[asterisk-users] Voice with echo

[Carlos Alberto Bernat Orozco]

Hi group

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:usuario1 at xxx.xxx.xxx.xxxx>;tag=124002584324
To: <sip:xxx.xxx.xxx.xxx>;tag=as162c41e3
Call-ID: 388DD798-A10D-4CE0-BBCF-57523808EDFF at 10.1.3.164
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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