[Gimli]

Yep, that's exactly it.

 

[glasgow]

Now I have a voicemail associated with my account, so this is the final destination of the call as the help tells. If I set "busy" to "none" and my line is busy when somebody calls in, he will get into my voicemail and charge will occur. If I set "busy" to "system/busy", he will get a busy tone and there will no charge involved. Is that correct? I set "no answer" to "none" and set the dial time out to 60s because I want my phone's answering machine to handle this (6 rings, which is less than 60s).

And also, if I send caller ID filtering to system/message or hangup, when this caller calls in, I will not be charged, correct? I would like to block some telemarketing calls at voip.ms side instead of on my phone, which has limited block number storage.

EDIT: I just got the answer:

The routing of the system to a busy will not incur any charges. If it is a recording to be played like "not in service"/"disconnected", the duration of the playing of the recording is charged. The same applies to the CallerID FIltering too.

 

[endiz]

Voip.ms quality issues

Hey Folks,

I'm having serious quality issues with my voip.ms line. I have a TekSavvy 15/1 connection, connected to a DDWRT router with a PAP2T acting as my SIP gateway. I configured my DID line according to http://www.toao.net/25-linksys-ata-configuration , and also set my RTP Packet Size: to 0.010.

From my end the voice quality is fine, but the caller on the other end can barely make-out anything I'm saying. Funny thing is, the issue is intermittent. Sometimes, quality is great, other times I sound like a symbiotic alien.

As a side note, I do have a SIP client (CSipSimple) configured on my android, which works 100% without any quality issues. This leads me to believe that the PAP2T is configured incorrectly. I have reset the configuration several times, but nothing seems to solve the issue.

Any help would be appreciated. Thanks!

 

[glasgow]

My RTP Packet size is 0.02, if that matters. I have no problem with calling quality so far.

 

[endiz]

I tried changing it to 0.02 and I'm still having issues

Some additional information, I also have QOS enabled and I am not utilizing p2p while trying to make the phone call.

 

[restamp]

Hello. I'm new to VOIP, to voip.ms, and to this forum. I recently acquired a PAP2T and funded an account with voip.ms to experiment with internet telephony. So far, everything seems to have fallen into place fairly nicely. I'm able to make and receive VOIP calls and the quality seems OK. Voicemail seems to work fairly seamlessly, and for a fraction of the cost of my landline I now have a wealth of telephone options at my fingertips that I used to only dream of. Then I discovered this thread, and over the past several days I've read through its contents. It seems the principals here possess a wellspring of knowledge about voip.ms.

I do have a few questions I'd like to throw out on several VOIP topics:

1. Does anyone know if it is possible to point a Google Voice number to a voip.ms SIP URI? It seems, with the exception of Gizmo, all they want to deal with are POTS numbers. It would surely be nice if we could integrate Google Voice with a voip.ms subaccount, though.

2. Does voip.ms allow 900 numbers to draw from our voip.ms account? If so (or for that matter, even if not), what sort of sequence would I put in my dialplan to immediately go to fast busy (or something similar) if someone enters a number that starts with '900' or '1900'? Something like "1900!S0"? "1900S0!"? Something else? Or would it be better to capture the whole number before rejecting it?

3. I see people occasionally suggesting that people set the "VIA" entries on the SIP page of the PAP2T to "yes" (Handle VIA received, etc) when they are having difficulties with registration. What do these VIA entries do?

4. Finally, sheesh, this thread is over 1600 posts long. Has anyone ever considered expanding this discussion into its own subforum?

Thanks in advance. Frankly, this stuff is amazing.

 

[Gimli]

Alright, the only one I can really be of any meaningful help on is #2:
2. Does voip.ms allow 900 numbers to draw from our voip.ms account? If so (or for that matter, even if not), what sort of sequence would I put in my dialplan to immediately go to fast busy (or something similar) if someone enters a number that starts with '900' or '1900'? Something like "1900!S0"? "1900S0!"? Something else? Or would it be better to capture the whole number before rejecting it?

Add the following string at the end of your calling plan and 900/976 numbers won't be connected: 1900!|900!|1976!|976!

 

[Mango]

Welcome to the forum, restamp

3. I see people occasionally suggesting that people set the "VIA" entries on the SIP page of the PAP2T to "yes" (Handle VIA received, etc) when they are having difficulties with registration. What do these VIA entries do?

I can answer this one. VIA is a field in a SIP header. It looks like this:

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=q9hX4bT-3eb1876b

Sometimes, as in the example above, the IP address that appears in that field is a local (non-routable) IP address. One way of handling VoIP devices behind NAT is if the VoIP device replaces its IP address in the VIA header with the public IP address.

In 99% of cases, this will not be necessary, since VoIP.ms correctly compensates for this by replying to the IP address that the traffic arrived from. If I was the one that mentioned it before (and I think I was) it was probably just for troubleshooting purposes and not a blanket recommendation for everyone.

4. Finally, sheesh, this thread is over 1600 posts long. Has anyone ever considered expanding this discussion into its own subforum?

In fact I was just thinking the same thing to myself yesterday. What do the rest of you think? Do we need a separate VoIP.ms forum? Or should we lock this thread and just create lots of threads about VoIP.ms in the VoIP providers forum? I agree that it's difficult to find references to something that happened more than a couple of days ago.

I'd vote for the latter (creating smaller threads for each topic about VoIP.ms in the VoIP providers forum) but my opinion could be swayed. Hugh, what is your opinion?

Frankly, this stuff is amazing.

I agree! Isn't it grand?

m.

 

[glasgow]

1. Does anyone know if it is possible to point a Google Voice number to a voip.ms SIP URI? It seems, with the exception of Gizmo, all they want to deal with are POTS numbers. It would surely be nice if we could integrate Google Voice with a voip.ms subaccount, though.

I will try this one though I am a rookie too.

I tried and succeeded to forward my GV inbound calls to a voip.ms subaccount or main account. The solution is you register a free IPKall account, you will get a free US number. All calls to this number will be forwarded to SIP URI you set up in IPKall account. Now just add this number into your GV account and forward GV calls to it. That's it. You can get the SIP URI of your main account from the manage DID/view, and SIP URI of your subaccount from edit subaccount.

 

[glasgow]

In fact I was just thinking the same thing to myself yesterday. What do the rest of you think? Do we need a separate VoIP.ms forum? Or should we lock this thread and just create lots of threads about VoIP.ms in the VoIP providers forum? I agree that it's difficult to find references to something that happened more than a couple of days ago.

I'd vote for the latter (creating smaller threads for each topic about VoIP.ms in the VoIP providers forum) but my opinion could be swayed. Hugh, what is your opinion?
m.

I agree with you Mango. A sub-forum with smaller threads will be much easier for people to discuss specific topics and to help people find useful information more quickly. Voip.ms deserves a sub-forum, IMHO.

 

[BGY11]

I tried using 3CXPhone on my Android over my home WiFi connection and the results weren't good. Significant delays and the person I called could barely make out anything I said. I'm sure I don't have things configured right but since I'm not using an ATA adapter I'm not too sure what I'm doing.

If I need a separate adapter then how would I use voip.ms over a 3G cell connection?

(edit: Yes, I do realize some of these may be newbie questions - I am going to look into this as I'm definitely interested in using voip.ms).

 

[restamp]

First, thanks for the quick feedback! Some additional comments:

Gimli, thanks for responding. I think the sequence "1900!", without the S0, would not be successful at blocking 1-900 numbers. Here's why: "1900!" only blocks the string "1900". It would not block, say "19005551212". So, one could either use the blocking sequence "1900xxxxxxx!" (to wait for the entire number to be entered, and then block it), or go to fast busy immediately (without allowing any more digits to be entered) after the initial "1900" is received. I played around with my PAP2T this evening, and found that the sequence "1900!S0" does indeed go to fast busy immediately after "1900" is entered. (I didn't check if "1900S0!" also worked.) Thus, I have coded the sequences "1900!S0|900!S0" in my dialplan, and they do appear to block all attempts to access 900 numbers. But, this brings up a new question:

Does anyone know how long a dialplan can be?

Also, I'm taking voip.ms a their word that they will disallow all calls outside of USA48 and Canada if I turn off international dialing. (Would be nice to have had AK and HI included, though.) This would presumably block the infamous '809' area code. BTW, there seem to be a lot of potentially troublesome area codes in NA. For instance, see here.

Mango, thanks for the info on the "VIA" options. I guess I'll leave them turned off. The only real problem I have encountered with VoIP so far is with my old ZyXEL router, which seems to have a bug in the NAT table code which manifests itself when the NAT table wraps. Apparently, they never envisioned a UDP session being held up that long. But, it takes more than a 24 hour day for the NAT table to wrap, so I have worked around the problem by programming a reset of the NAT table in the wee hours every morning. Oh, and BTW, thanks for your excellent PAP configuration guide. It was most helpful when I was setting mine up.

glasgow, thanks for the pointer to IPKall. I'd never heard of them before and it sounds promising. But, it sure presents one complicated path when you include Google Voice. Is the audio delay reasonable on these calls? Actually, it almost makes sense to drop GV and just go with IPKall, but I guess that would force my friends to always call me long distance. (The old landline concepts of LATAs are so dated today it's a wonder they are still viable.)

Anyway, thanks for all the help, folks. I look forward to following the discussions.

 

[99semaj]

I agree with you Mango. A sub-forum with smaller threads will be much easier for people to discuss specific topics and to help people find useful information more quickly. Voip.ms deserves a sub-forum, IMHO.

+1

Maybe it could be called BYOD VOIP to reflect the much more technical nature of VOIP device configuration, than say, your typical consumer wondering about Vonage.

 

[Mango]

Do we really need a separate forum? I ask because I counted the threads that had posts in February that I thought weren't about BYOD...and there were about three or four

It looks like we all agree that smaller threads would be easier to follow.

I just noticed the VoIP providers forum is now a regular forum rather than a subforum. We're growing up

m.

 

[Gimli]

Gimli, thanks for responding. I think the sequence "1900!", without the S0, would not be successful at blocking 1-900 numbers. Here's why: "1900!" only blocks the string "1900". It would not block, say "19005551212". So, one could either use the blocking sequence "1900xxxxxxx!" (to wait for the entire number to be entered, and then block it), or go to fast busy immediately (without allowing any more digits to be entered) after the initial "1900" is received. I played around with my PAP2T this evening, and found that the sequence "1900!S0" does indeed go to fast busy immediately after "1900" is entered. (I didn't check if "1900S0!" also worked.) Thus, I have coded the sequences "1900!S0|900!S0" in my dialplan, and they do appear to block all attempts to access 900 numbers. But, this brings up a new question:

Nope, ! acts like S0 on it's own. As soon as the string is entered it goes to fast-busy signal.

Does anyone know how long a dialplan can be?

It can be pretty long but when it gets too long you may start running into issues where it takes so long for the ATA to process that some of it's functions don't work. I've never seen it though.

Here's mine:
(1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|<311:18662992002>|<511:18883550511>|911S0|1900!|900!|1976!|976!)

It's simple and works great!

 

[restamp]

Nope, ! acts like S0 on it's own. As soon as the string is entered it goes to fast-busy signal.

Wow. So it does. Thanks for the correction, Gimli. I've changed my dialplan to take that into account.

Does anyone know if 900 number dialing is even allowed by voip.ms? If you disable international dialing, it seems they take great pains to ensure that you'll never get hit by abnormal charges. I wonder if they may lump 900 numbers in with international calls.

[dialplans] can be pretty long but when it gets too long you may start running into issues where it takes so long for the ATA to process that some of it's functions don't work. I've never seen it though.

Here's mine:
(1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|<311:18662992002>|<511:18883550511>|911S0|1900!|900!|1976!|976!)

It's simple and works great!

Do you subscribe to e911? I don't (on my voip line), but I did look up the Sheriff's regular phone number and encode it in my dialplan as you have for 311 and 511. I debated whether or not to add an 'S0' to the end, though: On the one hand, it makes the call get placed a little faster, but OTOH, I can envision trying to dial, say, area code 914 and accidentally double-tapping the '1'. The 'S0' would immediately force the call to 911 whereas leaving it off would give me a couple seconds to abort the call before it was placed.

Also, I'm curious: Isn't "*xx" a subset of the sequence "*xx.", so why use both?

Anyway, thanks for setting me straight on the '!' vs. "S0" thing.

 

[restamp]

I'm sending this as a separate post because I'm going to need to use a couple URLs in it, which will probably get it held for moderator review.

glasgow, you mentioned making use of IPKall to interface your voip.ms account with Google Voice. Before I go on, a bit of background: In addition to my interest in VoIP, I'm also involved with running Linux on small arm platforms such as the Seagate Dockstar (now discontinued) and the multi-vendor SheevaPlug. The beauty of these devices is that they are capable of running a modern Linux with a power footprint of only a couple watts. Thus, they make great 24/7 servers for a variety of purposes.

In the Dockstar forum, someone just recently described how to configure Asterisk to interact with Google Voice as if it were Google Chat, the Google web-based interface for making free phone calls. Last night, I configured a Dockstar to run Asterisk and I now have the second line on my PAP2T swung over to interfacing with Google Voice directly, via my new Asterisk server. (The first line remains linked to my voip.ms account.) So far, it's worked quite well. If someone calls my Google Voice number, my PAP2T line2 rings the attached phone, and if I place an outgoing call, it connects (via Asterisk) through Google directly. Very simple interface and the sound quality is very good.


Enjoy!

 

[Mango]

I just tried two 1-900 numbers and neither worked.

--

You guys have to explain to my wife that it was for a scientific purpose, if she finds out I have been Googling for and calling 1-900 numbers.

m.

 

[Gimli]

Do you subscribe to e911?

I do. I figure having the ability to have my address sent automatically is helpful for emergencies where I may not be able to speak.

Also, I'm curious: Isn't "*xx" a subset of the sequence "*xx.", so why use both?

The way I understand it the "." means an unlimited repeat of the last character but there has to be at least one. So *xx. is actually * followed by at least three numbers. Either way it was in a generic dial plan I copied as a source for mine. I could probably just remove it as I don't use and * numbers with more than 2 digits.

 

[Mango]

Close. The . means a repeat of zero or more of the previous element. So *xx. means * followed by one digit followed by zero or more digits. *0 would match, *01 would match, *012345 would match, etc.