my mistake you post it! could you pos it in file.conf format?
On 7/25/06, Marco Mouta <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> 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:[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 > > > > Thanks for any help > > > Carlos bernat > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- Best regards, Marco Mouta
-- Best regards, Marco Mouta _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
