I have my * box installed with a public IP address and I'm testing two extensions. I'm using SJphone for softphone. When I make the call from an extension to another, the voice sounds with echo. Besides sounds like creepy and it seems like a radio (for making a description)
This is my sip.conf :
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
User Agent: Asterisk PBX
MWI checking interval: 10 secs
Reg. context: (not set)
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
IP ToS: 0x0
OSP Support: No
SIP realtime: Disabled
Global Signalling Settings:
---------------------------
Codecs: gsm,ulaw
Relax DTMF: No
Compact SIP headers: No
RTP Timeout: 60
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
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: default
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language: (Defaults to English)
Musicclass: default
Voice Mail Extension: asterisk
And these are my extensions:
;***************** extension de usuario 1 ******************
exten => 2426098,1,dial(SIP/usuario1)
exten => usuario1,1,goto(2426098,1) ; To be able to dial with text, "usuario1"
;***************** extension de usuario 2 ******************
exten => 2418150,1,dial(SIP/usuario2)
exten => usuario2,1,goto(2418150,1) ; To be able to dial with text, "usuario2"
This is an output for the conversation: ********************
--- (8 headers 0 lines)---
Looking for 200.30.115.163 in default (domain )
Transmitting (no NAT) to 10.1.3.164:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.1.3.164;rport;branch=z9hG4bK0a0103a40000001044c56ac500006b480000061a;received=10.1.3.164
From: <sip:[EMAIL PROTECTED]>;tag=124002584324
To: <sip:xxx.xxx.xxx.xxx>;tag=as162c41e3
Call-ID: [EMAIL PROTECTED]
CSeq: 222 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:xxx.xxxx.xxxx.xxxx >
Accept: application/sdp
Content-Length: 0
I don't know if there is some problem with the codecs or on my configuration. Do I have to change some line?
Thanks for any help.
Carlos Bernat
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