RE: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?

2004-08-16 Thread Chris Blunt 
I have opened up all the ports specified other than the 1-2 range as
my router just can't cope with that.  Unfortunately I still get no sound.

Is IAX the best route or is registering my FWD connection through SIP.conf
the best solution, what do people recommend?

I have dug through the WIKI, and what instruction I can find, but to no
avail, examples or working confs would be fantastic just for comparison.

Thanks

Again

Chris



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Edward Eastman
Sent: 16 August 2004 00:44
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Inbound Free World Dialup - extension not
ringing?

IAX2 uses udp port 4569, so you’ll probably have to open that up on your
firewall/router.

http://www.voip-info.org/ is a good starting place for any asterisk problems
- specifically:

http://www.voip-info.org/wiki-Asterisk+firewall+rules
http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD

HTH

Ed


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Blunt
Sent: 15 August 2004 23:06
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Inbound Free World Dialup - extension not
ringing?

Hi Lyle, 

Thank you so much for your help, I think your information points to using
IAX2 rather than registering with FWD from the sip.conf

I have made an attempt to understand this, added the appropriate information
into iax.conf, remove old info from sip.conf, gone to fwd and ticked the IAX
registration box, and I now get my local sip phone ringing when I dial in
from FWD!   Hurrah, unfortunately I get no sound in either direction.  Do
you have any experience of this or could it be due to me being inside a NAT
firewall?  I have port 5060 forwarded to my * server, should I forward any
other ports? (I can only forward a maximum 20 single ports due to a
limitation on my home router).

As yet I am unable to make outgoing calls over FWD, I figured I would look
at this next.

Is there a NAT solution that could be used with sip.conf rather than the
IAX?

Again your help is most appreciated.

Best regards

Chris


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese
Sent: 15 August 2004 15:14
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Inbound Free World Dialup - extension not
ringing?

You need a defination for the inbound FWD and what to do with that.
 
In my extensions.conf, I have:
 
[globals]
FWDNUMBER=123456 ; your actual fwd number
FWDCIDNAME='My Name'
FWDPASSWORD=myfwdpasswd
FWDRINGS=sip/office
FWDVMMBOX=1010
 
[fwd_out]
exten = _123.,1,SetCallerId,${FWDCIDNAME}  ; replace 123 with the desired
access code to dial out via FWD
exten =
_123.,2,Dail(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN}:3},60
,r)
exten = _123.,3,Congestion
 
[local]
include = fwd_out  :add to local context
 
[default]
 
;inbound dialing from FWD
exten = ${FWDNUMBER},1,Goto(housemenu,s,1)  ; I have mine set to hit a
menu, no reason you cann't forward to an extension instead
 
- Original Message - 
From: Chris Blunt 
To: [EMAIL PROTECTED] 
Sent: Sunday, August 15, 2004 3:29 AM
Subject: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?


Hi to all the * people out there,

Please kind to me as I am both new to Asterisk and to Linux – But I am
learning fast.

My config is quite simple, I’m just following examples and the Wiki:  I have
two PC’s running X-Lite phones, these work without problems between each
other, and I have a GS BudgeTone-100 registered to Free World Dial UP
(working no problem).

I have tried to set up Asterisk to accept calls from FWD on another number I
have registered, but I can’t get my local X-Lite to ring on an inbound call
from FWD, and I get the busy tone on the BT100

When I sip debug, I can see that I am registered with FWD, and when I call
the number from the BT100 I can see all the incoming information but still
nothing on my X-Lite.

My extensions.conf:


[general]
static=yes
writeprotect=no

[globals]


[sip]
exten = 1,1,Dial(SIP/phone1,20,tr)
exten = 2,1,Dial(SIP/phone2,20,tr)
exten = 2,2,VoiceMail,u1234
exten = 2,102,VoiceMail,b1234
;exten = 1000,1,Dial(SIP/phone1SIP/phone2,20,tr)
exten = 1001,1,Ringing
exten = 1001,2,Wait(2)
exten = 1001,3,VoicemailMain,s1234
exten = 6601,1,WaitMusicOnHold(60)
exten = 232999,1,Dial(SIP/phone1,30,tr)
exten = 232999,2,Hangup


I am behind a NATed fire wall, but I’m not sure that is related.

Any ideas or help (working simple confs) would be much appreciated.



Best regards

--
 
Chris Blunt
 
SIP: [EMAIL PROTECTED]
 
 


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[Asterisk-Users] Inbound Free World Dialup - extension not ringing?

2004-08-15 Thread Chris Blunt 










Hi to all the * people out there,



Please kind to me as I am both new to Asterisk and to Linux
 But I am learning fast.



My config is quite simple, Im just following examples
and the Wiki: I have two PCs running X-Lite phones, these work
without problems between each other, and I have a GS BudgeTone-100 registered
to Free World Dial UP (working no problem).



I have tried to set up Asterisk to accept calls from FWD on
another number I have registered, but I cant get my local X-Lite to ring
on an inbound call from FWD, and I
get the busy tone on the BT100



When I sip debug, I can see that I am registered with FWD,
and when I call the number from the BT100 I can see all the incoming
information but still nothing on my X-Lite.



My extensions.conf:





[general]

static=yes

writeprotect=no



[globals]





[sip]

exten = 1,1,Dial(SIP/phone1,20,tr)

exten = 2,1,Dial(SIP/phone2,20,tr)

exten = 2,2,VoiceMail,u1234

exten = 2,102,VoiceMail,b1234

;exten = 1000,1,Dial(SIP/phone1SIP/phone2,20,tr)

exten = 1001,1,Ringing

exten = 1001,2,Wait(2)

exten = 1001,3,VoicemailMain,s1234

exten = 6601,1,WaitMusicOnHold(60)

exten = 232999,1,Dial(SIP/phone1,30,tr)

exten = 232999,2,Hangup





I am behind a NATed fire wall, but Im not sure that
is related.



Any ideas or help (working simple confs) would be much appreciated.







Best regards



--



Chris Blunt



SIP: [EMAIL PROTECTED]

Re: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?

2004-08-15 Thread Lyle Giese 



You need a defination for the inbound FWD and what 
to do with that.

In my extensions.conf, I have:

[globals]
FWDNUMBER=123456 ;your actual fwd 
number
FWDCIDNAME='My Name'
FWDPASSWORD=myfwdpasswd
FWDRINGS=sip/office
FWDVMMBOX=1010

[fwd_out]
exten = _123.,1,SetCallerId,${FWDCIDNAME} 
; replace 123 with the desired access code to dial out via FWD
exten = 
_123.,2,Dail(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN}:3},60,r)
exten = _123.,3,Congestion

[local]
include = fwd_out :add to local 
context

[default]

;inbound dialing from FWD
exten = 
${FWDNUMBER},1,Goto(housemenu,s,1) ; I have mine set to hit a menu, no 
reason you cann't forward to an extension instead


  - Original Message - 
  From: 
  Chris Blunt 
  To: [EMAIL PROTECTED] 
  
  Sent: Sunday, August 15, 2004 3:29 
  AM
  Subject: [Asterisk-Users] Inbound Free 
  World Dialup - extension not ringing?
  
  
  
  Hi to all the * people out 
  there,
  
  Please kind to me as I am both new 
  to Asterisk and to Linux – But I am learning 
fast.
  
  My config is quite simple, I’m 
  just following examples and the Wiki: I have two PC’s running X-Lite 
  phones, these work without problems between each other, and I have a GS 
  BudgeTone-100 registered to Free World Dial UP (working no 
  problem).
  
  I have tried to set up Asterisk to 
  accept calls from FWD on another number I have registered, but I can’t get my 
  local X-Lite to ring on an inbound call from FWD, and I get the busy tone on the 
  BT100
  
  When I sip debug, I can see that I 
  am registered with FWD, and when I call the number from the BT100 I can see 
  all the incoming information but still nothing on my 
  X-Lite.
  
  My 
  extensions.conf:
  
  
  [general]
  static=yes
  writeprotect=no
  
  [globals]
  
  
  [sip]
  exten = 
  1,1,Dial(SIP/phone1,20,tr)
  exten = 
  2,1,Dial(SIP/phone2,20,tr)
  exten = 
  2,2,VoiceMail,u1234
  exten = 
  2,102,VoiceMail,b1234
  ;exten = 
  1000,1,Dial(SIP/phone1SIP/phone2,20,tr)
  exten = 
  1001,1,Ringing
  exten = 
  1001,2,Wait(2)
  exten = 
  1001,3,VoicemailMain,s1234
  exten = 
  6601,1,WaitMusicOnHold(60)
  exten = 
  232999,1,Dial(SIP/phone1,30,tr)
  exten = 
  232999,2,Hangup
  
  
  I am behind a NATed fire wall, but 
  I’m not sure that is related.
  
  Any ideas or help (working simple 
  confs) would be much appreciated.
  
  
  
  Best 
  regards
  
  --
  
  Chris 
  Blunt
  
  SIP: 
  [EMAIL PROTECTED]

RE: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?

2004-08-15 Thread Edward Eastman 
IAX2 uses udp port 4569, so you’ll probably have to open that up on your
firewall/router.

http://www.voip-info.org/ is a good starting place for any asterisk problems
- specifically:

http://www.voip-info.org/wiki-Asterisk+firewall+rules
http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD

HTH

Ed


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Blunt
Sent: 15 August 2004 23:06
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Inbound Free World Dialup - extension not
ringing?

Hi Lyle, 

Thank you so much for your help, I think your information points to using
IAX2 rather than registering with FWD from the sip.conf

I have made an attempt to understand this, added the appropriate information
into iax.conf, remove old info from sip.conf, gone to fwd and ticked the IAX
registration box, and I now get my local sip phone ringing when I dial in
from FWD!   Hurrah, unfortunately I get no sound in either direction.  Do
you have any experience of this or could it be due to me being inside a NAT
firewall?  I have port 5060 forwarded to my * server, should I forward any
other ports? (I can only forward a maximum 20 single ports due to a
limitation on my home router).

As yet I am unable to make outgoing calls over FWD, I figured I would look
at this next.

Is there a NAT solution that could be used with sip.conf rather than the
IAX?

Again your help is most appreciated.

Best regards

Chris


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese
Sent: 15 August 2004 15:14
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Inbound Free World Dialup - extension not
ringing?

You need a defination for the inbound FWD and what to do with that.
 
In my extensions.conf, I have:
 
[globals]
FWDNUMBER=123456 ; your actual fwd number
FWDCIDNAME='My Name'
FWDPASSWORD=myfwdpasswd
FWDRINGS=sip/office
FWDVMMBOX=1010
 
[fwd_out]
exten = _123.,1,SetCallerId,${FWDCIDNAME}  ; replace 123 with the desired
access code to dial out via FWD
exten =
_123.,2,Dail(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN}:3},60
,r)
exten = _123.,3,Congestion
 
[local]
include = fwd_out  :add to local context
 
[default]
 
;inbound dialing from FWD
exten = ${FWDNUMBER},1,Goto(housemenu,s,1)  ; I have mine set to hit a
menu, no reason you cann't forward to an extension instead
 
- Original Message - 
From: Chris Blunt 
To: [EMAIL PROTECTED] 
Sent: Sunday, August 15, 2004 3:29 AM
Subject: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?


Hi to all the * people out there,

Please kind to me as I am both new to Asterisk and to Linux – But I am
learning fast.

My config is quite simple, I’m just following examples and the Wiki:  I have
two PC’s running X-Lite phones, these work without problems between each
other, and I have a GS BudgeTone-100 registered to Free World Dial UP
(working no problem).

I have tried to set up Asterisk to accept calls from FWD on another number I
have registered, but I can’t get my local X-Lite to ring on an inbound call
from FWD, and I get the busy tone on the BT100

When I sip debug, I can see that I am registered with FWD, and when I call
the number from the BT100 I can see all the incoming information but still
nothing on my X-Lite.

My extensions.conf:


[general]
static=yes
writeprotect=no

[globals]


[sip]
exten = 1,1,Dial(SIP/phone1,20,tr)
exten = 2,1,Dial(SIP/phone2,20,tr)
exten = 2,2,VoiceMail,u1234
exten = 2,102,VoiceMail,b1234
;exten = 1000,1,Dial(SIP/phone1SIP/phone2,20,tr)
exten = 1001,1,Ringing
exten = 1001,2,Wait(2)
exten = 1001,3,VoicemailMain,s1234
exten = 6601,1,WaitMusicOnHold(60)
exten = 232999,1,Dial(SIP/phone1,30,tr)
exten = 232999,2,Hangup


I am behind a NATed fire wall, but I’m not sure that is related.

Any ideas or help (working simple confs) would be much appreciated.



Best regards

--
 
Chris Blunt
 
SIP: [EMAIL PROTECTED]
 
 


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