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 _______________________________________________ --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