X-Lite / EyeBeam configuration for registered users, please

[aegyssus]

Hi,
Can anyone please provide the configuration details for using XLite client with a registered account?
I put the DID as username and also as auth username along w/ the passwd but I cannot hear anything. However, I can take the voicemail (*98) with no problem, i.e. I can hear what the operator says and I can dial my options.
Calling works as people at the other end can hear me but I cannot hear anything. I have about ten SIP accounts (Callcentric, VoipUser, JustVoip etc) and all of them work with no problem, so I would seriously doubt that the firewall configuration is part of the problem.
I've tried freephoneline.ca and voip.freephoneline.ca as domains (finally I left freephoneline.ca) and voip.freephoneline.ca as outbound proxy. I've played with all the codecs but no combination worked, so I left them all enabled... Nothing worked so far.
Any suggestion will be highly appreciated.

Many thanks,
aegyssus

[Krazie243]

If a router exists, using softphones usually requires UDP 13000 and UDP 13001 to be open. If they are not open, traffic coming into the router can be blocked.

[aegyssus]

Thanks for your message Krazie243!
I opened up those ports and still nothing.
However, everything is smooth with your FPL client... and for the other SIP providers I do not need to open any special port.
SIP Communicator works too, it's just that XLite refuses to work with FPL...
Thanks for any suggestion,
aegyssus

[Krazie243]

The Freephoneline package comes pre-packaged with the configuration details to connect to the server. X-Lite would need you to enter the SIP settings that can be purchased from us ($50) The X-lite program could possibly use other ports, but that isn't our software. You might want to read the documentation on the X-Lite to find out what restrictions there are in using it. The good thing about the Freephoneline package is that there is a way to test connection and see what is blocking it.

It is good that your Freephoneline package is working, if you've bought the configuration file, feel free to give our call center a call and we could help find out what settings are out of place on the software.

[aegyssus]

Thanks for your message. It's exactly the topic: that's what I meant by "registred users" in the topic's name: those who payed for the configuration.
I've already called the tech support; they keep saying that 13000-13001 port forwarding is the issue. I even sniffed the UDP (RTP and SIP) packets w/ FPL softphone and I see no port used in that range. I just moved on and installed asterisk (trixbox) without touching anything on my router. After one looooong day I made it work and I'm happy.
For those who wanna give a try here is my trixbox configuration:

Code: Select all

PEER DETAILS
============

host=voip.freephoneline.ca
username=1514MY_NUMBER
secret=MY_PASSWD
type=friend
disallow=all
allow=gsm&ulaw
fromdomain=voip.freephoneline.ca
canreinvite=yes
qualify=no
insecure=very


INCOMING SETTINGS
=================
username=1514MY_NUMBER
secret=MY_PASSWD
type=friend
disallow=all
allow=gsm&ulaw
fromdomain=voip.freephoneline.ca
canreinvite=yes
qualify=no
insecure=very

REGISTER STRING
=================
1514MY_NUMBER:MY_PASSWD@voip.freephoneline.ca/1514MY_NUMBER

I know it's not perfect but it works both ways; any comment is welcomed.

Thanks!

[Krazie243]

It is good that you found settings that help solve the problem with that software. Thanks for posting it. It reminded me of another thing: the G729a and G711u codecs are not universal names. The G729a is the one we use the most but under some conditions G711u is used. And its alternate name is the PCMU codec. Some older devices do not use the commonly used numbered system which can lead to confusion. If this is an issue for anyone, just google each of the available codecs to find out which one is the equivalent of what you are looking for.

[clee666]

Thanks for your message. It's exactly the topic: that's what I meant by "registred users" in the topic's name: those who payed for the configuration.
I've already called the tech support; they keep saying that 13000-13001 port forwarding is the issue. I even sniffed the UDP (RTP and SIP) packets w/ FPL softphone and I see no port used in that range. I just moved on and installed asterisk (trixbox) without touching anything on my router. After one looooong day I made it work and I'm happy.
For those who wanna give a try here is my trixbox configuration:

Code: Select all

PEER DETAILS
============

host=voip.freephoneline.ca
username=1514MY_NUMBER
secret=MY_PASSWD
type=friend
disallow=all
allow=gsm&ulaw
fromdomain=voip.freephoneline.ca
canreinvite=yes
qualify=no
insecure=very


INCOMING SETTINGS
=================
username=1514MY_NUMBER
secret=MY_PASSWD
type=friend
disallow=all
allow=gsm&ulaw
fromdomain=voip.freephoneline.ca
canreinvite=yes
qualify=no
insecure=very

REGISTER STRING
=================
1514MY_NUMBER:MY_PASSWD@voip.freephoneline.ca/1514MY_NUMBER

I know it's not perfect but it works both ways; any comment is welcomed.

Thanks!

Hi!
What did you use in the fields Trunk Name and User Context in Trixbox?
Thanks for sharing you infos!

[aegyssus]

sorry for not checking the forum for so long... probably you've fixed it by now but just for the sake of the conversation...
I did not put anything specific there; in my case I put "FPL" but I do not think the trunk name is important. However, it's important to put your phone number in the user context field (e.g. 15145551234).
thanks,