[Other] Questions/Issues Regarding Voip.ms VOIP Service

[Other] Questions/Issues Regarding Voip.ms VOIP Service
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anon @ 15th Sep 02:35AM:
[Other] Questions/Issues Regarding Voip.ms VOIP Service

I have a few questions about my new Voip.ms account and would appreciate any response :). FYI, I am using a Linksys PAP2 device loaded with 3.1.23(LS) firmware.

1. Is there a way to block my Caller ID? I have tried *67xxxxxxxxxx with no luck. I sometimes get a busy tone. The weird thing is after I call my cell phone with *67 prefix, and hang up, I sometimes receive another call with blocked CID from my Voip.ms phone without even me initiating the call!?!? :huh:

2. Has anyone experienced a busy tone between two outgoing calls in a row? Say, you dial a number and hang up after a few seconds, then decide to call that number again within a 3 or 4 sec. interval. The call will not go through...just a busy tone.

3. Is there a way to customize the voicemail's greeting message?

4. In PAP2 web portal, under "Line 1" in the "Display Name" field, can I leave it blank or have my phone number there instead of my name?
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anon @ 15th Sep 07:44AM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by Typhooon :

I have a few questions about my new Voip.ms account and would appreciate any response :). FYI, I am using a Linksys PAP2 device loaded with 3.1.23(LS) firmware.

1. Is there a way to block my Caller ID? I have tried *67xxxxxxxxxx with no luck.

I have it set in 'account settings', but Darkev said in his review of voip.ms that he has it working with *67. Maybe he will pop in with a tip.

2. Has anyone experienced a busy tone between two outgoing calls in a row? Say, you dial a number and hang up after a few seconds, then decide to call that number again within a 3 or 4 sec. interval. The call will not go through...just a busy tone.

Haven't seen that with voip.ms.

3. Is there a way to customize the voicemail's greeting message?

When you call your mailbox to play back messages, there is also a menu choice that will allow you to record your own greeting. I think you will also want to change the settings in manage did / voicemail to turn off the default greeting.

4. In PAP2 web portal, under "Line 1" in the "Display Name" field, can I leave it blank or have my phone number there instead of my name?

Either one will work. I have a combination of letters and numbers in mine.
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PX Eliezer @ 15th Sep 08:51AM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by Typhooon :

In PAP2 web portal, under "Line 1" in the "Display Name" field, can I leave it blank or have my phone number there instead of my name?

Just be aware that it doesn't really get you much. This "Display Name" will only appear to the other party if you are calling another Voip.MS customer, or someone within your LAN, or if you are calling another VOIP customer by direct SIP calling (including also Sipbroker).

The "Display Name" on the PAP2T or other adapter has no impact on how you show up when calling someone on the regular phone network (POTS/PSTN), except possibly in some Canadian situations.
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PX Eliezer @ 15th Sep 08:59AM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by Typhooon :

Has anyone experienced a busy tone between two outgoing calls in a row? Say, you dial a number and hang up after a few seconds, then decide to call that number again within a 3 or 4 sec. interval. The call will not go through...just a busy tone.

This is not surprising at all in any VoIP situation. VoIP call disconnection is not as efficient as in traditional telephony. This also relates to CPC duration and other settings on your adapter.

Geez---you like to take coffee breaks. Let the system rest for a few seconds!
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Mango @ 15th Sep 03:28PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by Typhooon :

2. Has anyone experienced a busy tone between two outgoing calls in a row? Say, you dial a number and hang up after a few seconds, then decide to call that number again within a 3 or 4 sec. interval. The call will not go through...just a busy tone.

As far as I can tell, this is due to a DNS bug with Linksys devices. Both my PAP2T (with different firmware from yours, so that's not the problem) and my SPA921 exhibit this behaviour. I do not know why more people haven't reported this. I have tried using VoIP.ms' DNS server, my ISP's DNS server, and a local DNS server I set up on my internal network and the problem still occurs. The only way I have found to fix it is to set the proxy to the IP address of the SIP server, eg, 67.215.241.250 instead of sip.us3.voip.ms. The IP addresses of the severs may be found on the Account Information page at VoIP.ms.

If anyone has a better solution I would be very interested to hear it.

said by Typhooon :

4. In PAP2 web portal, under "Line 1" in the "Display Name" field, can I leave it blank or have my phone number there instead of my name?

Sure, you may put anything you want in there. Mine currently says "It's Mango!". This amuses my friends.

I see that your internet provider is Novus so I suspect that you live in Vancouver, BC. If this is the case, you should have fairly good success using Caller ID Name if you use premium routing at VoIP.ms. I say "fairly good success" because it doesn't work 100% of the time. Presumably VoIP.ms has multiple carriers for outgoing calls and only some of them support Caller ID Delivery for Canada.

Of course, if this isn't what you want, then by all means feel free to put something other than your name there. I'm not sure how to use per-call Caller ID Number Blocking, but removing my number from "CallerID Number" in the VoIP.ms portal caused it to be blocked.

If you don't mind checking for me, what is your ping time to sip.us3.voip.ms when using Novus?

m.
--
Mango's recommended PAP2T settings
Linksys/Cisco dial plan tips and tricks
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Typhoon @ 16th Sep 08:57PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

^^ I just pinged their US3 server; the latency is 48ms - quite high.

How's the call quality via the Value routing for you? Mine is crappy! I have already emailed them about it.

Have you noticed any difference in using different DNS servers? I am using public DNS' 208.67.222.222/20 as my primary and secondary servers, I don't know if I should change them.
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PX Eliezer @ 16th Sep 10:22PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by Typhoon :

How's the call quality via the Value routing for you? Mine is crappy! I have already emailed them about it.

How is your call quality with Premium routing?
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Typhoon @ 17th Sep 12:06AM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by PX Eliezer :

How is your call quality with Premium routing?

I haven't thoroughly tested their Premium routing yet. Based on my initial tests, it seems to be much better than the Value routing. It also handles DTMF tones better.
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Mango @ 17th Sep 01:38AM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by Typhoon :

Have you noticed any difference in using different DNS servers? I am using public DNS' 208.67.222.222/20 as my primary and secondary servers, I don't know if I should change them.

I believe my post above addresses this.
--
Mango's recommended PAP2T settings
Linksys/Cisco dial plan tips and tricks
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Typhoon @ 17th Sep 02:01PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

After all the issues that I had with the Value routing, I switched to premium routing last night. Everything seems to be honky dory (at least for now) except for the incoming calls. Since last night, when calling my voip.ms line I sometimes get either a busy tone or no answer on the voip.ms line - the phone doesn't ring. The weird thing is the call would show up in the CDR page!? Anyway to fix it?

The other issue the Failover feature: to test, I disconnected my PAP2 from the Internet, and then called the Voip.ms line. I get a busy tone after 20 sec. of silence. I have set the failover to rout calls to my cell phone in case of failure.
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shighfield @ 17th Sep 02:47PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

I used to use Premium routing for everything at voip.ms just so I wouldn't have any issues, and about a month ago I switched to value just to see what it was like and haven't switched back.

I have found sometimes when I get a call the other end says I sound like i"m in a tin can but that is about 1 in 50 calls and if I call them back or if they call me right back it's fine.

If I'm making a very important call for work at home I may switch to premium just in case(tm) but so far I haven't had to.
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soitgoes2 @ 17th Sep 02:50PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by shighfield :

If I'm making a very important call for work at home I may switch to premium just in case(tm) but so far I haven't had to.

You can always use per-call override (044+Country Code+number for premium; 033+Country Code+number for value)
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Mango @ 17th Sep 04:15PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

Does that work for you for North America calls? It doesn't seem to for me.

m.
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soitgoes2 @ 17th Sep 04:18PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by Mango :

Does that work for you for North America calls? It doesn't seem to for me.

Maybe not. It does say international on the site.
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Typhoon @ 17th Sep 04:34PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

Any one experienced similar issues as in below?

Copy/Paste from my previous post:

"After all the issues that I had with the Value routing, I switched to premium routing last night. Everything seems to be honky dory (at least for now) except for the incoming calls. Since last night, when calling my voip.ms line I sometimes get either a busy tone or no answer on the voip.ms line - the phone doesn't ring. The weird thing is the call would show up in the CDR page!? Anyway to fix it?

The other issue the Failover feature: to test, I disconnected my PAP2 from the Internet, and then called the Voip.ms line. I get a busy tone after 20 sec. of silence. I have set the failover to rout calls to my cell phone in case of failure."
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MartinM @ 19th Sep 11:10AM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

Value/Premium doesn't affect inbound (Incoming).

If you are simply pulling out the chord from the Adapter, it can explain the issue you are having with the failover, if you are behind nat, your router is basically acting like a firewall, if it drop the incoming SIP packets instead of rejecting, the call will "hang" and retry on our side a few times.

Solution to this is
1. Do not just pull off the chord
2. Lower the registration time to something like 2 minutes, pulling the chord will NOT unregister your device, but if it's set to a low amount, it will be unregistered on our side after 2 minutes, and your fail over will work properly.

*67 is currently a feature that is Device Side. Linksys support it normally. It's not a voip.ms feature. basically what it will do is set your Linksys to mask the username as Anonymous or blocked ( I don't remember exactly). You should review the feature codes of your adapter and make sure your dial plan allows *xx to be dialed

To determine if the quality issue of incoming calls is caused by the number (it rarely is the number, but it can happen, everything is possible), you can point the DID to our echo test system, and call the number from another line outside VoIP.ms. When the echo test instruction message is done, speak, what you will say is repeated back to you as soon as it is received. This is the best way to the determine if the DID issue is locate between you and our server, or between us and our provider.

Don't hesitate to contact our support again.
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Typhoon @ 19th Sep 02:44PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by MartinM :

Value/Premium doesn't affect inbound (Incoming).

If you are simply pulling out the chord from the Adapter, it can explain the issue you are having with the failover, if you are behind nat, your router is basically acting like a firewall, if it drop the incoming SIP packets instead of rejecting, the call will "hang" and retry on our side a few times.

Solution to this is
1. Do not just pull off the chord
2. Lower the registration time to something like 2 minutes, pulling the chord will NOT unregister your device, but if it's set to a low amount, it will be unregistered on our side after 2 minutes, and your fail over will work properly.

*67 is currently a feature that is Device Side. Linksys support it normally. It's not a voip.ms feature. basically what it will do is set your Linksys to mask the username as Anonymous or blocked ( I don't remember exactly). You should review the feature codes of your adapter and make sure your dial plan allows *xx to be dialed

To determine if the quality issue of incoming calls is caused by the number (it rarely is the number, but it can happen, everything is possible), you can point the DID to our echo test system, and call the number from another line outside VoIP.ms. When the echo test instruction message is done, speak, what you will say is repeated back to you as soon as it is received. This is the best way to the determine if the DID issue is locate between you and our server, or between us and our provider.

Don't hesitate to contact our support again.

Hi Martin,

Lowering the Registration Expiry time to 300 has fixed the failover issue. Also, I have not experienced any dropped calls after setting the ET time to 300, and I hope that never happens again.

With regards to CID block feature, I realised that this feature only works through the Premium routing. Maybe it's just me, but it doesn't work via the Value routing.

DTMF tones, also, don't work properly via the Value routing.
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PX Eliezer @ 19th Sep 04:24PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by Typhoon :

With regards to CID block feature, I realised that this feature only works through the Premium routing. Maybe it's just me, but it doesn't work via the Value routing.

DTMF tones, also, don't work properly via the Value routing.

What setting do you have for DTMF?

With Voip.MS this needs to be set on the Account Settings page on their website (dashboard) in addition to being set on your ATA.

Try "Auto" if you haven't already.
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Mango @ 19th Sep 05:20PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

If "Auto" doesn't work, try setting it to "Info" for value routing.
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Arne Bolen @ 19th Sep 07:10PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by MartinM :

*67 is currently a feature that is Device Side. Linksys support it normally. It's not a voip.ms feature. basically what it will do is set your Linksys to mask the username as Anonymous or blocked ( I don't remember exactly).

*67 does not work if there is a number in the "CallerID Number" field in "Edit Sub Account". If there is a number in this field it will be sent no matter what the Linksys device says.

I have just tested this with my Linksys SPA962. I know the *67 is working on my Linksys SPA962 because when I tested it with another provider my number was blocked and *68 unblocked my number again.

*67 will not work with VoIP.ms because the number in the "CallerID Number" field will always be shown to the other party. The only way to block the number is to delete the number in this field.

If I ever would like to block my Caller ID I would just delete the number in the "CallerID Number" field. But it would of course be better if VoIP.ms could get *67 and *68 to work serverside because it is not practical for everyone to log in and delete the number in the "CallerID Number" field. Reseller and subaccount users would not be able to log in and delete the number.
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PX Eliezer @ 19th Sep 07:58PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by Arne Bolen :

*67 will not work with VoIP.ms because the number in the "CallerID Number" field will always be shown to the other party. The only way to block the number is to delete the number in this field.

Mmmph. Thanks for the heads up. I honestly never realized that. Guess they may want to consider adding that in a future upspiff.

It should be possible. *67 (Caller ID blocking per call) and *69 (Call return) work fine with one other provider I am familiar with.

--------------------------------------

It just goes to show, no single company has a complete set of features.
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MartinM @ 20th Sep 11:10AM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

I filled in a suggestion to block the number when we receive "blocked", "unavailable", "anonymous" etc, in our website, and asked our dev to do it next week. I highly suggest everyone to use that feature, we do read everything and accept quite a lot of suggestions.

Thanks for the feedback.
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Arne Bolen @ 20th Sep 11:13AM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by MartinM :

I filled in a suggestion to block the number when we receive "blocked", "unavailable", "anonymous" etc, in our website, and asked our dev to do it next week. I highly suggest everyone to use that feature, we do read everything and accept quite a lot of suggestions.

Thanks a lot Martin.
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MartinM @ 20th Sep 11:13AM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by PX Eliezer :

It just goes to show, no single company has a complete set of features.

We know you love the C letter :) No company is complete of course, but by the end of the year, the one you are refering to is gonna lag behind.
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mshe @ 21st Sep 02:08AM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

I have this issue with a 866 conference bridge I call daily - I have to try the bridge 3 - 4 times before it'll connect. Tech support was unable to replicate, but I can do so easily.

In the CDR reports, it shows up as "failed" on voip.ms side. No idea why.
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hockeynomad @ 26th Sep 03:44PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

Not sure what he means here, but I am easily able to use the *67 and *69 features, and the CDR verifies that.
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Typhoon @ 28th Sep 07:30PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

For some reason, I cannot dial into "Canada Only" toll free numbers when using a US server. Is there a way to route outgoing 1-800 numbers to a Canadian server in PAP2?
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Arne Bolen @ 28th Sep 08:22PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by Typhoon :

For some reason, I cannot dial into "Canada Only" toll free numbers when using a US server. Is there a way to route outgoing 1-800 numbers to a Canadian server in PAP2?

One way around this problem could be using Voxalot with two VoIP.ms sub accounts. One sub account is registered to a Canadian server and the other to a US server. By creating rules in the Voxalot account you can decide where different calls are routed.
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Mango @ 28th Sep 08:46PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

Add this to your dial plan:

18[6780][6780][2-9]xxxxxx<:@sip.ca1.voip.ms>S0

This way, 1-800 (866,877,888) numbers will be routed through US1, however if you omit the 1 you may still dial US-only toll-free numbers.
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priller @ 28th Sep 10:24PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by Mango :

Add this to your dial plan:

18[6780][6780][2-9]xxxxxx<:@sip.ca1.voip.ms>S0

This way, 1-800 (866,877,888) numbers will be routed through US1, however if you omit the 1 you may still dial US-only toll-free numbers.

Doing it that way can also yield unexpected results since it also matches numerous area codes other than the Toll Free 8xx numbers, such as:

860 Connecticut
878 Pennsylvania
870 Arkansas
867 Yukon/N.W.Terr´s
...etc

It's better to make specific entries, if all you want to send are toll free.

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Mango @ 28th Sep 10:28PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

Figures the one time I left that out someone would call me on it ;)
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priller @ 28th Sep 10:29PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

Busted!! :D
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Typhoon @ 28th Sep 11:22PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by priller :

I guess the this kind of dial plan is not compatible with PAP2. I tried it with no luck! Any ideas?

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PX Eliezer @ 28th Sep 11:28PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

You have to have IP dialing enabled on your PAP2T.
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Typhoon @ 30th Sep 08:52PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by PX Eliezer :

You have to have IP dialing enabled on your PAP2T.

I'm confused now. I added the

to my dial plan with no luck. 1-800 numbers would go through the US3 server. Even with the IP dialing enabled.

So I deleted my dial plan completely, and only added the
script. Interestingly enough, It let me dial into Canadian 1-800 numbers via CA1 server.

To me, it seems that something in my dial plan is overriding the script. Any ideas?

Here is my dial plan:

([2-9]xx[2-9]xxxxxxS0|1[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|[3468]11|911S0|822|4443|4747|[2-9]xxxxxx|4XXX|xxxxxxxxxxxx.)

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PX Eliezer @ 30th Sep 08:59PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

When you post your dial plan, put a [code=..]
before it.

Continued on next post!
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PX Eliezer @ 30th Sep 09:00PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

Then put a [/code] after it!

That's how you post dial plans!
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Typhoon @ 30th Sep 09:23PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by PX Eliezer :

Then put a [/code] after it!

That's how you post dial plans!

Post fixed!
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priller @ 30th Sep 09:49PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

The dial plan is read left to right until a match is found. When a match is found, it exits and the call is placed.

So, if you have 1[2-9]xx[2-9]xxxxxxS0 before the 1800xxxxxxx entries, it will match on the 1[2-9]xx[2-9]xxxxxxS0 and place the call to US3 and never look at the 1800 entry,

The more specific entries need to be placed first in the dial plan.
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cowcowcow @ 30th Sep 10:03PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

Blind me, I thought the "Display name" field is just for ME to differentiate it from the other line. I never thought it gets sent out to the people I call. I better change mine!

While we are on voip.ms configuration, I wonder if there is any way I can delete some of the default VM folders like "work" Family" and "friends". I believe I can add new ones, just want to keep it simple, I only want "new" and "old".

Sorry for the hijack. Figure it is such a small issue, it doesn't warrant a new thread.
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AVonGauss @ 30th Sep 10:12PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

There is another related thread, »VoIP.ms Outbound Caller ID Name for Canadians. I was wondering how it was getting the name as I didn't see a field in the interface, but figured that was probably because I was in the US and it isn't available at the moment.
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Typhoon @ 30th Sep 11:02PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by priller :

The dial plan is read left to right until a match is found. When a match is found, it exits and the call is placed.

So, if you have 1[2-9]xx[2-9]xxxxxxS0 before the 1800xxxxxxx entries, it will match on the 1[2-9]xx[2-9]xxxxxxS0 and place the call to US3 and never look at the 1800 entry,

The more specific entries need to be placed first in the dial plan.

Thanks to your advice, it's working fine now.
Thanks!
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trev @ 1st Oct 02:44PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by cowcowcow :

While we are on voip.ms configuration, I wonder if there is any way I can delete some of the default VM folders like "work" Family" and "friends". I believe I can add new ones, just want to keep it simple, I only want "new" and "old".

Those sound like standard asterisk voice mail folders. Unless they have customized the voicemail application, there's likely no easy way to disable or hide those.
--
Wondering what I do? Find out at »www.digitalcon.ca
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Typhoon @ 8th Oct 04:19PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

Another issue popped up today. So it seems that for some unknown reason my PAP2 loses registration (at some random point of time) and can't connect to the login server even after rebooting the box. The only way to fix it is to switch to another server and then switch back to the initial server. I talked to voip.ms and they said that other customers have reported a similar issue, but there is no specific cause for it, and that the device should be able to register. Any feedback is much appreciated.
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Mango @ 8th Oct 05:45PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

I am having the same problem, which began as soon as I started using Asterisk. I use Asterisk, a PAP2T, and a SPA921, and they all register to VoIP.ms.

I was told to change on the devices EXT SIP PORT to 5060 and SIP PORT to something unique for each adapter, like 5061 and 5062. This worked, but then the devices could not reliably call each other. There would often be one-way audio, or the caller would hear a reorder tone even though the callee's phone would ring.

The only way I have found to make everything work is to have Asterisk register first, and then the devices. If a device registers before Asterisk has a chance to, it messes everything up.

If anyone knows of a better solution I would be grateful.
--
Who is the best VoIP provider? | Which ATA should I buy? | Dial Plan Tips and Tricks
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trev @ 8th Oct 05:49PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

I have this working happily with the EXT SIP PORT blank and SIP PORT unique for each device. I'm not using VoIP.ms, but I would suspect clearing your EXT SIP PORT will resolve the problem.
--
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Mango @ 8th Oct 06:14PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

By golly, this worked like a charm. Thanks!
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Typhoon @ 8th Oct 09:02PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

But in my case I only have one ATA box (PAP2) with the EXT SIP PORT blank and SIP PORT set to 5060. I think my problem is different from Mango2's since my box initially connects to their server and then after a few days it suddenly can't register again. I have to switch to a different server after which I can switch back to my desired server. Any way to fix this?
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trev @ 8th Oct 11:40PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

Yup, your problem would be different. I'd suspect your router, although I don't have any suggestions on what to look for exactly.
--
Wondering what I do? Find out at »www.digitalcon.ca
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Typhoon @ 10th Oct 09:48PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by Typhoon :

But in my case I only have one ATA box (PAP2) with the EXT SIP PORT blank and SIP PORT set to 5060. I think my problem is different from Mango2's since my box initially connects to their server and then after a few days it suddenly can't register again. I have to switch to a different server after which I can switch back to my desired server. Any way to fix this?

My PAP2 lost registration again. This is becoming annoying. Is there any ways to see detailed logs for a PAP2? I think it's time for some serious debugging.
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Mango @ 10th Oct 10:25PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

Just a thought...could you be experiencing the Linksys DNS bug? If your proxy is set to something like sip.us3.voip.ms, try using its IP address instead.

You can download Kiwi Syslog Server, run it, and then set the Debug Server on the PAP2T (System tab) to the IP of your computer. My debug level is at 2 which seems appropriate.

Like Trev, I also suspect your router. Does rebooting the router solve the problem?
--
Who is the best VoIP provider? | Which ATA should I buy? | Dial Plan Tips and Tricks
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Typhoon @ 10th Oct 10:56PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by Mango :

Just a thought...could you be experiencing the Linksys DNS bug? If your proxy is set to something like sip.us3.voip.ms, try using its IP address instead.

You can download Kiwi Syslog Server, run it, and then set the Debug Server on the PAP2T (System tab) to the IP of your computer. My debug level is at 2 which seems appropriate.

Like Trev, I also suspect your router. Does rebooting the router solve the problem?

- I am already using the IP address (as per your suggestion in an earlier post) instead of sip.us3.voip.ms.

- I will download Kiwi Syslog Server; I gather that I need a computer to be ON all the time to record the logs. Is there an easier way of recording the logs? like internally within the PAP2?

- I'll reboot the router next time my PAP2 fails. What makes the debugging hard for me is that the failure happens every two or three days which makes it hard to reproduce.

- The following is a list of suspects:

1. The US3 server. I just switched to US2 server to see if same thing will happen again.

2. Primary and Secondary DNS servers; I am using 208.67.222.222/20 right now. I will clear them out if suspect one is not an issue. If the problem persists then I will use my own ISP's DNS (Novus) servers.

3. Router: I'll keep an eye on the logs for the router. I'll reboot the router next time after failure. Eventually, I may put the PAP2 in the DMZ. I am currently using a Linksys WRT54GL with Tomato F/W ver. 1.25. I'd appreciate any feedback with regard to correct settings for this specific router.

I'll change one variable at a time. I can't think of anything else right now. Any other thought are greatly appreciated.
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PX Eliezer @ 10th Oct 11:20PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

Seems as though you are using OpenDNS (208.67.222.222/220).

I have never had a problem with them for Voip.MS or any other provider.

------------------------------

Also (sorry if this was already covered):

a) On your PAP2 do you have these set:

Nat Keep Alive: Yes
Nat Mapping/Traversal: Yes

b) Have you shortened your registration interval? Voip.MS says 3600 which may be kinda long. Maybe try something shorter.

c) On the Voip.MS website (dashboard) do you have NAT (Network Address Translation) set to Yes ??
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Typhoon @ 11th Oct 12:06AM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

a) Yes, I have both NAT Keep Alive and Net Mapping enabled.

b) Registration interval is set to 300. The problem is it suddenly loses registration and can not re-register again.

c) Yes, it is checked.
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Stewart @ 11th Oct 01:20AM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

In case this is the retry/nonce problem with some versions of Asterisk, try setting SIP T1 to 1 (so it happens less often) and Reg Retry Long Intvl to 30 (so it recovers quickly when it does happen).
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Typhoon @ 15th Oct 10:56PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

Just an update: It seems that there was something wrong on Voip.ms side as they fixed the problem after I told them about the issue for the second time. I'll switch back to US3 to see if that happens again.
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bbtech6650 @ 16th Oct 03:07AM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

Mango,

Here I was going to have you use IAX for your asterisk install...

Thats what I am currently doing.
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Typhoon @ 26th Oct 08:55PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

My PAP2 still loses registration after about 2 days. This is frustrating. I rebooted the router, put the PAP2 in DMZ zone, even connected it directly to the Internet with no luck. The only way to fix it is to switch to a different server and then move back the desired one. Router logs show no dropped or denied packets. The only thing I can think is there is some setting somewhere in the PAP2 that is preventing registration after an interval. Any feedback is greatly appreciated.

On a different topic, does any one know how voip.ms US servers treat Canadian 911 calls? i.e. if I'm using a US server, will 911 calls route to a Canadian call center or an American?
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trev @ 26th Oct 10:29PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by Typhoon :

On a different topic, does any one know how does voip.ms US servers treat Canadian 911 calls? i.e. if I'm using a US server, will 911 calls route to a Canadian call center or an American?

Presumably they'll route all their 911 calls to the same service regardless of which server you are registered to. As long as you have your address registered it will likely be connected to the right place.
--
Wondering what I do? Find out at »www.digitalcon.ca
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PX Eliezer @ 26th Oct 10:52PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by Typhoon :

The only thing I can think is there is some settings somewhere in the PAP2 that prevents registration after an interval. Any feedback is greatly appreciated.

You need to post your config settings ("Advanced") otherwise how else can it be seen how your settings differ from the norm?

I agree with Trev's response, BUT this is so important, you really should check it with their customer service. I'd create a support ticket rather than do live chat.
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Stewart @ 26th Oct 11:44PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

said by Typhoon :

My PAP2 still loses registration after about 2 days.

What kind of modem? If not cable, configured as router? Router model/version? Post screenshots of System, SIP and Line 1 pages of PAP2 (with personal info masked, of course).
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Typhoon @ 28th Oct 09:26PM:
PAP2 Loses Registration after about Two Days

I am using an FTTP (Fiber to Premises) service, so my router is directly connected to the wall RJ-45 socket.

Router: Linksys WRT54GL - F/W: Tomato Ver. 1.25
ATA: Linksys PAP2

Like I said in my previuos post, there is some setting somewhere in the PAP2 that is preventing registration after an interval. Any feedback is greatly appreciated.
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Stewart @ 29th Oct 01:43AM:
Re: PAP2 Loses Registration after about Two Days

Does your router get a public IP address on WAN side? If not, please explain.

Do you have static DHCP configured in your router for the PAP2? If not, are you sure that the PAP2's private IP address has not changed?

When you lose registration, has your public IP address or router's WAN IP address (if different) changed?

When you lose registration, does rebooting the PAP2 regain it? Rebooting both router and PAP2?

Did you lose registration while you had Reg Retry Long Intvl set to 30?
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Typhoon @ 29th Oct 03:30PM:
Re: PAP2 Loses Registration after about Two Days

- The router gets a public IP address (dynamic); however, it hardly changes.

- No, I have not set static IP for the PAP2. I just double checked the IP address and it's the same as before. No change.

- I had rebooted the PAP2 and router in previous incidents, but this time I decided to power cycle the PAP2 first, which didn't help; however, power cycling the router restored connection. This is strange since it only happens with voip.ms.

- I'll try with the Reg Retry Long Intvl = 30 again, to see if this prevents the issue.

I may switch to dd-wrt firmware to see if it makes any difference.
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PX Eliezer @ 29th Oct 03:43PM:
Re: PAP2 Loses Registration after about Two Days

I have used a PAP2T with Voip.MS.

Pretty much the only difference I can see between my settings and yours is that mine only works with "DNS SRV" set as YES.

Otherwise I agree, it's really starting to look like a router issue.
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N9MD @ 29th Oct 03:51PM:
Re: PAP2 Loses Registration after about Two Days

I just noticed that you have Line 1 on the PAP2T-NA showing Use OB Proxy in Dialog set to Yes yet you do not show a name for the Outbound Proxy (in the left column). I have my PAP2's Voip.ms Use OB Proxy in Dialog set to No.

I'm not sure if this makes a difference with regard to your problem.
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Stewart @ 29th Oct 04:15PM:
Re: PAP2 Loses Registration after about Two Days

This smells like a poisoned NAT association issue. To confirm that, the next time it fails, reboot the PAP2 only, verify that it still won't register, change the Line 1 SIP Port from 5060 to e.g. 5080 (but keep the same server). If it now registers, that's IMO reasonably convincing evidence.

For that issue, using proper port forwards (instead of DMZ) is likely to fix it with your present router. Of course, you should use either static IP on the PAP2 or static DHCP in the router, to ensure that the forwarding goes reliably to the right place.
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Typhoon @ 29th Oct 04:26PM:
Re: PAP2 Loses Registration after about Two Days

What are DNS SRV and Use OB Proxy in Dialog settings for?
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Typhoon @ 29th Oct 04:29PM:
Re: PAP2 Loses Registration after about Two Days

Thanks for the advice Stewart. I'll change the SIP port to 5080 next time it fails. Hopefully, I can find the culprit.
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N9MD @ 29th Oct 05:05PM:
Re: PAP2 Loses Registration after about Two Days

said by Typhoon :

What are DNS SRV and Use OB Proxy in Dialog settings for?

Apologies ... but my skills (as a gastroenterologist) only extend to experimenting with various settings on PAP2Ts ... with a wide variety of VoIP providers. For CallCentric I set Use DNS SRV to Yes and Use OB Proxy in Dialog to Yes. For all the others --- VOIPo, ViaTalk, StanaPhone, FreeDigits, Voip.ms, Gizmo5 (now GEO), SipGate, Voxox, VoiceStick, AxVoice --- these are set to No.

For providers calling for OutBound Proxy (& Dialog) use --- VoiceStick, FreeWorldDialup, CallCentric, FreeDigits --- I entered a name for the Outbound Proxy ... such as outbound.freedigits.net.

Perhaps one of our technical members will jump in with a definition of the three Outbound Proxy settings.

»[wtf?] I do NOT understand Outbound Proxy on Linksys devices
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Typhoon @ 29th Oct 10:34PM:
Re: PAP2 Loses Registration after about Two Days

Stewart, I just put the router in the DMZ zone. Hopefully, this will eliminate the issue.

N9MD, I'll set "Use OB Proxy in Dialog" to No to see if that makes any difference.

Anyone else has any opinions about "Use OB Proxy in Dialog" and "Use DNS SRV" settings?
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trev @ 30th Oct 12:07AM:
Re: PAP2 Loses Registration after about Two Days

said by Typhoon :

Anyone else has any opinions about "Use OB Proxy in Dialog" and "Use DNS SRV" settings?

The OB Proxy is something totally irrelevant for most situations. Unless you're going through a SIP proxy before getting to your ITSP, you don't need this.

DNS SRV is a way to make VoIP more user friendly. For example, SRV records allow someone to call typhoon@hisplace.ca instead of having to call 17005551234@sip.typhoonsprovider.net. In the context you're probably more curious about it basically allows your ITSP provide a single domain where your device can utilize automatic fail over to a secondary or third gateway if the primary becomes unreachable for any reason.
--
Wondering what I do? Find out at »www.digitalcon.ca
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Gobe @ 30th Oct 04:41PM:
Re: PAP2 Loses Registration after about Two Days

I am experiencing problem all day today without changing any configuration......ATA device seems to fail registration after each call then re-acquire only to fail registration again after the next call....don't know what could be the problem......I haven't changed anything .......aggh.....

UPDATE @ 18:19 Problem fixed by using the IP address instead of the hostname, I guess I just discovered the Cisco/Linksys SPA-2102 DNS bug today. Curious, I am using the latest firmware 5.2.10 - is this a new bug ?

.
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FrugalNerd @ 31st Oct 12:24AM:
Re: PAP2 Loses Registration after about Two Days

said by Typhoon :

Stewart, I just put the router in the DMZ zone. Hopefully, this will eliminate the issue.

N9MD, I'll set "Use OB Proxy in Dialog" to No to see if that makes any difference.

Anyone else has any opinions about "Use OB Proxy in Dialog" and "Use DNS SRV" settings?

Typhoon, I've been successfully using a PAP2T with voip.ms. I'm assuming (but could be proven wrong...) that the PAP2T setup would be nearly identical to the PAP2 that you have.

I've documented step-by-step instructions for voip.ms using a PAP2T: »www.frugalnerd.com/doc/voip-pap2t-setup

If you're game you could reset your PAP2 to factory defaults and try following these steps.

FYI I'm using dd-wrt but others have followed these steps using other routers and I haven't heard any reports of lost registrations to voip.ms.

-Scott
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Mango @ 31st Oct 12:57AM:
Re: PAP2 Loses Registration after about Two Days

said by Gobe :

I am using the latest firmware 5.2.10 - is this a new bug ?

It also appears (albeit in a different form) in my PAP2T and my SPA-921 IP phone. You'd think Sipura/Linksys/Cisco would have noticed by now.
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Typhoon @ 31st Oct 04:20PM:
Re: PAP2 Loses Registration after about Two Days

said by FrugalNerd :

Typhoon, I've been successfully using a PAP2T with voip.ms. I'm assuming (but could be proven wrong...) that the PAP2T setup would be nearly identical to the PAP2 that you have.

I've documented step-by-step instructions for voip.ms using a PAP2T: »www.frugalnerd.com/doc/voip-pap2t-setup

If you're game you could reset your PAP2 to factory defaults and try following these steps.

FYI I'm using dd-wrt but others have followed these steps using other routers and I haven't heard any reports of lost registrations to voip.ms.

-Scott

Thanks Scott. You gotta nice step-by-step guide there. There are a few differences between your recommended settings and mine. I will apply your settings next time it fails. Right now I've put my PAP2 in DMZ zone to see if I can isolate the issue.

Mango, if you don't mind me asking, what is your ping time to us3 server. Mine is reduced to ~33ms!
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Stewart @ 31st Oct 05:13PM:
Re: [Other] Questions/Issues Regarding Voip.ms VOIP Service

If you suspect a DNS problem, try changing your Primary DNS to e.g. 4.2.2.1 .

IMO it's very unlikely that DMZ will help, unless it happens to cover up a bug in your router firmware. DMZ means: If an incoming packet would otherwise be discarded, send it to the DMZ host, with the destination port unchanged.

Now, unless your problem is caused by DNS or another low-level networking issue, it's likely that failure to register is one of:

1. Outbound REGISTER request not being properly sent.
2. Server rejects request, ignores it, or sends reply to wrong place.
3. Incoming reply not being properly received.

DMZ clearly won't help (1) or (2). It could help (3) but it's hard to think of a situation, given that (1) and (2) are ok, where that would arise, except for a router bug.
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Mango @ 31st Oct 06:16PM:
Re: PAP2 Loses Registration after about Two Days

said by Typhoon :

Mango, if you don't mind me asking, what is your ping time to us3 server. Mine is reduced to ~33ms!

Well, as a matter of fact...

My ping time to US3 is

120ms! That's right, 120ms! And do you know why? Because I switched from Shaw to Telus and

Telus routes traffic to US3 via

Washington, DC!

I raised it with their support but their TSR didn't even know what latency was. I posted in the Telus forum and was given some suggestions but those haven't panned out so far.

I'm using CA1 at the moment. 80ms isn't so bad I guess :(

m.
--
Who is the best VoIP provider? | Which ATA should I buy? | Dial Plan Tips and Tricks
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Typhoon @ 5th Nov 06:00PM:
Re: PAP2 Loses Registration after about Two Days

said by Stewart :

This smells like a poisoned NAT association issue. To confirm that, the next time it fails, reboot the PAP2 only, verify that it still won't register, change the Line 1 SIP Port from 5060 to e.g. 5080 (but keep the same server). If it now registers, that's IMO reasonably convincing evidence.

For that issue, using proper port forwards (instead of DMZ) is likely to fix it with your present router. Of course, you should use either static IP on the PAP2 or static DHCP in the router, to ensure that the forwarding goes reliably to the right place.

Stewart, my PAP2 failed again today; so this time, without changing the server, I rebooted the PAP2, verified that it didn't register, and changed the SIP port from 5060 to 5080. The change made the PAP2 work again. I switched back to 5060 without any issues. The part that I don't understand is I had already put the PAP2 in the DMZ with static internal IP, so I don't know what exactly is causing the issue. :mad:
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crazyk4952 @ 10th Nov 11:45PM:
Re: PAP2 Loses Registration after about Two Days

Were you ever able to get this issue resolved, or is your PAP2T still failing?
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scooby @ 11th Nov 12:41AM:
Re: PAP2 Loses Registration after about Two Days

I'm experiencing the same issue here. Its quite weird. My PAP2T will stay registered for a few days and then it needs to be rebooted. A couple times that did not even work so I changed from US4 to US3 and so far it always come back after a reboot. I have Voicepulse on the other line and it never has this kind of issue.

I will try the HOWTO by FrugalNerd the next time it fails to register. I'm using U-Verse and have tried putting the PAP2T in the DMZ as well.
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Stewart @ 11th Nov 01:04AM:
Re: PAP2 Loses Registration after about Two Days

If a reboot fixes it, there was most likely a transient network error. Set Reg Retry Long Intvl to 30; with luck it will regain registration quickly.
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crazyk4952 @ 11th Nov 08:33AM:
Re: PAP2 Loses Registration after about Two Days

I believe I have experienced the same issue as well. I have VOIP.ms configured for line 2 on my PAP2T. Last night, voip.ms was not showing it as registered. However, I still got a dial tone, was able to place outgoing calls, and the green "line 2" light was on. I was not able to get incoming calls.

I set the PAP to register to the main account (instead of a sub account) and this fixed the problem, although probably temporarily.

I just set the Reg Retry Long Intvl to 30. I will see if this helps.
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FrugalNerd @ 14th Nov 07:03PM:
Re: PAP2 Loses Registration after about Two Days

said by scooby :

... A couple times that did not even work so I changed from US4 to US3 and so far it always come back after a reboot. ...

scooby, has your PAP2T been keeping its registration?

I live pretty close to you in the Chicago northwest suburbs and my current recommendation is to use US2 (sip.us2.voip.ms).

US3 is pretty far from us in Los Angeles. And I had some problems a few weeks ago with US1 and US4 - voice quality issues, not registration issues but these may have gone away since I experienced them.
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crazyk4952 @ 14th Nov 07:09PM:
Re: PAP2 Loses Registration after about Two Days

I have Set the Reg Retry Long Intvl to 30 and this seems to have resolved the problem. Over the last several days, I have been logging into my voip.ms account and have verified my device was registered every time. I also have not missed any incoming calls.

Before I changed this setting, I guess my device would fail an attempt to register and then wait 20 minutes before trying again?

As long as it is fixed, that's all I care about!
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Typhoon @ 20th Nov 09:01PM:
Re: PAP2 Loses Registration after about Two Days

Resetting my PAP2 to factory settings, seems to have helped a bit, but not fixed the issue completely. One thing I suspect is that after the reset, the reg. exp. interval was reverted back to 3600. This time it took 10 days until my pap2 failed as opposed to every 2 days. I have changed my LAN's internal IP address to a standard Private IP address to see if that can solve the problem.
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PX Eliezer @ 20th Nov 09:07PM:
Re: PAP2 Loses Registration after about Two Days

said by FrugalNerd :

And I had some problems a few weeks ago with US1 and US4 - voice quality issues, not registration issues but these may have gone away since I experienced them.

I've had voice quality issues with US4 (NY) even though it's close to me!

For me US1 (Houston) worked better for voice quality, even being so far away!
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Typhoon @ 22nd Nov 01:29AM:
Question about Calling Ques/Ring Groups

...maybe someone here can help me understand "ring groups" better:

So what I want to do is I would like my voip.ms DID ring five times and then:

If no answer --> call gets forwarded to my cell after the 5th ring.

If no answer again after 5 rings--> call gets forwarded to my voip.ms voice mail.

Can I implement the above using ring groups?
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Mango @ 22nd Nov 01:35AM:
Re: Question about Calling Ques/Ring Groups

What you're describing is sequential ringing. VoIP.ms has simultaneous ringing. What would happen is your cell and your VoIP.ms DID would ring at the same time, and you can decide which device you would like to answer your call with.

The advantage is that the caller does not need to wait a potential ten rings to get your voicemail.

m.
--
Who is the best VoIP provider? | Which ATA should I buy? | Dial Plan Tips and Tricks
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