anonymizer.com
1 edit | reply to DracoFelis
I have a few Sipura tricks listed on the page, How to Distribute VoIP Throughout a Home, which I will copy here so you don't have to search through the page to find them (they are near the bottom of the page). This is copied verbatim from the page so ignore phrases like "as mentioned earlier" since they don't apply here:
Additional hints for emulating "real" telephone service (Sipura adapters only, may also apply to Linksys adapters that use Sipura technology)
In this section we will just mention some Sipura adapter defaults that you or your service provider may want to change in order to provide service that better emulates regular wireline telephone service. Except perhaps for the first, these are NOT essential settings, and none of them in any way affect call quality. To change these settings, go to the web interface for your Sipura adapter (enter the local IP address for the Sipura in web browser), then click on "advanced" in the upper right corner of the screen, then click on the "Regional" tab. If this tab is not visible you may need to do an "Admin Login", which requires a password, or you may need to ask your VoIP provider to make the desired changes for you. Please note that the tone settings shown below are for the United States and Canada only.
•In case you missed it above, in the U.S. and Canada, under Ring and Call Waiting Tone Spec, the Ring Voltage should be set to 90 and (this is most important) the Ring Frequency to 20 — this not only allows older phones with mechanical bells to work, but it just might help in a few odd cases where Caller ID doesn't seem to work properly on a particular phone. In fact, if you have any weird problems with equipment that worked fine with traditional phone service not working with VoIP, and that equipment is activated by a ring signal, this may be the problem. Sipura adapters default to a ring frequency of 25 Hz, which is NOT the frequency usually used in the United States and Canada.
•Also as mentioned earlier, under the Control Timer Values (sec) section, we suggest setting CPC Delay to 10 and CPC duration to 1, because if you have one or more phones with a "hold" button and you ever put a call on hold and then no one picks it up, this will release the hold (freeing the phone line) when the caller hangs up. Note this will not help if you accidentally leave an outgoing call on hold — at present the Sipura doesn't have any good way to release outgoing calls accidentially left on hold automatically (perhaps Sipura might consider adding this in a future firmware release — it would great if they would add an "Off Hook Warning Disconnect" signal, which would be like a CPC disconnect, except that it would activate just before the Off Hook Warning Tone plays).
•Lengthen the dial tone to 20 seconds (some people find the default 10 seconds too short): Change Call Progress Tones | Dial Tone to 350@-19,440@-19;20(*/0/1+2)
•Lengthen Second Dial Tone, Outside Dial Tone, Prompt Tone, Busy Tone, Reorder Tone, MWI Dial Tone, and Cfwd Dial Tone to 20 seconds: These settings, like the basic dial tone mentioned above, are under Call Progress Tones - in all the existing strings find ;10( and change it to ;20(
•Left a phone off hook accidentally? Found that the Sipura's off hook warning tone borders on pathetic? Here's a much better one. This gives you 30 seconds of warning warble tone followed by 30 seconds of the genuine off hook warning tone used by most phone companies: Change Call Progress Tones | Off Hook Warning Tone to 480@-10,620@-16,1400@0,2060@0,2450@0,2600@0;30(.2/0/1,.2/0/2);30(.1/.1/3+4+5+6)
Note that if for some reason you don't want the warning warble (which we highly recommend because it will cause most people to hang up before being blasted with the off-hook warning), you can use 1400@0,2060@0,2450@0,2600@0;30(.1/.1/1+2+3+4)
Obviously, you should not make either of these changes if you made the change to generate the "D" touch tone prior to the Off Hook Warning Tone, as described under "Special considerations for business customers" above. [The pertinent section is copied below]
•If you have an answering machine or similar device that accepts touch tones for control functions, and you find that when you call in and try to use tones to activate the unit it does not respond properly, check under the Miscellaneous section to see what the DTMF Playback Level and the DTMF Playback Length are set to. The default DTMF Playback Level is -10.0 which is often too low, while the default DTMF Playback Length is .1 which is very often too short.
•A few people may wish to go into the individual "Line" tabs ("Line 1" and "Line 2") and go to the FXS Port Polarity Configuration section, and set Caller Conn Polarity to Reverse. All this will do for most people is give you an audible "click" when a call you place is connected, and again when the called party hangs up. What it actually happening is that the polatity of the line is reversed when the outgoing call is connected. Some advanced phone systems may be able to use this information to avoid the "line left on hold" problem, but most "hold" buttons on telephones will NOT release just because line polarity reverses (you may be able to build a circuit that responds to polarity reversals and generates a CPC disconnect signal after a polarity reversal). Note that if for some reason you want the line polarity to reverse when you are the called party, then you will need to set the Callee Conn Polarity to Reverse.
•Finally, if you have changed any of the above settings (as opposed to your service provider), you will want to make sure that you go to the "Provisioning" tab and set Provision Enable to no. However, most commercial service providers will not allow you to do this, since it prevents them from making changes to your service (which could overwrite the changes you have made above). That is why, if you want any of the above settings changed and you are a commercial VoIP service customer, it is better to get your service provider to make the changes in their system so that any future updates will not overwrite the changes.
By the way, one way to avoid a line left on hold is to not use a hold button on a telephone, but instead use "flash hold" — hit the flash button (or flash the hookswitch), then when you hear the second dial tone, hang up. Your phones will emit a short ring every few seconds until you pick the line back up, so unless you have the ringers turned off, it will be pretty hard to forget about the call on hold. This gives you a way to put a call on hold on one phone, and then pick it up at another. Some VoIP providers may disable this capability, though we don't know of any that do, and also we do not know if this works with any brand of VoIP adapter other than Sipura. One other hint appeared earlier on the page, but it's it's for business customers trying to use VoIP with a small PBX. This is slightly abbreviated from what's on the page (residential customers can probably skip this entirely, unless you have a phone with a "hold" button and someone keeps leaving the line on hold):
..... certain VoIP adapters do not generate what is known as the CPC signal. Without getting into a long technical dissertation, this is a momentary drop in power on the line (as if the line were completely disconnected for a few moments), or a polarity reversal on the line. Either of these signals can be used to cause telephone hold circuits to release automatically. The CPC signal (but not the polarity reversal) can also be useful with some consumer grade equipment, including telephones with "hold" buttons and some types of answering machines. However, most residential customers won't notice much of an impact from the lack of the CPC signal.
On a "normal" telephone line, this voltage drop or polarity reversal usually occurs if you hold the line open after the other party has hung up (to a person listening on the line, it sounds like either just a single click, or a click, a VERY short pause of "dead air", and another click. But in either case, it happens very quickly so if you're not paying close attention you could easily miss it!).
As mentioned, the purpose of this signal is to release a line inadvertently left on "hold." So, when connecting a small business phone system that is designed to work with regular business phone lines to a VoIP adapter that does not supply the CPC signal, the problem appears to be that if a call is accidentally left on hold, the line will never release. In some slightly more expensive systems, it's possible to "park" a call while waiting for another extension to pick it up, and a "parked" call might never be released without manual intervention. Thus, probably with distressing regularity, the line gets held open because someone put a call on hold and forgot about it, and with no CPC signal the line was never released.
If you think you have this problem, contact your VoIP provider — they may be able to enable CPC on your adapter. For example, Sipura adapters have CPC settings under the "Regional" tab — try setting "CPC Delay" to 10 and "CPC Duration" to 1 — your VoIP provider may have locked out these settings; if so, they would have to change these settings for you. But note that Sipura's CPC functions only on incoming calls; it will do nothing to help a situation where an outgoing call has been abandoned on hold. If you really need a CPC-like disconnect following outgoing calls using a Sipura adapter, there is one thing you could try, if you have access to the Sipura's advanced settings. Go to the "Regional" tab, under Call Progress Tones, and set the Off Hook Warning Tone to 941@-16,1633@-16,1400@0,2060@0,2450@0,2600@0;1(*/0/1+2);30(.1/.1/3+4+5+6)
This will generate one second of a "D" touch tone (941 Hz + 1633 Hz) just prior to playing the Off Hook Warning Tone. The "D" tone cannot be generated from a standard 12-button touch tone pad, but can be used directly by some types of equipment as a disconnect signal. If your equipment cannot be directly programmed to release a line upon hearing the "D" tone, you can buy a "DTMF Flash Generator" that should disconnect the line upon hearing the "D" tone (see next paragraph).
Assuming that none of the above suggestions work, we can point you to a place where you can probably find a solution to this problem, but please be aware that we've not personally tested any of this equipment. Reports from actual users of this equipment would be welcome. Go to Mike Sandman's "Wizard's Tool Box" page and scroll about ¼ of the way down the page — search for the items entitled "CPC GENERATOR", "MAKE A SILENCE DETECTOR, TO CREATE A CPC SIGNAL", and "CallSaver: Disconnects a Phone Line that's Left Off Hook!" ONE of these devices will probably solve your problem, depending on the VoIP adapter in use and how it actually reacts when a phone is left off-hook after the call is ended (for example, VoicePulse returns a fast busy signal when the other end disconnects, and Vonage sometimes does this as well, so from the descriptions given it is likely that the "CPC GENERATOR", which detects a dial tone or a busy signal, would be the device to use to create the CPC signal). If you have a Sipura adapter and can reprogram it to generate the "D" touch tone as described in the above paragraph, then the "DTMF Flash Generator" (not quite halfway down Mike's page) may be able to listen for it and generate a useable CPC signal. Sorry if this is a bit long, but it fits in this thread and I thought it might be useful for someone. | |
DracoFelis
Premium
join:2003-06-15
| reply to balta
said by balta:I've two providers ne [Provider A] gives me a PSTN number, the other [Provider B] doesn't. Nonetheless provider B has better rates, but unfortunatelly seems to require an outbound proxy. The problem I get at this stage is that GTWn dial plan aliases do not allow oubound proxies (only the proxy) part, so I can't set Provider A on line 1, and and Provider B as a dialplan gateway. Hmmm... That's a tuff one. So far the "outbound proxy" is optional with all my providers. Some of them offer it, but you don't "need it" if/when STUN is setup correctly. So I can usefully use the "gateway" options, since I leave "outbound proxy" (not to be confused with the SIP gateway) blank for all my providers.
You might try setting up your "provider B" without an "outbound proxy" (even though "provider B" tells you that you "need" an outbound proxy), but with the following STUN settings (which I use), and see if it works (worth a try, in any event): "STUN Server: stun.fwdnet.net", "STUN Enable: yes", "Substitute VIA Addr: yes", "Send Resp To Src Port: yes", and "NAT Mapping Enable: yes" on the line1 tab (with all other STUN settings off). It also wouldn't hurt to tell your router to forward your SIP ports to your SPA-3000. These are the settings I use, and they essentially make it look like (to the remote providers) like your Sipura is not behind a NAT router (even when it is). With these settings I have only needed the normal SIP proxy (not the optional "outbound proxy") with all the providers I use (which doesn't mean that your "provider B" might not be the exception to this rule). Don't know if it will work in your case or not, but it is worth a try "just in case", since if the experiment does work you have a way to put "provider B" on one of the 4 "gateway" entries (and thereby making it trivial to do what you desire)!
If the above (valid STUN settings, but no "outbound proxy") doesn't work with "provider B", my next question is: Does "provider A" give you the option to forward inbound calls to a SIP URI (internet address)? If so, I have a theory about how you might be able to use this "forwarding" to let "provider A" ring "Line 1", even if/when you put "provider B" (and it's outbound proxy) on the "Line 1" settings. Again I'm not sure if this will work or not (I can't test it by myself, so PM me if you are interested in setting up some tests for this), and it will (in any event) only work if/when "provider A" allows SIP forwarding.
Finally, you might want to look at my previously posted SPA-2000 trick for forwarding one line to the other. I have NOT tested the "forward one line to the other" trick on my SPA-3000, so YMMV. However, if that trick can be made to work on an SPA-3000, you could (in theory) put "provider A" (the inbound one) on the "PSTN" line, forward the "PSTN" line to "Line 1" (of the same adapter), and then put "provider B" (complete with it's outbound proxy) on "Line 1". I have no idea if this convoluted setup will work or not (without running some experiments), but it would (in theory) be a way to have both providers on the same adapter with different "outbound proxy" settings.
And if all else fails, you might also try asking for help in the Voxilla.com Sipura forum here: »voxilla.com/forum-viewforum-f-14.html
NOTE: If you do find a solution to the issue of different "outbound proxy" settings on an SPA-3000, please post it here. I'm sure other people would like to know what "trick" is needed to make things work in that case! | |
balta
@netvisao.pt
| Well it almost work Thanks a lot for the advance...
As per your instructions, I managed to get media through : unfortunately only one way : from home to the outside. From the phone I don't seem to send the udp packets for the sipura.
Everything works when its the main provider, so it must be some trick with the receiving port. Although I have the SIP syslogd files I can't get it clear what it might be. If you're available to help me, send me an email and I can show you the log snippet where the reason may be.
Thanks again | |
DracoFelis
Premium
join:2003-06-15
| said by balta:Well it almost work Thanks a lot for the advance... As per your instructions, I managed to get media through : unfortunately only one way : from home to the outside. From the phone I don't seem to send the udp packets for the sipura. The fact that you can talk out, but you can't hear, often means a "firewall" (router) issue. What often happens, is that the router's firewall (and NAT) "blocks" (doesn't forward to the Sipura) inbound voice packets. Since inbound voice packets are blocked at your router, you can't hear the other party. But since most routers (by default) let any packet go out, your voice packets make it though your router to the other party (and they can therefore hear you). The classic "one way audio".
First check for the obvious, and triple check your Sipura settings (including making sure that the Sipura Nat settings are "yes" for the line/gateway entries, assuming you are behind a NAT router). I remember I was getting "one way audio" (even after setting up STUN), until I discovered that I had forgot to set "NAT Mapping Enable: yes" on the "Line 1" tab! So double-check your Sipura settings first!
However, if the NAT/STUN settings are proper in the Sipura, the most likely culprit is your router's handling of inbound packets on the the SIP ports. Assuming it's a router issue, you may be able to figure out what is wrong by looking at the router logs. Does your router allow "syslog"? If so, turn on the router's log feature, and then see which packets are being dropped at the router's firewall (when you are getting the one way audio). Chances are, those are the ports you need to forward to your Sipura, to get things working... | |
DracoFelis
Premium
join:2003-06-15
| reply to MichiganTelephone
Wow! Those are some nice (and really NOT obvious) settings. I've already added several of those tips to my SPA-3000. Thanks for posting.
said by MichiganTelephone:
•Left a phone off hook accidentally? Found that the Sipura's off hook warning tone borders on pathetic? Here's a much better one. This gives you 30 seconds of warning warble tone followed by 30 seconds of the genuine off hook warning tone used by most phone companies: Change Call Progress Tones | Off Hook Warning Tone to 480@-10,620@-16,1400@0,2060@0,2450@0,2600@0;30(.2/0/1,.2/0/2);30(.1/.1/3+4+5+6)
Very nice setting. But why end the "off hook warning tone" at 30 seconds past the "warning warble"? For example, what happens if the cat nock's the phone off the hook, while you are at the store? Wouldn't you want to know about that ASAP when you get home (instead of having the "warning tone" time out after 30 seconds of really annoying sound)? After all, you can always end the annoying sound by hanging up the phone!
So I modified your suggestion, to keep the annoying tone up for 32000 seconds (a little under 9 hours of the loud annoying "you left the phone off the hook" sounds):
480@-10,620@-16,1400@0,2060@0,2450@0,2600@0;30(.2/0/1,.2/0/2);32000(.1/.1/3+4+5+6)
said by MichiganTelephone:
Sorry if this is a bit long, but it fits in this thread and I thought it might be useful for someone.
IMHO your post fits this thread very well. Thanks for tips. | |
gnexus
join:2005-06-24
| reply to DracoFelis
said by DracoFelis :The fact that you can talk out, but you can't hear, often means a "firewall" (router) issue. What often happens, is that the router's firewall (and NAT) "blocks" (doesn't forward to the Sipura) inbound voice packets. Since inbound voice packets are blocked at your router, you can't hear the other party. But since most routers (by default) let any packet go out, your voice packets make it though your router to the other party (and they can therefore hear you). The classic "one way audio". This is another reason, besides QOS, why I was thinking of getting an unlocked Linksys router. I even had this exact problem with the Vonage Linksys and my XTen softphone and had to map ports.
I suppose you don't know which version of Sipura ATA the Linksys have? I know it's certainly not SP-3000 functionality, but is it maybe SP-2001? Any help would be greatly appreciated.
Thanks again for your awesome posts. I plan to use this info when I get my ATA. | |
random1739
join:2004-12-16
Australia
| reply to DracoFelis
I've got a SPA-2000. Is there a way to increase ring tone/dialling volume from the voice box? Or doesn't the voice box have a volume control?
Currently, the dial tone is extremely quiet and difficult to hear  | |
DracoFelis
Premium
join:2003-06-15
3 edits | said by random1739 :I've got a SPA-2000. Is there a way to increase ring tone/dialling volume from the voice box? Or doesn't the voice box have a volume control? Currently, the dial tone is extremely quiet and difficult to hear Probably, but we would have to figure out the correct settings.
Based upon MichiganTelephone's post, it appears that all the "tones" are very configurable (pattern, volume, and duration) from the "Call Progress Tones" section of the "Regional" tab. The problem is, what is the format of those tone strings (and how long can the tone string fields be)? I didn't see that format info anywhere in my Sipura manual, so at the moment I can only guess.
With the proper format for that "tone info", we should (at least in theory) be able to change the tones pretty much however we want. So if anyone finds a reference to the format used by Sipura adapters to describe the "tone" entries, please post a link to that reference here.
[Edit]
Check the note (by MichiganTelephone) below! It looks like we now have the details necessary to construct the "Call Progress Tones" strings. So all that should be necessary to have a louder dial-tone, is to adjust the volume setting of the tones in the "Dial Tone:" string.
[Edit2]
It works. Use the description in the note below to find the tone volume settings (expressed as negative numbers) within the strings, and raise the volume (by lowering the negative numbers). For example, I went though and changed many of the -19 volume tone entries to -17 (on my SPA-3000), to raise the volume "just a little". Obviously you can continue to raise the volume (by using progressively smaller negative numbers) to whatever volume level you find pleasing. | |
MichiganTelephone
@anonymizer.com
| Regarding the dial tone volume: There is a volume level (I think it's in decibels, but don't hold me to that) associated with each frequency of a tone. For example, I had suggested using a dial tone value of:
350@-19,440@-19;20(*/0/1+2)
The "-19" that appears (twice) in the above string is the volume. Normally people can't hear a volume change of less than about 3 db, so you could try changing both instances of "-19" to "-16", or "-13", or any higher volume level.
Now, regarding how these strings are constructed. As best I can determine, here's how it works. Take the off hook tone I suggested (and by the way, I limited the loud warning to 30 seconds because that's what phone companies usually do these days - I think the theory is that sometimes people take the phone off-hook on purpose and they don't want to be annoyed by the tone for the entire time. After all, certain activities for which you might spontaneously take the phone off the hook might be somewhat inhibited by a loud off-hook warning going off in the background)  - recall it was this:
480@-10,620@-16,1400@0,2060@0,2450@0,2600@0;30(.2/0/1,.2/0/2);30(.1/.1/3+4+5+6)
So to dissect this:
480@-10 is one of the tones of the initial warning warble (480 Hz at -10dB??? volume)
620@-16 is the other tone of the initial warning warble
(the volume levels are different because the human ear needs the lower tone to have more energy to perceive it as being at about the same level as the higher tone).
The remaining four values are the frequencies and volumes of the loud off-hook warning. I used 0 dB because we want them to be LOUD. Possibly they could be made even louder by using positive dB values, but I didn't want to blow out a phone's earpiece (or anyone's eardrum) and 0 dB just felt and sounded like the right value.
Now, note that the given order of the tones corresponds to the number we use to indicate them later in the string. For example the 480 Hz tone appears first in the string so it is "tone #1", the 620 Hz tone indicates "tone #2", etc. The first semicolon marks the end of the tone definitions and moves us to the first of the duration specifications.
Next we have 30(.2/0/1,.2/0/2) - the 30 is the total duration in seconds. The .2/0/1 means .2 seconds on, zero seconds silence (we want the next tone to follow immediately) of tone #1 (the 480 Hz tone). The .2/0/2 means .2 seconds on, zero seconds silence, of tone #2 (the 620 Hz tone). 480 and 620 just happen to be standard frequencies used in telephony, though you could certainly use others.
Next another semicolon marks the end of that duration specification. Next we have the final duration specification, for the real on-hook blast:
30(.1/.1/3+4+5+6)
And this means we want thirty seconds (and yes, you can change that) of the four tones (#3, #4, #5, and #6), which is to say, the 1400 Hz, 2060 Hz, 2450 Hz, and 2600 Hz tones, with duration of .1 seconds on followed by .1 seconds of silence.
Note that in the case of a continuous tone, such as a dial tone that we simply want to run for 20 seconds, it's specified like this:
20(*/0/1+2)
The asterisk simply means "on for the full specified duration, zero silence"
I do not know if there is a maximum length to the string, nor if there is a maximum number of frequencies that can be specified in one command. Maybe you could make the dial tone play a little tune if you were sufficiently motivated, but my goal was to get these tones to the standards used in the U.S.A. and Canada.
I hope this helps those of you who want to customize your tones. Just remember not to get so clever that you confuse others living in, or visiting your home! | |
ags
join:2001-08-29
Alhambra, CA
| reply to DracoFelis
I have a Sipura 2002 behind a router and get an internal IP
(192.168.0.2) from that router.
I have a friend in another city who has service from Vonage. His vonage box is behind a router.
How can I call him by dialing his external IP and have his Vonage box attached phone ring and start a conversation.
To recap, How can I dial from my Sipura 2002 to connect with his Vonage box connected phone by dialing his external IP.
I do have FWD and my friend does not. It use to be FWD and Vonage has peering agrrement and I can call him using my FWD account. Unfortunately the peering agrement between FWD and Vonage has been terminated.
Any suggestion will be highly appreciated. | |
DracoFelis
Premium
join:2003-06-15
1 edit | said by ags :I have a Sipura 2002 behind a router and get an internal IP (192.168.0.2) from that router. I have a friend in another city who has service from Vonage. His vonage box is behind a router. How can I call him by dialing his external IP and have his Vonage box attached phone ring and start a conversation. Vonage used to allow people to call their customers by SIP URIs (internet addresses). However, I think they may have turned that feature off at their end, at the same time they dropped the FWD=>Vonage gateway. And since Vonage "locks" their adapters, there is no way for your friend to get into his/her adapter and override Vonage's settings.
[Edit]Does anyone with Vonage know if Vonage still allows SIP URI's for calling their subscribers? If so, what is the format of the needed SIP URI for calling a Vonage subscriber?
In the off chance that Vonage still allows SIP URI calls to their customers, and you can figure out what the proper SIP URI is for your friend's Vonage number/account, it would be pretty trivial to modify your "dial plan" (in a way similar to my FWD calling trick) to call your friend's Vonage SIP URI. But this will ONLY work if Vonage still allows this feature. If I'm correct that Vonage has turned this feature off at their end, than there is nothing you can do to override Vonage's settings (as SIP to SIP calls require both sides to allow them)!
NOTE: If Vonage has blocked things on their end, you could always try convincing your friend to sign up with FWD directly. If your friend was willing to spend $100 for a Sipura SPA-3000, (s)he could even hook up the Vonage adapter to the "Line" port of the SPA-3000. With proper SPA-3000 configuration, this arrangement (Vonage adapter off the SPA-3000's line port, and then the "phone" plugged into the SPA-3000) should allow your friend to use the same "phone" to make/receive normal LD calls via the Vonage adapter (because an SPA-3000 can "pass through" the "Line" port to the "phone") AND also make/receive calls via VoIP accounts programmed directly into the SPA-3000. Of course, once this setup is ready, it's trivial for your friend to signup with FWD, and program his/her FWD account as one of the VoIP providers in his/her SPA-3000 (allowing the two of you to chat as long as you want FWD account to FWD account)! | |
MichiganTelephone
@anonymizer.com
| reply to DracoFelis
said by DracoFelis :said by MichiganTelephone:
•Left a phone off hook accidentally? Found that the Sipura's off hook warning tone borders on pathetic? Here's a much better one. This gives you 30 seconds of warning warble tone followed by 30 seconds of the genuine off hook warning tone used by most phone companies: Change Call Progress Tones | Off Hook Warning Tone to 480@-10,620@-16,1400@0,2060@0,2450@0,2600@0;30(.2/0/1,.2/0/2);30(.1/.1/3+4+5+6)
Very nice setting. But why end the "off hook warning tone" at 30 seconds past the "warning warble"? For example, what happens if the cat nock's the phone off the hook, while you are at the store? Wouldn't you want to know about that ASAP when you get home (instead of having the "warning tone" time out after 30 seconds of really annoying sound)? After all, you can always end the annoying sound by hanging up the phone! So I modified your suggestion, to keep the annoying tone up for 32000 seconds (a little under 9 hours of the loud annoying "you left the phone off the hook" sounds): 480@-10,620@-16,1400@0,2060@0,2450@0,2600@0;30(.2/0/1,.2/0/2);32000(.1/.1/3+4+5+6) I got to thinking about this and realized that you can have a truly indefinite tone if you want - just replace the "32000" in your string above with * (a single asterisk), so it would look like this:
480@-10,620@-16,1400@0,2060@0,2450@0,2600@0;30(.2/0/1,.2/0/2);*(.1/.1/3+4+5+6)
I added a notation about this on my web page, and also a section about how to increase the volume of the dial tone and other tones. I would suggest that under the Regional tab, you look in the "Change Call Progress Tones" section and if you see any volume levels of @-19 (or anything less than @-16) you try changing them to @-16 - note there are usually two or more such values in each line. Only do that if you think the present tone volumes are a bit low (I actually do think they are a bit lower than the standard, but that's just a subjective observation) and make sure you change all the volume levels shown in each line if you change any of them.
Since readers of this forum are a bit more technically-mined than most, I'll just point out that you can probably do a lot of specific tricks with the tones if you like. For example, if you have multi-line phones in a small-office setting, you could change the frequency of the dial tone coming out of the Sipura to a much higher-pitched tone (try frequencies in the 600 to 800 Hz range, for example you could try something like 620 and 660 Hz) and train people that the line(s) with the "normal" dial tone are to be used for local calls only and all 911 calls, and the line(s) with the strange, high-pitched dial tone are to be used for all toll calls. Be careful when picking frequencies so that you don't inadvertently pick one that's too close to a frequency found in one of the touch-tone pairs, such as 697 Hz or 770 Hz. | |
DracoFelis
Premium
join:2003-06-15
| reply to DracoFelis
Calling SIP URIs via the "Speed Dials"...
Fun and games with "speed dials":
Here's another one I just figured out. The Sipura adapters have 8 "speed dial" entries (numbered 2 to 9) per line (they are on the advanced settings of the "user" tab). These settings can be set from the web interface, and then called up from your "phone" (you do need "Speed Dial Serv: yes" on the line you wish to use the Speed Dials on) by pressing a digit followed by the "#" key (for example to use speed dial 2, press "2#" on the phone).
NOTE: The "digit #" seems to work even if/when your "Dial Plan" doesn't have a translation for that pattern. Apparently, just enabling the speed dial service, is enough to let the Sipura implicitly interpret the "digit #" sequence as a request for using a speed dial.
And the really nice thing about these "speed dials", is that they not only work with real numbers (that you could key in on the phone), but also with SIP URIs (full internet addresses for a SIP phone)! For example, if you setup "Speed Dial 3:" to "613@fwd.pulver.com", then pressing "3#" on the phone will immediately give you the free world dialup "echo test", even if/when FWD isn't the provider programmed into that line!
And while the speed dials should work (to easily call a SIP URI) with pretty much any Sipura adapter, you appear to have even more options with an SPA-3000. When I tested it, I was able to use the "@GWx" syntax with the Speed dials on my SPA-3000 (on the advanced "User 1" tab of my adapter). For example, I put "18005696972@GW2" into my "Speed Dial 4:" entry (as a test), and when I pressed "4#" on the phone, the call (to an AT&T calling card access number) went through VIA my "gateway 2" provider (in my case Teliax.com). So the speed dial not only worked to call 1-800-569-6972, but it even (correctly) did the call via my "gateway 2" VoIP provider!
So while there are only 8 of them per line, do remember the Sipura "speed dials", for when you have a specific SIP URI (internet address) you would like to call. Not only is it easier to put a specific SIP URI into a speed dial then it is to make a "Dial Plan" translation for that URI, but it also doesn't clutter up your "Dial Plan" with translations for single SIP URIs.
And even if your Sipura is "locked", the provider may let you into the "Speed Dial" entries of the "User x" tab (even if/when the provider doesn't let you modify the "Dial Plan"). If so, you might be able to enter in your own speed dials, and thereby call any SIP URI you like, even on a supposedly "locked" Sipura adapter...
Voice chatting/conferencing with other BBR VoIP members:
I was thinking. Is anyone interested in having a BroadBandReports VoIP "conference room", for when we want to chat with each other (instead of just typing at each other)?
It occurred to me that SIPphone allows you to setup your own voice conferencing for free, you just have to pick your own 7 digit conference room number (and then use it, no signup required). And since SIPphone lets you call any of their numbers (including their conference rooms) via SIP URI, you don't even need to sign up with a free SIPphone account to join in the fun. So what do people think about meeting at SIPphone conference room 227-8647 (aka BBR-VOIP)? All anyone would need to do to "join in", is to make a VoIP call to SIP URI: 12222278647@proxy01.sipphone.com
Of course, the easiest way for us Sipura owners to call a SIP URI, is to just put that URI into a "Speed Dial" entry (and make sure the speed dial service is enabled). For example, if you have "Speed Dial 2: 12222278647@proxy01.sipphone.com", then merely pressing "2#" on your phone would allow you to join the conference. | |
DracoFelis
Premium
join:2003-06-15
1 edit | reply to DracoFelis
Blocking costly area codes via your "Dial Plan".
Considering how easy it is to dial high cost regions, and considering that some providers (including the DialPad.com I use) do NOT have an option to block costly calls from their portal, I thought I would put some number "blocking" into my Sipura dial plan.
While blocking "international dialing" is pretty obvious (just don't allow more than 11 digits of dialing), some countries near the USA (for example the Bahama's) are considered costly international calls, yet they dial as if they were USA numbers (ouch)! So I googled for a list of area codes, and used that info to construct a list of dial plan patterns that my Sipura should disallow/block (Sipura adapters let you disallow a pattern by putting an "!" at the end of the pattern string). I make no promises that this list is accurate/complete (feel free to modify it, and post those modifications to this thread, if you find something wrong). And I also made no effort to block Canadian numbers (as many providers, including mine, charge Canada numbers the same as the USA).
However, FWIW here is my 1st attempt at a "costly area code block list". If you want to use this list, I would suggest putting it as the first entry in your dial plan, so that the entries in this list override any entries that allow normal 11 digit dialing.
124[26]x.!|1441x.!|1456x.!|1473x.!|1555x.!|1590x.!|1649x.!|1758x.!|1767x.!|1809x.!|
126[48]x.!|1[27]84x.!|134[05]x.!|16[68]4x.!|186[89]x.!|1[89]76x.!|1[79]00x.!|[34678]11! | |
WhyADuck
Premium
join:2003-03-05
1 edit | reply to DracoFelis
Re: [Equipment] Useful Sipura tricks...
I just came across a Product Review of the Sipura Analog Telephone Adapter (SPA-2100) at Tom's Networking and it contains an interesting evaluation of the quality of service (QoS) features. According to the test, it would appear that much better results are obtained with QoS set to TBF (Token Bucket Filter) as opposed to CBQ (Class Based Queueing) or QoS disabled. The summary states, "Enabling TBF-based QoS (Sipura's recommendation), essentially eliminates packet loss, but doesn't do much for jitter. On the other hand, switching to CBQ-based QoS appears to increase packet discards, but significantly lower jitter." Apparently packet discards cause greater voice degradation than jitter.
Those of you who are interested in QoS on the SPA-2100 will probably find the test results and associated comments interesting. And yes, I'm aware that the QoS can't control what happens to your packets after they get out onto the 'net, but if you do any significant amount of uploading, it could definitely make a difference. | |
DracoFelis
Premium
join:2003-06-15
| reply to DracoFelis
Voice encryption...
I currently have no way of telling how good the encryption built into the Sipura adapters is (so it may be easier to "crack" then people believe). And furthermore, the only place I've found to get the encryption keys is the Voxilla.com web site (and no matter how secure a web site is, it is still a "3rd party" that has access to your keys, and from a security/encryption standpoint that is "a bad thing"). And finally, encryption will only work with other sites/adapters that support encryption, which most of them don't (so most of your calls will still probably behave the same as if you didn't have encryption enabled).
So all things considered, I wouldn't trust Sipura "secure calls" (encryption) to keep people from "listening in" on your calls. However, it might "slow them down" some, and IMHO you have nothing to lose by enabling this feature. After all, if the encryption works, you may have prevented some "eves-dropper" from intercepting you call. But even if the encryption fails, you are no worse off then before (because the "normal case" is to send your voice "in the clear"). With that in mind, here is how I just enabled voice encryption on my Sipura adapters (tested by calling between my SPA-3000, and my older SPA-2000).
1) Sign up with a free account at »www.voxilla.com. This is necessary, as the only place I am currently aware of that allows you to get the encryption keys is the voxilla web site, and they require you to be a "member" to run their "wizards".
2) Go to the Voxilla Sipura encryption Wizard. You can either find the link (on the left hand panel) at the main voxilla web site, or the current "direct link" is at this URL: »voxilla.com/certrequest.php
3) On the above web page, completely fill out the form. Apparently the form will fail unless ALL of the field (including the "Your name or alias" field) are filled in. In the case of the "Your name or alias" field entry, if you don't want to fill it in, do what I did, and just use a single space character as your "name". This Wizard will allow you to push a set of keys (public and private) to your Sipura. Don't forget that this Wizard needs to be run once for each "Line" that the Sipura has (for example, my SPA-2000 has two "lines", and each line needs a different "key").
4) Check to make sure that the Voxilla encryption wizard pushed encryption keys to your adapter. You can verify this by looking at the (admin login, advanced) "Line x" tab fields: "Mini Certificate:" and "SRTP Private Key:". If these fields are still empty/blank (the default for Sipura adapters) than the Wizard didn't do its job. However, if the "Mini Certificate:" has a bunch of characters in it, and the "SRTP Private Key:" shows "*************" (indicating that something hidden is in that field), than the public/private keys were entered into your Sipura (which is what you want to have happen).
5) The Voxilla wizard suggests you enter "*18" to do a "secure" call. However, why would you want to bother with that? Wouldn't you want the Sipura to just "default" to "secure" mode when it can? To make "secure"/"encrypted" calls the default (while still allowing other calls when encryption isn't available), go over to the (admin login, advanced) user tab for the line, and change "Secure Call Setting:" to "yes".
6) At this point, the Sipura should work the same as it did before, EXCEPT when you call DIRECTLY (not via a 3rd party) some location that supports encrypted calls (for example, another Sipura with this feature enabled). When encryption is supported (by both sides), the Sipura appears to take an extra second or so to initially connect, and then beeps at you three times (to let you know that the call is "secure"). I also noticed a little extra (maybe 1/3 second?) latency/lag in the call, but the sound was otherwise "clear" when I tried this on my LAN between my SPA-2000 and SPA-3000.
NOTE: I have not yet had an opportunity to test encryption with "Free World Dialup" (so YMMV). But according to posts I've seen in the past, FWD does support (pass though) Sipura voice encryption when all of the following are the case: 1) Both parties (the caller and the called party) in the call have Sipura adapters with encryption keys installed (and remember they are NOT installed by default, you have to use the Voxilla wizard to get them). 2) Both parties are on FWD directly, not via some "peering partner". 3) Both parties are using "fwd.pulver.com" as their proxy (i.e. neither party is using the alternate "fwdnat.pulver.com"). and 4) the party making the call has told their adapter to make a "secure call" (for example by having "Secure Call Setting: yes").
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