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mrp40

6 posts

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#73585 16-Dec-2010 15:10
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Hi all,

I have an Asterisk 1.6 box set up with a VFX line for home. It's been working fine for a year or so. A friend of mine registered an account and I said I'd register it on my Asterisk server so he could use it from different devices.

When I try to register OR call out using his account credentials the sip handshake always fails with 401 Unauthorized, from two asterisk servers, both of which are existing setups that work. WxC said that you can't register two numbers on the same server because the port numbers are colliding.

I will get into the details of the issue later if required. But at the moment I know:


  • I can register successfully  with my account from two separate servers at two different physical locations on different networks

  • I can't register with my friends account on either server - even when simply changing the details of my account's config (ie, just one registration)

  • I believe it is an account/authentication setup error at their end

  • They say otherwise


Is anyone here running a single Asterisk server with multiple SIP trunks to WxC VFX?
Is it possible?
Are there any gotchas (other than simple configuration mistakes)?

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maverick
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  #418162 16-Dec-2010 16:17
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401 means unauthorized which means the credentials are incorrect in some way.... send me your details and I will take a look .

This isn't a SIP trunk by the way it's a simple single registration, SIP trunking is completely different





Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well Wink

             

https://www.facebook.com/wxccommunications



mrp40

6 posts

Wannabe Geek


  #418164 16-Dec-2010 16:25
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Thanks maverick. One of your colleagues has contacted me through private message to get the details and confirmed that it was to do with a change to the account setup due to a spelling mistake and we were not issued with an updated email with the new credentials.

Hopefully this comes through shortly and I will update this thread with the outcome.

maverick: This isn't a SIP trunk by the way it's a simple single registration, SIP trunking is completely different

Thanks for pointing that out. I used the term based on what I found in discussions on Google.

maverick
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  #418169 16-Dec-2010 16:34
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No worries




Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well Wink

             

https://www.facebook.com/wxccommunications



mrp40

6 posts

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  #418429 17-Dec-2010 10:46
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Thanks to the VFX team we now have two VFX lines running on our Asterisk server! Laughing It was an account setup issue due to a spelling mistake when the main account username was set up originally.

So yes, Asterisk and VFX does support it without too much of an issue. The only gotcha I found was that you can't put the two VFX SIP accounts in sip.conf into different dialplan contexts from extensions.conf because the incoming calls always seem to go to the default context in extensions.conf. From what I read it's due to the fact that it matches the incoming calls based on ip:port, which is the same for both lines. There may be a way around it but at this point the following works for me.

 sip.conf:

[general]
registertimeout=20
regseconds=180
register=1234567:secret:username@pan.wxnz.net/1234567
register=2345678:secret:username@pan.wxnz.net/2345678

[vfx]
type=peer
fromuser=1234567 ; this is your VFX Number without the leading 0 i.e. 9950XXXX
host=pan.wxnz.net
insecure=invite,port
canreinvite=no
nat=yes
secret=secret
username=username
context=doesntmakeanydifference

[vfx-two]
type=peer
fromuser=2345678 ; this is your VFX Number without the leading 0 i.e. 9950XXXX
host=pan.wxnz.net
insecure=invite,port
canreinvite=no
nat=yes
secret=secret
username=username
context=doesntmakeanydifference

[phone1](natted-phone,my-codecs)
secret=passwd
context=vfx

[phone2](natted-phone,my-codecs)
secret=passwd
context=vfx-two

extensions.conf:

[default]
include => incoming-vfx
include => incoming-vfx-two

[macro-vfx]
exten => s,1,NoOp(Dialing from context ${MACRO_CONTEXT})
; Use the correct outgoing VFX line by matching context to SIP account name 
exten => s,n,Dial(SIP/${MACRO_CONTEXT}/${ARG1})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Hangup
exten => _s-.,1,NoOp

[vfx]
exten => _X.,1,Macro(vfx,${EXTEN})

[vfx-two]
exten => _X.,1,Macro(vfx,${EXTEN})

[incoming-vfx]
exten => 1234567,1,Dial(${EVERYONE}, 40, Tt)

[incoming-vfx-two]
exten => 2345678,1,Dial(${SOMEONE}, 40, Tt)


Obviously that is a simplified dialplan, but you get the idea...

sbiddle
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  #418510 17-Dec-2010 14:03
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mrp40:So yes, Asterisk and VFX does support it without too much of an issue. The only gotcha I found was that you can't put the two VFX SIP accounts in sip.conf into different dialplan contexts from extensions.conf because the incoming calls always seem to go to the default context in extensions.conf. From what I read it's due to the fact that it matches the incoming calls based on ip:port, which is the same for both lines. There may be a way around it but at this point the following works for me.

 


Register them on different ports.


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