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| reply to DracoFelis
Re: [Equipment] Useful Sipura tricks... Here's how to take control of how often your adapter "registers":
I'm sure you all know about the "Register Expires:" parameter (on the "Line x" tab). This is supposed to control how often your adapter registers (the parameter is in seconds).
However, I have been trying to figure out why the registration interval could sometimes be a lot shorter, or a lot shorter than I supposedly set in "Register Expires:". I finally found some of the other settings (and other factors that effect registration). Here they are (as best I can figure):
It appears that the SIP provider you try to contact can override the registration interval (by the SIP messages they send when you try to register). This can greatly change how often you thought you would register...
To take control back, here are some other settings to adjust:
You can specify a MINIMUM normal registration interval, below which the provider can't register more often, by setting the "Reg Min Expires:" value (advanced "SIP" tab).
You can specify a MAXIMUM normal registration interval which you will always reregister by (even if the provider wants to register less often) by setting "Reg Max Expires:" (again, SIP tab).
In addition, the two parameters "Reg Retry Intvl:" and "Reg Retry Long Intvl:" (also "SIP" tab) seem to control how long you wait to try registering again, if a registration attempt "fails". This can be important, as you often want to try again "reasonably soon" if/when the registration has a problem and "fails". Because while you aren't registered, you aren't receiving inbound calls (and with some providers, you also can't make outbound calls when not registered). |
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| reply to DracoFelis
Hi,
I have just bought SPA 3000. I have two SIP accounts and need help as how to set-up the two to ring to one phone.
Thanks |
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ebruce
join:2001-12-04
East Falmouth, MA | said by halalani:Hi, I have just bought SPA 3000. I have two SIP accounts and need help as how to set-up the two to ring to one phone. Thanks If I'm not mistaken, you can only have 1 incoming SIP provider (and one incoming PSTN) with a SPA-3000. It only allows for multiple outgoing SIP providers. |
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| reply to halalani
said by halalani:I have just bought SPA 3000. I have two SIP accounts and need help as how to set-up the two to ring to one phone. Reminder:
This is NOT a questions thread, this is a "tricks" (solutions) thread. Please start new threads for questions, and only post "useful tricks" here.
That said, here is the "tricks" you can use to do this:
1) If both providers require you to be "registered" (to receive inbound calls), you are SOL, as the SPA-3000 only gives you one "registered" inbound VoIP line for receiving calls (the PSTN VoIP provider is "special", and can NOT be used for normal inbound calls).
2) However, if one provider will accept inbound "SIP URI" (VoIP address) calls, and the other provider will let you "forward" to a SIP URI, than it is easy. In that case, have the one provider "registered" as your "Line 1" provider, and have the other provider forward to the 1st provider. Voila, both will "ring your phone" (one directly, and the other via the forwarding).
3) If at least one of the providers allow you to forward to a SIP URI, but the other provider won't accept the inbound call, than you can try setting up this "trick" to forward directly to your adapter. It's a PITA to do, but it does work nicely once you get all the details right:
»faq.sipbroker.com/tiki-index.php···20Sipura.
NOTE: The above details all talk about getting a 2nd/3rd/etc provider to "ring your adapter". However, since you only have one main provider on the SPA-3000, you also have to use some "trick" to call out via those other providers. The easiest way to handle that, is the SPA-3000 "gateway" slots. Please look at older posts in this thread for details about how to properly use the SPA-3000's "gateways"... |
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 Huh
@cegetel.net | reply to DracoFelis
Does the SPA-3102 improve upon "two registered providers not allowed" problem? |
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 SuperCPAPremium
join:2002-11-28
Dayton, OH | reply to DracoFelis
SPA2K and SPA3K
I’m not sure if this qualifies as a “trick”, or just a combination or iteration of utilizing the power of the SPA’s.
My office setup. Each of the SPA’s is connected to a two line phone, and the SPA’s are on the same LAN. I have a fixed IP for a WAN address, and set the SPA’s also to a fixed internal IP. A variation of this procedure may also work with a dynamic WAN IP if utilizing something like DynDNS.org, if supported by your router.
The SPA3K is set up with Broad Voice on Line 1, port 5060, and Delta Three on the PSTN Line, port 5061. The SPA2K has FWD on line 1, port 5062, and Voice Pulse Connect on Line 2, port 5063. Only the Broad Voice line accepts an incomming call. The other lines are outgoing only.
The objective is effectively a direct dial hot line form the phone attached to the SPA2K to the SPA3K PSTN Line.
On the SPA3K SIP Tab, enable “Send Resp To Src Port” down by the STUN settings. I’m not sure if the is required, but it allows both calls to my FWD and IPKall numbers to ring the SPA3K Line 1 directly.
On the Line Tab’s of each ATA adapter ensure that “Use DNS SRV” and “DNS SRV Auto Prefix” is set to no. I also have enabled “Make Call Without Reg” and “Receive Call Without Reg” on both adapters.
The real power of both the SPA’s is in the dial plan.
For the SPA3K, add:
to the Line 1 dial plan. On the PSTN Line, add:
to “Dial Plan 1”. You could use any default area code. On the VoIP To PSTN Gateway Setup, set “One Stage Dialing” to yes. Set the “Line 1 VoIP Caller DP”, “Line 1 Fallback DP”, and “VoIP Caller Default DP” to 1. Add your WAN IP address to the “VoIP Access List”.
For the SPA2K, for the Line 1 and Line 2 dial plans add:
<#0:><:User ID SPA3K PSTN Line@Your WAN IP Address:5061>
911<:User ID SPA3K PSTN Line@Your WAN IP Address:5061>.
The result is that dialing #0 from any of the three VoIP Lines results in a PSTN dial tone, and all calls to 911 from the SPA2K are now automatically dialed from the PSTN Line on the SPA3K. I did make a test call to 911 form the SPA2K, and it does work.
I’m still thinking about PSTN Line access security, but for me it’s generally not an issue, as there is no long distance provider set up through the phone company, and the dial plan restricts PSTN calls to the local area code. It works for me, my office set up, and the way that the phones are used. Any comments, criticism or suggestions would be appreciated. |
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| reply to DracoFelis
I'm using an SPA-2100 on ZingoTel. At first, I was quite happy with the box since calls sounded great. I later was running some speed tests on my cable connection using »www.speakeasy.net/speedtest/ and I noticed that I was getting very poor performance on my upload speed. When I went to my cable tech support, we unhooked the SPA-2100, and damned if the cable speed didn't pop up to what it should be. The SPA-2100 was eating up 2/3 of my cable connection bandwidth!!!
I called and talked with ZingoTel, and it turns out that the QoS (Quality of Service) feature in the SPA-2100 (turned on by default) does indeed eat up the bandwidth coming out of the box. I even found some references on the web to this issue:
(»faq.sipbroker.com/tiki-index.php···SPA-2100)
Under Sipura Flaws, I found: "QoS reduces available bandwidth even when there is no active phone call. I don't think there is any good reason to reduce available bandwidth except during an active phone call. (However, ALL the devices I know of that implement QoS do it this way, so it's not just a Sipura problem.)"
The only way to beat this was for them to turn of the QoS (which I did), and then my bandwidth went back to normal. Problem was, the sound on the calls wasn't very good. So I decided to upgrade my cable bandwidth in the hopes that would help. It didn't help, but now I'm addicted to the higher speed! LOL!
Now to be clear, I had the SPA-2100 set up in a typical manner where it was connected between my cable modem and my PC, acting as a router. I suffered with this for a while, but then about a week ago, I bought a wireless router. My girl friend wanted to be able to connect with her laptop when she was at my place, and I liked the idea of a hardware firewall over a software firewall, so I got the new router.
So now I had the cable modem, then the SPA-2100, then the wireless router, and then my PC, all daisy chained together. All was working good except the quality of the phone service sucked.
Well I decided something just wasn't right, and maybe I needed a different setup since this just wasn't getting the job done. So I called back to the ZingoTel folks to try to get them to send me a different ATA box, but this time I got someone in tech support that was a bit more knowledgeable. He told me that if I connected the SPA-2100 to my new router and also connected my computer to the router (instead of the back of the SPA-2100), that my speed wouldn't be killed by the QoS, the SPA-2100 would work great, and we could turn the QoS back on for good quality calls. He was exactly right. It now works perfectly and I have great sound and good speed.
It turns out that when you connect your PC to the router built into the SPA-2100 box (as you're typically instructed to do), the QoS severely limits the bandwidth that gets through to your PC. However, if you instead have another router available to you, just connect both the SPA-2100 and the PC up to that router, and all will be good.
Once caveat. He did tell me that I'd need to adjust the router's firewall setting for the SPA-2100. This is done by first going to the phone, dailing **** and then 110#. This will give you the IP address that the router has assigned to the SPA-2100. Then go into the router set up, and add that IP address to the router's DMZ so that the SPA-2100 is now outside the router's firewall.
I'm not sure this is really necessary since the phone works fine without doing it. My guess is that he wanted me to do this is only so that their tech support can talk to the SPA-2100 without my router firewall getting in the way. I've actually turned that off since I think it's good that the SPA-2100 is behind my firewall, too. However, I may need to enable it if I talk to their tech support in the future.
Hope you find this useful.
John |
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| reply to DracoFelis
Thanks to all for the useful tips. I was forced to solve a few problems that I could not find mentioned, I hope these tips are helpful.
I have an SPA3000 (Linksys version) on Broadvoice. I am using a analog POTS (Plain old telephone service) line for local calls and 911, I am using BV/VoIP for LD calls.
(*xx|#xx|91xS0|[346]11|0|00|[2-9]xxxxxxS0|1xxx[2-9]xxxxxx.|xxxxxxxxxx.)
I ran into some "interesting" problems where calls from FXS/LINE1 to the gw0/FXO/PSTN would end up with a dial tone or sometimes a recording. In this dial plan 7 digit calls or 91x calls.
Upgrading to the latest SW (BV is one version behind) did not help. After much research, I determined the problem was the DTMF Playback level default is too high for my analog line. Adjusting to -3 to -5 allowed for all local calls to work correctly.
In the process of debugging this issue I changed the "PSTN Answer Delay:" to 1 and forgot to change it back to the default of 16. This created a similar but unrelated issue when receiving calls on the PSTN. Calls were correctly transferred to line one but callers would hear a dial tone and the FXS line would ring once or not at all. (I think this tone is the VoIP1 interface dial tone), either way this issues were addressed by increasing this delay to something like five+ rings or 16+ seconds.
The fine posts regarding BV were on target, I chose them over Sunrocket because they support for BYOD and they were on board of providing me the admin PWDs.
One final item, I ran ping tests to all of the BV proxies and changed their setting from NYC to MIA, the proxy that had the best ping times for me. Call quality was noticeably improved. An easy optimization if your VoIP provider has multiple proxies.
I hope these tips help as much as many of the previous posts helped me.
Thanks again
Doug |
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| reply to SirRealNWGA
Re: [Equipment] Useful Sipura tricks... the only ports Zingotel uses to get into the device are either 80 or 8765. The 10000-20000 UDP, 5060 UDP, and 443 TCP are fowarded so packets always reach their destination and don't get caught up and lost on your network. These ports are usually the same no matter what VoIP service you use. |
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| reply to DracoFelis
said by DracoFelis:said by Pelikano :
Enable IP Dialing: yes
If you are going to suggest a setting, please also include a short description as to why you want to use that setting. Otherwise, nobody knows the context of your "trick". I am glad Pelikano suggested Enable IP Dialing: Yes
I have 2 PAP2. One has regular PAP2 fw, and the other has SPA 1001 fw. I have no problem of calling a sipphone.com conferance feature using speed dial (e.g. 12226253703@proxy01.sipphone.com on Speed dial 5) from SPA 1001, but not PAP2. All settings are the same. So, I wasn't able to understand why SPA 1001 calls 12226253703@proxy01.sipphone.com but not PAP2.
Right after Enable IP Dialing: Yes , I was able to call from PAP2. So I suggest every one to do the same if they are having problems making IP calls.
--
SUKRU BEY, a Linksys PAP2 user |
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| reply to ctylor
This tip (see cytlor's post on page 8) should be a sticky. Thanks ctylor.
I couldn't get the RTP packet size down to 0.03; had to settle at 0.02 with latency of medium.
Made a huge difference in quality of calls.
Thanks again |
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cwwny
join:2006-07-25
Brooklyn, NY | reply to DracoFelis
I have a Linksys PAP2 device and followed DracoFelis directions to get forward all inbound calls to line2 from line1. Please help......
The phone on line2 actually rings but when I pickup the caller's device is still ringing. So my line1 provider is forwarding my line2user@externaldnsname:5061 to line2 and the phone ring but why does it not pickup?
I have every settings enabled that is documented. The following is my SIP settings and Make Call Without Reg and Ans Call Without Reg both enabled.
Reg Retry Intvl: 30
Reg Retry Long Intvl: 300
Handle VIA received: yes
Handle VIA rport: yes
Insert VIA received: yes
Insert VIA rport: yes
Substitute VIA Addr: yes
Send Resp To Src Port: yes
STUN Enable: yes
STUN Test Enable: yes
STUN Server: stun.sipgate.net:10000
NAT Keep Alive Intvl: 15 |
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| Does anyone know any tricks to use the remote sipura page, the sipura I have is remote on a slow speed link, when I try to access the sipura page, it opens and cant configure as jumping to different screens takes a while and times out, have used IE, mozilla |
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| > DracoFelis : since this thread is getting pretty long, to the point where I can't even go through all of it to find out if someone knows the trick to configure the Linksys to have it dial out a remote IP phone on the Net directly... maybe it'd be a good idea to extract the tricks into a single document and make it sticky? |
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maziloFrom MaziloPremium
join:2002-05-30
Lilburn, GA
kudos:1 | said by vincentdelporte :
maybe it'd be a good idea to extract the tricks into a single document and make it sticky? Well, this may not be a bad idea if you start it out  .
--
Mazi (UK Non-Geo Phone: +44-703-194-2574) |
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| reply to DracoFelis
Re: How to do an SPA-3000 setup like mine... Those screenshots came in handy. I had made a bunch of useful changes the other day, but for one. Someone had written that a good value for DTMF Playback Level was -1. Well, VOIP calls worked fine. The next day, though, when I made an FXO call, I got a long dialtone AFTER dialing the number. Looking at your regional tab, I changed the value to -9 and that solved the problem. I don't see your PSTN tab values, though!!! [I'm going to print out those screenshots you posted.] Can you post a screenshot of your PSTN tab? |
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