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bushguy

@96.63.23.x

service provider as well as me are 100% stumped

Hello Forum,
I'm at a loss and hope someone can help. We live remote. xplornet is our link to the outside world. We have a surfbeam modem. With our current setup, I have found only one commercial voip service that will work. That is net2phone. net2phone is an old legacy softphone product that works well and is used to call landlines. skype and zoiper work fine as a p2p (computer to computer) chat.

I have been in remote camps that have our same satellite system and xplornet provider and a standard phone/ATA that also works well. The point being, I know for a fact that voip will work with our xplornet satellite connection.

I have been trying to find an alternative to net2phone. The latest attempt has been with another softphone product. It is Zoiper communicator 3.0 linked to voipvoip as the voip service provider. I can use zoiper to make p2p calls fine. It fails when I try to call landlines. Both Zoiper support and voipvoip engineers have tried to help and are baffled and have thrown in the towel and given up.

Here's what happens. I try to make a call to a landline. It rings and is answered. I can hear the other party talking clearly. All they hear from me is a blip, blip, blip. One way communication.

This is a SIP account. We have tried the following... ports 5060 (default) , 5065 , 80 and 443. Ports 80 and 443 were completely blocked and would not allow the program to even register itself.

I have the following codecs available: GSM , Speex, iLBC-30 , iLBC-20 , a-law , u-law . I tried each one individually and eliminated the rest. a-law and u-law were one way conversation, The rest of the codecs would not connect and gave me various error messages.

My computer is directly wired to the surfbeam modem. There is no router. I know I am behind an xplornet NAT and have a dynamic IP address. I tried stun.voipvoip.com:5060 as a STUN configuration and that failed. Same problem. One way conversation.

Other things they had me do: change the SIP server name sip3.voipvoip.com to 69.90.209.57 I also can confirm that
BLF, Subscribe Presence, Publish presence, Use Rport, Force RFC-3264, Send KPML are all disabled.

Since we live so remote, phone is not a nicety, it is critical for us and I hope someone out there has some ideas to try. I know there is some setting that will allow this to function. Zoiper and voipvoip have given up on us. Thank you so much in advance for your help and time.

Mango
What router are you using?
Premium
join:2008-12-25
www.toao.net
kudos:13
Reviews:
·AcroVoice
·Callcentric
·Anveo
·Shaw
You may want to try to use a service provider called Callcentric. They have an excellent reputation in this forum and perform well behind many different types of NAT. You can sign up for a free account and test your configuration without depositing any funds by dialing 1-777-123-4567 from your softphone. If it works, their rates to call landlines are quite reasonable. If it doesn't work, their technical support is excellent and they will be delighted to help you.

Here are their instructions for configuring Zoiper: »www.callcentric.com/support/device/zoiper

Good luck,
Mango

gweidenh

join:2002-05-18
Houston, TX
kudos:3
reply to bushguy
A couple things that come to mind:

You mention that Skype works fine. Have you tested Skype while making outgoing PSTN calls? I suspect that will work as well.

Skype has a few advantages that may play into your favor:
Skype is excellent at getting around NAT and firewalls.
Skype using very low bitrate codecs that may work very well on a satellite connection versus the more typical G.711 used by many SIP providers.

So at a high level, to save you from pulling out your hair, may I suggest using Skype for your phone service? 911 would be a potential problem since they do not provide that service.

If you want to explore your SIP options more, I would highly recommend taking Wireshark captures of your traffic while making a call via a softphone. Wireshark is free and suprisingly simple to use.

Most competent VoIP providers (I recommend Flowroute) will be able to analyze both your trace and their trace to determine why the call did not connect.

I would recommend you take a trace using net2phone and voipvoip and comparing the differences on your side first. What codec is net2phone using? Then study the headers of the incoming and outgoing packets. Is the header information different between the two services? Does one have the internal IP address in the header vs the external IP address? There likely is a clue in these captures that will help identify the 1-way audio issue.

These captures in conjuction with captures obtained by your VoIP provider should isolate the issue.

I do fear however, that no setting on your end may fix this issue. If xplornet systems are not particularly SIP friendly, that would be out of your control. Hence my original recommendation to utilize a non-SIP based service.

I would be happy to help analyze the Wireshark traces for you as well.


bushguy

@96.63.23.x
reply to Mango
Thank you Mango. That was one of the first voip providers I tried. I believe I downloaded the recommended softphone from callcentric's site. I was unable to make calls and ultimately callcentric support told me they couldn't help us. In desperation I can maybe give it another try. Unfortunately I struck out once with them but I never used them with zoiper so I'll look at that info you provided. Thanks again Mango. I hope your day goes well.


cb14

join:2013-02-04
Miami Beach, FL
Reviews:
·T-Mobile US
·localphone.com
·callwithus
·Google Voice
·Callcentric
·AT&T U-Verse
·magicjack.com
reply to bushguy
Normally, you should be able on any internet gateway to disable NAT- put the device into DMZ.off course you will not want to do it with a softphone, but I do not see any reason not to do it with an Obi 202. BTW Google Voice via Obi seems to work even where SIP providers fail due to NAT problems.


bushguy

@96.63.23.x
reply to gweidenh
gweideh.. Thank you for your response. I should clarify one thing in regards to skype. skype only works computer to computer. I signed up for the skype paid telephone service and it bombed and would not work. I'm a novice with the technical lingo but I believe trying to make a call to a landline phone via skype is the same as trying to make an outgoing PSTN call?

We're 100 miles from nowhere via floatplane so 911 not an issue at all with any service. I will look into this wireshark which I've never heard of and sounds mighty interesting. voipvoip engineers did have me making calls to trace the route but I never heard what they found and they had me try the STUN routine after tracing and when it failed they bailed out on us.

I wish I could duplicate the net2phone settings. Unfortunately, it is so basic and old that there isn't much in the way of settings. I am in preferences... I see "full duplex" , "voice activity detection", and suppress silence" are check marked.
2 VFPP (voice frames per packet) is selected. On the network page all that is selected for options is Automatic Gateway selection and Don't connect through a SOCKS proxy server. Not much information there. Now there are a couple blocks that are shaded grey with the following info... Primary gateway call1.net2phone.com and alternate call2.net2phone.com and port 6801 is greyed in Uncertain what that means. net2phone is setup on our laptop and I have a new desktop that I am trying to get a voip alternative on.

I will look into the suggested wireshark and can't thank you enough for your time and offer to help. I'm confident there must be some way to place calls since remote camps have the same satellite setup but are not using a softphone. They use real phone and adapter which must do something different that these softphones can't do. I'm assuming the phone/ adapter in camps is SIP based but uncertain. And I'm under the impression that this net2phone is SIP based so I know it works with xplornet. I'll post my findings and updates. Have a good day.


bushguy

@96.63.23.x
reply to cb14
cb14... Thank you for the response. Forgive me but some of the shorthand I am not familiar with. I can tell you that I talked with xplornet about their NAT. They won't disable it so I don't know any way to disable that. I don't know what DMZ.off is or how to do that. But all I have is softphone options (zoiper) currently on this computer and we will not head to civilization for another 6 months so I can't get any hardware solution here until then.

I am not familiar at all with hardware solutions but will keep the suggested Obi 202 in mind. Right now, my only option is to get a softphone to work. I failed with the callcentric softphone as well as ekiga softphone. I at least have zoiper p2p on this desktop and skype on the laptop working which allows us to stay in touch with family that also has zoiper or skype installed. I just need to make it work for land line phones. Net2phone works for now but it's only a matter of time before they shut it down. They only support up to windows XP and have no plans to update in the future. Thanks again. Have a good day.

gweidenh

join:2002-05-18
Houston, TX
kudos:3
reply to bushguy
said by bushguy :

...skype only works computer to computer. I signed up for the skype paid telephone service and it bombed and would not work.

That is very interesting. It may be worth doing a wireshark capture of a computer to computer call as well as a landline call to compare differences.


StillLearn
Premium
join:2002-03-21
Streamwood, IL
Reviews:
·AT&T Midwest
reply to bushguy
I suggest that you try a SIP provider and a soft phone that both do G.729. Set things up to use G.729.

Free version of 3CX does not support G.729 aka G729.

»www.3cx.com/phone-system/edition-comparison/
They offer a free trial of the business versions, so you could prove out the concept at least.


bushguy

@96.63.23.x
StillLearn... Thank you for the suggestion. I have heard that G729 is good with satellite. Zoiper has a paid upgrade that has that feature. I have been working with zoiper support and I offered to upgrade if they thought it would help. They never mentioned G729 as an option although they are well aware of our satellite setup. As well, voipvoip never suggested that either. I will ask them if they support g729.

On a side note, I have linux mint on this desktop. I found and downloaded wireshark easy enough. It is installed and I'm quite bewildered on the next step on how to capture data. I'll keep working on it but at least it is installed and I'm trying to figure out how to use it. Hopefully I can get some data to give you folks. I did hear back from zoiper and they have reconsidered helping us. Thanks. Have a good day.

SCADAGeo

join:2012-11-08
N California
kudos:2
reply to bushguy
said by bushguy :

This is a SIP account. We have tried the following... ports 5060 (default) , 5065 , 80 and 443. Ports 80 and 443 were completely blocked and would not allow the program to even register itself.

said by bushguy :

But all I have is softphone options (zoiper) currently on this computer and we will not head to civilization for another 6 months so I can't get any hardware solution here until then.
---
I failed with the callcentric softphone as well as ekiga softphone. I at least have zoiper p2p on this desktop and skype on the laptop working which allows us to stay in touch with family that also has zoiper or skype installed. I just need to make it work for land line phones. Net2phone works for now but it's only a matter of time before they shut it down. They only support up to windows XP and have no plans to update in the future. Thanks again. Have a good day.

CallCentric also supports port 10123 for registration.

What happens when ZoIPer is configured to use port 10123 with CallCentric, along with GSM codec (remove u-law for testing)?

gweidenh

join:2002-05-18
Houston, TX
kudos:3
reply to bushguy



The third icon brings up a dialog box with all your network interfaces. Pick the one that is currently showing traffic and press start.

Then the icon directly to the right with the red x stops the capture. Save the capture to a file.


jimk
Premium
join:2006-04-15
Raleigh, NC
Reviews:
·Time Warner Cable
·voip.ms
reply to bushguy
Have you tried a provider that supports the IAX2 protocol? It isn't very common, but it also has fewer issues with NAT than SIP does.

VoIP.ms supports it (along with the G.729a and GSM codecs for bandwidth efficiency), and the Zoiper softphone app also supports it.


bushguy

@96.63.23.x
reply to gweidenh
downloadwireshark 6.zip 1,074,109 bytes
gweidenh.... That was the thing that had me confused. There was no choices for devices listed. There was no listing. I did a bunch of research and found I needed to go to the command line and become a superuser. This Linux is all new to me. Not really comfortable in the command terminal. Anyway, I became a super user and then it listed 3 things. I chose what I think is the ethernet and started the program running. Then I launched Zoiper. It really started spitting out the lines. There are two things going on. Zoiper is registering the p2p account and it is registering the SIP account. As soon as everything registered I made a call out to a landline. It made the connection and I stayed on for about 15 seconds and then hung up. Then I stopped the wireshark.

In that 2 minutes it created over 5000 lines of stuff. I ran through it but honestly, I don't know what I'm looking at. I saved the file and will try to attach it here. I may fail since it is over a meg. I don't expect you or anybody to rummage through and go line by line. I hope there is some mechanism though that someone can open the file in wireshark and somehow let the program analyze what is happening and maybe something obvious jumps out as a problem.

My wife and I thank everybody for their help. Hopefully this log file gives some clues.


tommyanon

@comcast.net
have you tried to see if 'gmail calling' works?

if you do not have one setup a gmail account and on the left side of your email account will be an option to call phones(you have to install a plug in the first time) for incoming you would need a google voice account as well but not to test and make outgoing calls to US numbers.

if it works from gmail it should work from an obi device as well

gweidenh

join:2002-05-18
Houston, TX
kudos:3
reply to bushguy
I downloaded the file. You can remove the attachment from this post if you would like.

At first glance, it looks like a very normal call. Call setup is correct and the media stream clearly shows two way audio being sent.

You are using G.711 A-law which is a fairly high bandwidth codec. 64kbps in each direction. What type of upload speeds do you get when you do a speedtest.net test? What about a pingtest.net result?

I am concerned that even though you are sending outbound audio, it is not being received, or it is being received too late and being dismissed due to high latency.

A lower bitrate codec (iLBC, GSM, G.729) might improve the upstream connection for you.

Can you take a similar trace with net2phone and let me see the results?


bushguy

@96.63.23.x
reply to SCADAGeo
SCADAGeo... You brought up an interesting experiment. I spent a couple of hours on it. I opened up a callcentric account and set it up in zoiper. I compared the two accounts, callcentric and voipvoip by placing a test call to 1-408-647-4636 which is ulaw echo service. I ran through various configurations of individual codec as well as ports 5060 and 10123. Both accounts register just fine but even though the codec and port were the same each time between accounts, I got very different results.

voipvoip only allowed a connection with a-law and u-law and was one way conversation. I could hear the message and recorded my voice but playback was blips. All other codec gave me error message. Every combination with callcentric gave me a voice recording invalid number or not in service. So in other words, I would have a codec and port selected and then I would place a call from one account and then switch account and place the same call so I was making a fair comparison.

So I was unable to complete any call with callcentric regardless of codec or port. Thanks for the idea.


bushguy

@96.63.23.x
reply to jimk
Jimk... I have not tried any IAX provider but am certainly game to try. I am pretty clueless on the whole comm thing with protocols, codec etc. I'm getting a good education though from this panel. I have found the suggested voip.ms and will look things over tomorrow. It looks pretty interesting. Than you for the suggestion. I'll keep the panel informed if I give that a try.


bushguy

@96.63.23.x
reply to tommyanon
Tommyanon... I'm having trouble keeping up with all the various things I've tried the last year or so. Please correct me if I'm wrong but the suggested gmail calling would be the same as google talk? If so, I'm pretty certain I set up an an account and tried to make some calls. If memory serves me, it was a plugin or something with firefox that allowed a web based calling routine. Let me know if you had something else in mind. Bottom line is what ever web based service I used from Google, it did not work. Thanks for the thought though.

ps. I just checked the plugins on my laptop firefox and I see several google talk plugins and can confirm that service did not work.

Iscream
Premium
join:2009-02-17
New York, NY
kudos:6
Reviews:
·Verizon FiOS
reply to bushguy
I have to admit three facts you may want to take a note of:
1) in order to hear error messages from Callcentric - your internet connection must have an adequate available bandwidth and the codecs of your Zoiper SIP client and Callcentric's servers have matched - otherwise you'd not be able to hear anything or the voice would be severely distorted;

2) the error messages you received clearly spell that your SIP client (Zoiper) has sent WRONG numbers or at least - in a wrong format; for example - some digits could be missing, etc;

3) both #1 & #2 also clearly spell (to me) that [to some great extent] you don't have a NAT problem - otherwise you'd not hear anything.

You may try dialing Callcentric test number 1-777-1234567 (no dashes between numbers ) - if you get the same error message as above - then it's clear that your phone misdialed the number. Otherwise you'd be connected to Tell-Me information system which you may control by dialing DTMF tones (keys) and/or by voice.

You may also call into your own Voice Mail box (make sure you "purchased" one on your account - it's free) - VM is activated by dialing *123 or by dialing your own account's number. You'll be able to hear and follow prompts as well as record messages or greetings and listen to you recordings.

About Callcentric - whenever an account is set to troubleshooting mode - their system keeps recordings of all signalling passed between your device/soft-phone and their servers. Having a trouble-ticket open with their support would probably already have your issue on the way to be resolved. The support hours are 8:00AM till Midnight Easter time.

Oh, b/w - there is NO such thing as NAT problem when working with Callcentric network unless a customer has a special kind of SIP-hunting firewall or buggy SIP router-helper (some were encountered in past, but not any recent cases). No STUN configuration is needed either. On some rare occasions there are known cases of highly asymmetrical bandwidth provided to a customer by their ISP - it was quite sufficient even for G.711 codec on download (you hear voice undistorted) while upload bandwidth was very limited, below any acceptable limits (less than around 100Kbps which is required for normal U/a-law G.711 codec).

You're welcome to send a PM (you need to become registered to this forum) to me if/when you may want any escalation of your trouble-shooting. Somehow I believe that your issue is NO different than any other of millions already resolved for Callcentric customers.

hwittenb

join:2003-12-20
Reviews:
·Future Nine Corp..

1 edit
reply to bushguy
bushguy

As gweidenh said, the WireShark6 trace you posted shows a normal call setup thru sip3.voipvoip.com. The call is using G.711A (A-law) compression for the audio. In the sip signalling the Sip Invite is sent twice after a 500 ms timeout, and the answering Sip 200 OK signal was sent twice after a 1500 ms timeout. I would imagine these timeouts are probably due to the propagation delay or lost packets.

WireShark has some analysis tools that you can try to use to make sense of a call. I guess it varies with the version but my WireShark version (1.8.4) has a top toolbar that says "Telephony". You click "Telephony" and then "VoIP Calls" and WireShark scans your trace and shows you a listing of the voip calls in the trace.

With the call listing, you click on a call to highlight it and then you can click on Flow to see the sip signalling. You could also click on Player then decode and you can see each side of the audio on the call separately. You click on the check box under an audio stream and you can listen to that side of the call. In your case there is audio transmission both ways. You can hear the recorded message coming from the distant location and in the other stream you can hear your voice responding.

I would agree that you should try to get a lower bandwidth codec operating.

Your Zoiper softphone says they support the following codecs.
GSM
G.711 (a-law and u-law)
Speex
iLBC (iLBC 20 and iLBC 30)
G.729 – for Zoiper Communicator BIZ and Zoiper Classic BIZ with g.729

I'm not sure what codecs voipvoip.com supports, however I am sure they support other codecs besides G.711.


bushguy

@96.63.23.x
reply to gweidenh
gweidenh... We truly appreciate you taking a peek at that file. I can certainly remove the file. How do I do that?

One of the first things voipvoip had me do was a speed test. They needed to see at least 40 kbps upload and 40 kbps download. As you know, it is highly variable on speed and latency depending on conditions etc. Here are the results from the other day:

I went to testmy.net and ran the test twice.
2.7 Mbps 125 Kbps
2.0 Mbps 132 Kbps

I waited 10 minutes and ran again once.
2.1 meg down 117 K up

I did not record the ping but it usually ranges in the 800ms area. I did 2 speed test tonight which in my experience is when everyone is on the net and is the slowest. Ping 1 sec ave. , .6 Mbps down , 41Kbps up

I just did a speed test with testmynet. 323 Kbps and 87 Kbps. Not blazing but meeting the minimums required of voipvoip. I have tried the Ilbc-20 and ilbc-30 as well as GSM. None of those codec will even connect. They all give various SIP error messages when trying to place a call. G729 is an option and there was a previous suggestion on a trial I can try for free. I'll look into that. This zoiper has that as a premium feature and I wouldn't mind buying it if I knew it would work.

I will give a try to download wireshark on the windows xp laptop. That is the only functioning telephone(net2phone) we have other than SAT phone which is for emergencies. I hope I don't have to get in to the terminal to do anything. It would make me a bit nervous and don't want to take a chance on crashing the computer. It would be an excellent experiment though. I'll keep you posted on how I make out. Thanks.

Iscream
Premium
join:2009-02-17
New York, NY
kudos:6
Reviews:
·Verizon FiOS
I can't speak about all providers, but Callcentric supports _only_ G.711 and G.729a for calls terminated to PSTN (land and cellular lines). For SIP to SIP calls - almost any codec can be used (they just should match between SIP devices used on both ends of a call).

Now - seeing your stats - I may say that those are on an edge and below of barely acceptable minimums. The download speeds are okay - this is why you can hear error and diagnostic messages without problems.

The latency of 1 sec average coupled with a few lost or not in order delivered (jitter) packets - it may make a voice quality totally unacceptable for human's ear.

Having 41 kbps up - is not sufficient for even for G.729 considering packet jitter and even a small packet loss. The minimal compressed bandwidth of G.729 is 8kbps, but coupled with IP and UDP packet headers required to be passed over Internet - the bandwidth normally goes around 64kbps.

This is why even when your connection "works" you still hear "blip, blip..." on your own recording. If you have a troubleticket open with Callcentric - their system will record everything related to your calls including UDP/RTP packets - it will show your bandwidth available in both directions, jitter and packet loss.

B/w - Callcentric's built-in Windows softphone application supports G.729 codec; you may want to try it (it works only with Callcentric network, not with other providers). Or you may purchase a paid version of X-Lite or 3CX or Zoiper which support G.729.


tommyanon

@comcast.net
reply to bushguy
gtalk and gmail calling are the same service but accessed different ways.

i suspect the reason net2phone works for you is it is a very old service that was popular for use with dialup modems and therefore optimizes for such usage back in the mid 1990's. on thing that may help in your situation is to put a free PBXes account between you and your provider. you should be able to register your soft phone to the PBXes account using GSM CODEC and PBXes will trans code the audio into G711 for you. just be sure not to turn on the 'audio bypass' option which would defeat the point of using PBXes in the first place. otherwise you need to find a provider that support GSM CODEC for PSTN calls, i am not sure who does that if anyone.

gweidenh

join:2002-05-18
Houston, TX
kudos:3
To summarize what others have said:

Your upstream bandwidth is too low too support a G.711 call.
You will need to use a lower bandwidth codec in order to achieve two way audio.
A wireshark trace with net2phone would allow us to see which codec is being used.
As Tommy mentioned, as was going to be my next advise, you could set up a PBX account such that you use a lower bandwidth codec for your calls even if the end provider does not support said codec. This guide might be very helpful. »forum.xda-developers.com/showthr···=2057887
It is intended for and Android Cell phone over a 3G connection, but the principles are the same.


bushguy

@96.63.23.x
reply to Iscream
iscream... sorry, didn't realize your additional post. Good observations. We truly appreciate the offer to debug this and I will register and get you a pm at some point.

As far as callcentric. All I can tell you was that callcentric was one of the first attempts to find another voip provider. I'm hazy on details but within the last year, I signed up, downloaded a softphone from their site ( I think it was callcentrics own?) and tried test calls. It failed and I requested help from support. They made suggestions, made no mention of logfiles or what they thought was the problem and ultimately came back and said they couldn't help us.

Interesting insight on zoiper dialing the test number wrong for callcentric. I'm afraid I may have muddied the waters here. Here are the two numbers I was using. 7033763246 and 14086474636 Both are echo tests that allow me to record my voice. I have the phone number in memory and I have parameters set, go to voipvoip account, paste phone number, make call. Call either goes through or fails depending on codec and port. Then I select callcentric account, paste phone number and place call. If the call connected with voipvoip I would expect callcentric to make the connection too.
But in every case, callcentric connected and gave me the audio message "invalid or not in service (message 3002)"

This morning I used the phone number 17771234567 with port 10123 and made the connection to 411 information. I tried every codec available in zoiper and made the connection just fine. Audio was clear. However, the system could not hear me. Sorry I got this confusing. I assumed the audio echo tests were valid for any SIP based voip service. Obviously not although I don't know why. So anyways, I am able to connect properly to callcentric info line but one way.

Listening to that message from callcentric jars the memory cells. I do remember now talking to the system repeatedly to get weather and the nice lady saying "sorry, I didn't get that" I hope your day goes well.


tommyanon

@comcast.net
reply to Iscream
another thing you may try:

you mention skype works well for p2p calls.

there are a few services including ippi and confytel that offer a way to use there SIP services from a skype client. you would have a p2p skype call to the comfytel or ippi server and the call would go to the PSTN from there. similar concept to the PBX in the middle but you would use skype as the client which has proven to work for you.


bushguy

@96.63.23.x
reply to Iscream
iscream.. Thanks for your input. I understand your comments on upload speed. I'd point out a couple things. I use a couple different speed tests and there doesn't seem to be any rhyme or reason that one day one shows faster speeds, the next time the other shows faster speeds. And the differences seem to be significant. For example last night one site had it at 41, the other site had me more than double that at 87. Night is absolute worst case scenario since everybody is home from work and putzing the internet. I just ran the test again and have 130Kbps upload which is more typical.

Latency might be an issue. I don't mean to beat a dead horse but this is what I'm not understanding. With the speeds and latency issues of my satellite system and I believe xplornet service in general:
1. net2phone always has worked
2. skype has worked quite well especially before latest version updates from previous years
3 zoiper service p2p works fine
4. at least 4 camps I have been in have all had the surfbeam modem with xplornet service, telephone and ATA adapter and every system worked well

I do appreciate your help.


tommyanon

@comcast.net
said by bushguy :

at least 4 camps I have been in have all had the surfbeam modem with xplornet service, telephone and ATA adapter and every system worked well

have you inquired about what xpornet service plan they have and what VOIP provider they are using?


bushguy

@96.63.23.x
reply to gweidenh
gweidenh.. Last piece of the puzzle. Wireshark file for net2phone on windows xp laptop. That went well. Downloaded fine. I started wireshark and then opened net2phone. Then I duplicated the same call I did yesterday on the linux computer. It connected fine and then I spoke for about 15 seconds and then hung up and stopped wireshark.

I'll be very interested in what the difference is between net2phone and zoiper. Thank you.