Working Perfectly

[Bob432]

Hi,

 

I thought I'd get back here and report on progress for me. All works perfectly!

 

It did take quite a bit of work - but it's well worth the effort.

 

Both directions work! skype > sip and sip > skype

 

I use an epygi pbx.

 

So, it's just a little bit of feedback for those folks who keep trying to get it work - keep trying...for me, it finally did all work.

 

This forum doesn't allow for screenshots to be posted - but if someone wants, I guess I could organise something, somewhere. I think that would help as a hint for the next version/docs.

 

Things that helped for me were:

 

Within Skype app, have all audioin/out/ringing set to Uplink driver. And as the help docs state, make sure the computer audio settings are not set to uplink, but to your real sound card.

 

Within Skype app, privacy, manage other programs access - make sure Uplink.exe is allowed to use Skype. Then in Advanced, don't have automatically answer incoming calls. Have Associate Skype with callto:links (if that makes any difference I don't know).

 

In Uplink...Network, I turned off all switches (for me it seemed this worked best, as when they were on I had audio only in 1 direction) - and changed local sip port to 5060.

 

In Sip setup just enter the details for the user account setup on the pbx side. All of that was easy. Same on the pbx end, you want the pbx to register the pc (ip address) where Uplink is setup.

 

 

So, skype to sip went fine.

 

The trick to get sip to skype to work (for me) is in uplink/general set When Sips calls use the dialled number...then...in your handset (or perhaps at the pbx itself) you need to add a quickdial that includes the ext number for uplink (in my case it's just 4), then the username for skype. So, if I want to call skype user name "joe", my quickdial just shows 4joe. Most phones I would assume could be set to enter in such a quickdial. I use Aastra handsets, and the web based admin for these phones allow for quickdials.

 

Anyway, I hope this helps. If I can expand on any of this, I'd be happy to.

 

Uplink always complains that register attempts fail, etc. but that doesn't bother me, as it just all works - in both directions.

 

Nice work to the NCH guys.

[lnb2g]

Hi,

 

I thought I'd get back here and report on progress for me. All works perfectly!

 

It did take quite a bit of work - but it's well worth the effort.

 

Both directions work! skype > sip and sip > skype

 

I use an epygi pbx.

 

So, it's just a little bit of feedback for those folks who keep trying to get it work - keep trying...for me, it finally did all work.

 

This forum doesn't allow for screenshots to be posted - but if someone wants, I guess I could organise something, somewhere. I think that would help as a hint for the next version/docs.

 

Things that helped for me were:

 

Within Skype app, have all audioin/out/ringing set to Uplink driver. And as the help docs state, make sure the computer audio settings are not set to uplink, but to your real sound card.

 

Within Skype app, privacy, manage other programs access - make sure Uplink.exe is allowed to use Skype. Then in Advanced, don't have automatically answer incoming calls. Have Associate Skype with callto:links (if that makes any difference I don't know).

 

In Uplink...Network, I turned off all switches (for me it seemed this worked best, as when they were on I had audio only in 1 direction) - and changed local sip port to 5060.

 

In Sip setup just enter the details for the user account setup on the pbx side. All of that was easy. Same on the pbx end, you want the pbx to register the pc (ip address) where Uplink is setup.

So, skype to sip went fine.

 

The trick to get sip to skype to work (for me) is in uplink/general set When Sips calls use the dialled number...then...in your handset (or perhaps at the pbx itself) you need to add a quickdial that includes the ext number for uplink (in my case it's just 4), then the username for skype. So, if I want to call skype user name "joe", my quickdial just shows 4joe. Most phones I would assume could be set to enter in such a quickdial. I use Aastra handsets, and the web based admin for these phones allow for quickdials.

 

Anyway, I hope this helps. If I can expand on any of this, I'd be happy to.

 

Uplink always complains that register attempts fail, etc. but that doesn't bother me, as it just all works - in both directions.

 

Nice work to the NCH guys.

 

Since yo have it to work perfectly, i have a question.

 

I have it set up where someone call me on skype it ring my express talk but i can never pick it up it said

"something is wrong with your NAT device"

"it always assign 'odd' ports which are forbidden for rtp"

"check your NAT or turn off STUN"

 

do i have to do any port forwarding? or anything like that?

 

thanks in advance

[Bob432]

Since yo have it to work perfectly, i have a question.

 

I have it set up where someone call me on skype it ring my express talk but i can never pick it up it said

"something is wrong with your NAT device"

"it always assign 'odd' ports which are forbidden for rtp"

"check your NAT or turn off STUN"

 

do i have to do any port forwarding? or anything like that?

 

thanks in advance

 

 

I'm not sure - and not an expert with this...but what did work for me was to uncheck all the switches under the Network tab - and change the default local Sip port to 5060 (which is the default for everything on SIP), and leave local RTP at 8000.

 

My main reason for posting originally was to give hope to those that tried and failed. I tried very hard to get this to work - and with persistence it finally did.

 

Good luck with it.

[lnb2g]

I'm not sure - and not an expert with this...but what did work for me was to uncheck all the switches under the Network tab - and change the default local Sip port to 5060 (which is the default for everything on SIP), and leave local RTP at 8000.

 

My main reason for posting originally was to give hope to those that tried and failed. I tried very hard to get this to work - and with persistence it finally did.

 

Good luck with it.

what do you mean by switches? unchecked all the stun servers?

 

thanks

[Bob432]

what do you mean by switches? unchecked all the stun servers?

 

thanks

 

 

Yes - uncheck all those boxes.

[lnb2g]

Yes - uncheck all those boxes.

I unchecked all the stun servers and have it "Local SIP Port to listen on 5060" for all of them (pbx, and both of the express talk) now it give me a different error "error call attempt timed-out" and "the server is not replying"

 

even though i have them all listen on port 5060 as i mentioned above but the axon is listening port 5060 and the etalk on the same machine is on port 5061 (eg. 192.xxx.xxx.19:5060, and 192.xxx.xxx.19:5061) and the other etalk is on 192.xxx.xxx.21:5060), can you email all or you screen shots if you don mine. they might help

 

Thanks