[asterisk-users] Still voice with echo

[Marco Mouta]

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