[yf]

Folks:
I really need your help. I setup my SPA 2102 with Voxalot (US server). I have a remote SPA3102 which I registered my PSTN line with another Voxalot account (also U.S. server). When I dial my SPA 3102 I noticed something strange when I punch in my PIN to gain access to my PSTN dial tone. I noticed that the numbers I dial get repeated unintentionally. For example if I am dialing 37420 I found the number dialed 37777444222000 (something like that). I used to register with callcentric and FWD and both were working fine. I guess this has to do with the DTMF settings. I don’t know what the settings for Voxlot regarding the DTMF are. Can anyone help me please?

[hwittenb]

Generally your dtmf setting should be AVT. This doesn't mean that you can transmit dtmf without problems. My experience is that it will vary with different providers. AVT is RFC2833 where the dtmf characters are sent as specific packets in the rtp stream for the different digits. This is opposed to "Inband" where the digits are part of the digitized audio voice stream. Inband only works well with the G711 codec where the voice stream is not compressed. The problem encountered with AVT is several packets are sent for each character depending on how long you hold down the key with an "end" packet when you release the key. The packets are reassembled by your provider. If they get out of sequence there can be problems.

If you are trying to send dtmf digits to a 3102 that is giving you dial tone, I believe you are a lot better off if you are able to use "http digest" authentication. When you can do this, you send the digits you wish to dial as part of a sip invite, and I believe you open yourself up to a lot less trouble.

You can only use "http digest" authentication where you have another system, or you have an ata, giving you dial tone and that system (ata) has your 3102 ip address setup as the proxy and you have a userid and password in that system, or ata, that is sent to your 3102 as part of the call setup.

In other words, if you had a 2102 line tab setup with a distant 3102 ip address and port number as the proxy and in the 2102 line tab you had the userid and password that you had setup under VoIP Users and Passwords (HTTP Authentication) on the 3102's pstn tab, then you can directly dial the number you wish to be sent out on the distant 3102. You do not receive a dialtone from the distant 3102. You need to be using direct ip dialing to do this. Of course you also need to have answer call without reg YES.

I believe that you can do this with Voxalot, although I have not tested it. It does work with PBXes and as I mentioned it also will work with a regular Sipura adapter or a programmable softphone.

[yf]

Thank you hwittenb.

If you are trying to send dtmf digits to a 3102 that is giving you dial tone, I believe you are a lot better off if you are able to use "http digest" authentication. When you can do this, you send the digits you wish to dial as part of a sip invite, and I believe you open yourself up to a lot less trouble.

I wish I can use the "http digest" but both 2101 and 3102 are using dynamic IP address. So, I believe this option is not available for me right now.

My experience is that it will vary with different providers. AVT is RFC2833 where the dtmf characters are sent as specific packets in the rtp stream for the different digits. This is opposed to "Inband" where the digits are part of the digitized audio voice stream. Inband only works well with the G711 codec where the voice stream is not compressed. The problem encountered with AVT is several packets are sent for each character depending on how long you hold down the key with an "end" packet when you release the key. The packets are reassembled by your provider. If they get out of sequence there can be problems.

The information above made me think of changing a single setting in my 2102 (the dialing ATA). I reduced RTP Packet Size from the default (0.020) to 0.010. In one of Voxalote forum postings they recommend (0.050) which drove things crazy. I don't know of the "RTP Packet Size" relates to the problem or not. I notice the problem is currently solved.

Any comment from your side?

[hwittenb]

Originally Posted by yf

I wish I can use the "http digest" but both 2101 and 3102 are using dynamic IP address. So, I believe this option is not available for me right now.

There are two free services you see referenced a lot, dyndns.com and no-ip.com that can provide DNS translation for a symbolic address if you keep the service up to date. Some routers support DDNS which will notify the service if your ip address changes. In addition to the two above, my router help page (dd-wrt) also lists freedns.afraid.org, and zoneedit.com as providing the service. If your router doesn't support DDNS you can use a program running on a pc to update the service, but obviously then you need to keep your pc running.

The information above made me think of changing a single setting in my 2102 (the dialing ATA). I reduced RTP Packet Size from the default (0.020) to 0.010. In one of Voxalote forum postings they recommend (0.050) which drove things crazy. I don't know of the "RTP Packet Size" relates to the problem or not. I notice the problem is currently solved.

If you reduce the packet timing from 20 ms to 10 ms you send twice as many packets. It could impact things, who knows. I guess it depends of the exact nature of why the digit packets are not interpreted correctly. I think the default on the adapter is 30 ms. When you increase the number of packets sent you obviously increase your bandwidth requirement.

[yf]

Thank you again for the valuable tip about DNS translation. I have to read it carefully and see how it works. Fixing the IP address at one end will help me a lot. My router is 3Com. It has been ages since I went through its documentation. I will look into this issue. Having my PC on all the time is not an option. The power goes off from time to time. It is not that frequent but it may go off for 5 to 10 minutes every 3 to 4 days. Sometimes it says for a whole week. This is how I lose my IP address. I thought of buying a small UPS which I will do shortly.

The SPA that I reduced its RTP Packet Size to 0.010 is using a North America speedy DSL connection (I guess 5MB connection and 300+uploading). So, the big bandwidth is there. The other end is on a very limited bandwidth (512/128 kbps). I am also not sure of the Internet quality there as well.

The problem may be in the receiving SPA (which is connected to the low-bandwidth DSL). I mean maybe the way it re-arranges the receiving packets get affected by other reasons at that end. I noticed that I have some difficulties entering the PIN and dial a number if I access the PSTN number through one of the SIP broker PSTN access number. The same problem occurs (many numbers get repeated). I am not a network guru. I am just a sort of a “super user” who enjoys networking as a side hobby. All my experience is from limited readings and observation (through trial and error). I changed back the RTP Packet Size to the default 0.030 and the problem appeared once more. I am moving it back to 0.020 (recommended by Voxalot). So far it is working fine. If, for whatever reason cause any problem I will use a lesser setting (like 0.015 or 0.010).

Do you think the setting in “DTMF Playback Length” and “DTMF Playback Level” has to do anything with this issue?

[hwittenb]

Originally Posted by yf

[Do you think the setting in “DTMF Playback Length” and “DTMF Playback Level” has to do anything with this issue?[/size][/font]

I don't think it has anything to do with the problem. The parameters are under Miscellaneous on the Regional Tab. It is not clear to me whether they apply to the FSX interface or the FXO interface or both. The strong implication is that they apply to the FSX interface because other nearby parameters are labeled FXS. In other words it is the feedback to you of the tone on the attached handset. Perhaps it is the level and duration on the outgoing FXO (PSTN) port. Maybe someone really knows.

You must be running a trace to know that your problem is multiple dtmf digits are decoded for a single key depression. In this case the problem happens somewhere between the transmission of the digits to the network and the receiving of the digits at the distant end.

I doubt that the problem is in the distant SPA's reception of the rtp stream. If you think this is the case you could get the temporary ip address of the SPA, forward the port on its router, and call it directly using direct ip dialing and see how that works. You shouldn't have to change much on the distant SPA, just forward the port and set Answer without Reg YES.

It is interesting that you have anecdotal evidence that increasing the number of rtp packets effects the proper reception of the dtmf digits.

[yf]

hwittenb

Some routers support DDNS

I wonder how I find if my router supports DDNS. I am accessing the router remotely. I don't want to do anything wrong with it, otherwise I will lose access to it. Mine is 3Com OfficeConnect ADSL Wireless 11g Firewall Router (Model 3CRWDR100A-72).

As for the RTP I guess you are right as it has no relation to the DTMF.

[ymhee_bcex]

yf,

I am administering a machine remotely as well; and when it boots it runs inadyn that synchronizes with dyndns. I think it will help you, too.

[yf]

ymhee_bcex:
Thank you for the info. I am still away from my machine and will keep so for a while. If I have physcial access to my machines I would install a clinet and the problem will be solved. I was wondering if my router supports DDNS or not. If so then I can use it instead of downloading a client.