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- Jul 3, 2008
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hello,
i have two computers, one with windows and 3CX and the other with linux and Asterisk server. I was able to add the Asterisk server as a voip provider to the 3CX, but not yet the other was round. firstable I created an extension in 3CX(username=callerid=1030. pwd=Phone) and then i tried to add this extension to asterisk in the following way:
register => [email protected]
hone:[email protected]
[1030]
type=peer
context=3CX
secret=Phone
username=1030
;auth=user@realm
host=192.168.1.253
fromuser=1030
fromdomain=192.168.0.254
insecure=very
nat=yes
reinvite=no
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
callerid="1030"<1030>
where 192.168.1.253 is the IP adress of the windows computer on which 3CX is hosted and 192.168.0.254 is of the linux computer. I was able to log in from the linux computer via SJPhone, so my question how should the register command, I mean register => [email protected]
hone:[email protected], look like so that the 3CX accepts the new user. In deed there are many forms depending on the sip server, e.g. username
wd@sipserver[
ort] or username@sipserver
wd:username@proxy ans so on. So what is the standard by 3CX.
here some further information about how it is stored on the asterisk server:
* Name : 1030
Secret : <Set>
MD5Secret : <Not set>
Context : 3CX
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
FromUser : 1030
FromDomain : 192.168.0.254
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Dynamic : No
Callerid : "E1-PH00" <1000>
MaxCallBR : 384 kbps
Expire : -1
Insecure : port,invite
Nat : Always
ACL : No
T38 pt UDPTL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Video Support: No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
LastMsg : 0
ToHost : 192.168.1.253
Addr->IP : 192.168.1.253 Port 5060
Defaddr->IP : 0.0.0.0 Port 0
Def. Username: 1030
SIP Options : (none)
Codecs : 0x40e (gsm|ulaw|alaw|ilbc)
Codec Order : (ulaw:20,alaw:20,gsm:20,ilbc:30)
Auto-Framing: No
Status : Unmonitored
Useragent :
Reg. Contact :
i have two computers, one with windows and 3CX and the other with linux and Asterisk server. I was able to add the Asterisk server as a voip provider to the 3CX, but not yet the other was round. firstable I created an extension in 3CX(username=callerid=1030. pwd=Phone) and then i tried to add this extension to asterisk in the following way:
register => [email protected]
[1030]
type=peer
context=3CX
secret=Phone
username=1030
;auth=user@realm
host=192.168.1.253
fromuser=1030
fromdomain=192.168.0.254
insecure=very
nat=yes
reinvite=no
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
callerid="1030"<1030>
where 192.168.1.253 is the IP adress of the windows computer on which 3CX is hosted and 192.168.0.254 is of the linux computer. I was able to log in from the linux computer via SJPhone, so my question how should the register command, I mean register => [email protected]
here some further information about how it is stored on the asterisk server:
* Name : 1030
Secret : <Set>
MD5Secret : <Not set>
Context : 3CX
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
FromUser : 1030
FromDomain : 192.168.0.254
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Dynamic : No
Callerid : "E1-PH00" <1000>
MaxCallBR : 384 kbps
Expire : -1
Insecure : port,invite
Nat : Always
ACL : No
T38 pt UDPTL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Video Support: No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
LastMsg : 0
ToHost : 192.168.1.253
Addr->IP : 192.168.1.253 Port 5060
Defaddr->IP : 0.0.0.0 Port 0
Def. Username: 1030
SIP Options : (none)
Codecs : 0x40e (gsm|ulaw|alaw|ilbc)
Codec Order : (ulaw:20,alaw:20,gsm:20,ilbc:30)
Auto-Framing: No
Status : Unmonitored
Useragent :
Reg. Contact :