[asterisk-users] Still voice with echo

[Marco Mouta]

my mistake you post it! could you pos it in file.conf format?

On 7/25/06, Marco Mouta <marco.mouta at gmail.com> wrote:
> It seems you didn't post any thing about you [general] sip.conf
> neither allowed codecs
>
> On 7/25/06, Carlos Alberto Bernat Orozco <cabo81 at gmail.com> wrote:
> > Hi group
> >
> > Thanks Marty for your colaboration. I tried the my voice call with 2
> > extensions and SJphone as softphone as you know. For the test I used a
> > normal mic plug into the mic port from a laptop and made the call to another
> > pc wich has second extension. At first time I believed what you told me
> > about the feedback, but it's constant no matter if I put away from the
> > speakers. The voice sounds with echo and keeps constants when I say :"hello"
> > and sound very bad.
> >
> > I did this test on march of this year with the same configuration and it
> > sounds great but yesterday when I made a test again the voice was like I
> > just explain.
> >
> > I giving you again pieces of my sip.conf (with the two extensions wich I
> > didn't put in the other e-mail...)
> >
> > I don't know but I thinking on the type of dtmfmode as the main suspect...
> >
> > ;******************** Usuario 1 ************************
> > [usuario1]
> > type=friend
> >  host=dynamic
> >  dtmfmode=rfc2833
> >  username=usuario1
> >  secret=usuario1
> >
> >
> >
> > ;******************** Usuario 2 ************************
> > [usuario2]
> > type=friend
> >  host=dynamic
> >  dtmfmode=rfc2833
> >  username=usuario2
> >  secret=usuario2
> >
> >
> > 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 xxx.xxx.xxx.xxx 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
> >
> >
> >
> > Thanks for any help
> >
> >
> > Carlos bernat
> >
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>
>
> --
> Best regards,
>
> Marco Mouta
>


-- 
Best regards,

Marco Mouta