"Help- SIP number provider said" your asterisk still says "Expires: 120" in registration."

Hello Everyone,
Last month I downloaded and installed Trixbox I believe version:“v2.6.0.0” the latest version I need help changing the to defaultexpiry=60 in sip.conf becaust my SIP number provider said" your asterisk still says “Expires: 120” in registration request. 120 seconds are too much for your router."
I tried to use putty for SSH access my trixbox server to change the setting but I keep getting denied. I was told trixbox uses port 20. I tried port 20 still no access.
Any suggestions would help.

Thanks “fskrotzki” , I will double check the version number again.and try port 22
Is there a better sshd tool available.

Again, Thanks.

ssh uses port 22, NOT 20 unless you’ve done some reconfiguration of sshd on your box.

If you downloaded last month the version shoud have been That’s a BUG in the 2.6 series of tb. They are not updating the version number of the GUI with new releases. Now you’d say no big deal except that trixbox was a totaly hosed release that could not install the base OS correctly to start, so it never installed the rest of the system.

It is important to know what version you really installed for future referance.

Sip.conf get’s overwritten, so you need to add it to /etc/asterisk/sip_general_custom.conf.

If you had read the first two lines of the sip.conf it would have told you that FreePBX owns the SIP.conf and to not edit it as your changes will get lost.

the server side has the best out already, client side is your personal preferance more then anything. For windows I don’t like putty but that is me, I like SecureCRT, but there are MANY choices go google for them.

Would anyone happen to know if there is a program that records or transcode, wave audio files into the trixbox Linear sound format other than using trixbox. I’m not a programmer . Like everyone, else I’m trying to achieve the best audio possible
currently I’m using, Nuendo, soundforge, wavelab, sorenson squeez, and audacity for audio production and know my way around these programs very well.

Also would anyone happen to know how to raise the caller side audio volume.
When I call to test the IVR menu for audio quality the volume is very low.
I 've read quite a few post and because trixbox is so well thought out and packed with features I find it hard to believe there is no obvious way of doing this from the UI " user Interface". Well mabe in future versions of TB just for nonprogramers like me.

Thanks for the help.

fyi: trixbox does not own a audio format. it is nothing more then a collection of existing programs asterisk, Freepbx, etc bundled together.

So… search for things asterisk based will be better.

Also you are much better off startign a new topic then attempting to continue this one as people with audio experiance will not be looking here to help.