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JaymanOffline



Joined: Oct 05, 2006
Posts: 4

Status: Offline
Posted: Aug 19, 2007 - 03:16 PM Reply with quote Back to top
hi guys, i run trixbox and i have a problem with using voipuser to call out and receive calls.

i have a potential solution by adding the NAT block to the config files, that allows me to make the call but every extension and other trunk on the PBX has audio silence, or if i dont add the NAT block then i can use my other trunks and extensions but not use voipuser as a trunk. When i try and make the call i just get a busy error.

I have identified the cause to be a regestration error as the machine has sent teh request but not received an answer.
Quote:

asterisk1*CLI> sip show registry
Host Username Refresh State
voipuser.org:5060 Jayman 120 Request Sent

But when i have NAT infomation (my potential solution) entered the account registers and logs in ok.

Config for that trunk is
Quote:

canreinvite=no
context=from-pstn
fromdomain=voipuser.org
fromuser=Jayman
host=sip.voipuser.org
insecure=very
qualify=no
secret=xxx
type=friend
username=Jayman


REGISTER=>Jayman:xxx@voipuser.org/8449337390


General settings here....
Quote:

Global Settings:
----------------
SIP Port: 5060
Bindaddress: 0.0.0.0
Videosupport: No
AutoCreatePeer: No
Allow unknown access: Yes
Promsic. redir: No
SIP domain support: No
Call to non-local dom.: Yes
URI user is phone no: No
Our auth realm asterisk
Realm. auth: No
Always auth rejects: No
User Agent: Asterisk PBX
MWI checking interval: 10 secs
Reg. context: (not set)
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: Off
IP ToS: 0x68
OSP Support: No
SIP realtime: Disabled

Global Signalling Settings:
---------------------------
Codecs: ulaw,alaw
Relax DTMF: No
Compact SIP headers: No
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: No
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes

Default Settings:
-----------------
Context: from-sip-external
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language: (Defaults to English)
Musicclass: default
Voice Mail Extension: asterisk

----



And there is nothing unusual on the console output in debug mode,

what appears to be the problem?
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