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uk.telecom.voip (UK VOIP) (uk.telecom.voip) Discussion of topics relevant to packet based voice technologies including Voice over IP (VoIP), Fax over IP (FoIP), Voice over Frame Relay (VoFR), Voice over Broadband (VoB) and Voice on the Net (VoN) as well as service providers, hardware and software for use with these technologies. Advertising is not allowed.

[email protected] with Safecom SDSIP-5000 phone



 
 
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  #1  
Old April 27th 06, 11:49 PM posted to uk.telecom.voip
linker3000
external usenet poster
 
Posts: 75
Default [email protected] with Safecom SDSIP-5000 phone

I am on a steep learning curve with [email protected] - I have nearly got
things working and would appreciate any insight from anyone who's using
this phone with [email protected]

My main question is what do you put in the Username / Phone no field in
the SIP settings - when I was connecting directly to my VoIP provider
this was 'userid @voip.co.uk' - eg: '123456 @voip.co.uk' but what do I
use for the Asterisk server?

The phone was registering this afternoon but I cannot make outbound
calls - I was getting a lovely female voice telling me that all trunks
were busy (or something like that), but having tweaked further all I get
now is a beep-beep tone and the phone displays something like 'not
found' even though it's still registered - can't be more specific 'cos I
am posting from home and the phone's in the office.

All useful Asterisk tips welcome. I'm off to check the wiki etc.

Thanks
  #2  
Old April 28th 06, 12:38 AM posted to uk.telecom.voip
paul123
external usenet poster
 
Posts: 189
Default [email protected] with Safecom SDSIP-5000 phone


linker3000 wrote:
I am on a steep learning curve with [email protected] - I have nearly got
things working and would appreciate any insight from anyone who's using
this phone with [email protected]

My main question is what do you put in the Username / Phone no field in
the SIP settings - when I was connecting directly to my VoIP provider
this was 'userid @voip.co.uk' - eg: '123456 @voip.co.uk' but what do I
use for the Asterisk server?

The phone was registering this afternoon but I cannot make outbound
calls - I was getting a lovely female voice telling me that all trunks
were busy (or something like that), but having tweaked further all I get
now is a beep-beep tone and the phone displays something like 'not
found' even though it's still registered - can't be more specific 'cos I
am posting from home and the phone's in the office.

All useful Asterisk tips welcome. I'm off to check the wiki etc.

Thanks


I'm not using that phone but....(and I'm assuming you've set the phone
up as an extension of *) it sounds likeit could be an Outbound routing
problem.

What version are you using? 2.8?

If so, in FreePBX go to ToolsAsterisk info and check that both the
trunk and the extension you're using are registered.

If so, in SetupOutbound routes you need to set up the outbound routing
to point to your voip.co.uk trunk and in that trunk you need to check
your "Outgoing Dial Rules".

This isn't necessarily a solution, but a starting point.

  #3  
Old April 28th 06, 08:07 AM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 1
Default [email protected] with Safecom SDSIP-5000 phone

linker3000 wrote:
I am on a steep learning curve with [email protected] - I have nearly got
things working and would appreciate any insight from anyone who's using
this phone with [email protected]

Only from people using that phone? (I'm not)

My main question is what do you put in the Username / Phone no field in
the SIP settings - when I was connecting directly to my VoIP provider
this was 'userid @voip.co.uk' - eg: '123456 @voip.co.uk' but what do I
use for the Asterisk server?

The username is your Asterisk extension number for the phone.
The password is the extension's secret.

The phone was registering this afternoon but I cannot make outbound
calls - I was getting a lovely female voice telling me that all trunks
were busy (or something like that), but having tweaked further all I get
now is a beep-beep tone and the phone displays something like 'not
found' even though it's still registered - can't be more specific 'cos I
am posting from home and the phone's in the office.

What have you tweaked? Have you been playing with AMP/freePBX or have
you been delving into the .conf files?

Check your outbound routes; reboot your phone; undo what you tweaked;

All useful Asterisk tips welcome. I'm off to check the wiki etc.

voxilla is as good as the wiki, not forgetting http://nerdvittles.com
as well.

Thanks

  #4  
Old April 29th 06, 03:09 PM posted to uk.telecom.voip
linker3000
external usenet poster
 
Posts: 75
Default [email protected] with Safecom SDSIP-5000 phone

paul123 wrote:
linker3000 wrote:
I am on a steep learning curve with [email protected] - I have nearly got
things working and would appreciate any insight from anyone who's using
this phone with [email protected]

My main question is what do you put in the Username / Phone no field in
the SIP settings - when I was connecting directly to my VoIP provider
this was 'userid @voip.co.uk' - eg: '123456 @voip.co.uk' but what do I
use for the Asterisk server?

The phone was registering this afternoon but I cannot make outbound
calls - I was getting a lovely female voice telling me that all trunks
were busy (or something like that), but having tweaked further all I get
now is a beep-beep tone and the phone displays something like 'not
found' even though it's still registered - can't be more specific 'cos I
am posting from home and the phone's in the office.

All useful Asterisk tips welcome. I'm off to check the wiki etc.

Thanks


I'm not using that phone but....(and I'm assuming you've set the phone
up as an extension of *) it sounds likeit could be an Outbound routing
problem.

What version are you using? 2.8?

If so, in FreePBX go to ToolsAsterisk info and check that both the
trunk and the extension you're using are registered.

If so, in SetupOutbound routes you need to set up the outbound routing
to point to your voip.co.uk trunk and in that trunk you need to check
your "Outgoing Dial Rules".

This isn't necessarily a solution, but a starting point.

Thanks for the advice guys - things are getting better - the phones
register and we can dial between extensions. I ended up with the obvious
in the username / phone field - ie: (the server's IP
address) - which I had tried before but probably some other part of my
setup was causing problems.

Inbound calls also work but no outbound yet - probably a rules issue -
also no voicemail but group hunting works.

I'll get there!

 




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