[asterisk-users] Still voice with echo

[Carlos Alberto Bernat Orozco]

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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060725/2ba8650a/attachment.htm