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;received10.1.3.164 From: <sip:usuario1@xxx.xxx.xxx.xxxx>;tag=124002584324 To: <sip:xxx.xxx.xxx.xxx>;tag=as162c41e3 Call-ID: 388DD798-A10D-4CE0-BBCF-57523808EDFF@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
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@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@xxx.xxx.xxx.xxxx>;tag=124002584324 > To: <sip:xxx.xxx.xxx.xxx>;tag=as162c41e3 > Call-ID: 388DD798-A10D-4CE0-BBCF-57523808EDFF@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 > > _______________________________________________ > --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
Carlos Alberto Bernat Orozco 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 had similar echo problems using my laptop but it was resolved by a combination of using a headset (mic and headphones) and turning down the mic gain. It takes very little pickup from the speakers to the mic to cause an objectionable amount of background echo/noise so possibly just moving the mic away some is not sufficient. Also you might make sure that the PC has enough power to deliver the sound fast enough (would the echo test in asterisk check this?). That is, if I use an old slow box that does not run a soft phone very well you will hear slurring of the sound probably as the jitter buffer tries to compensate for the breaks in the data stream? Mike
First at all, thanks guys for the support!! I've been doing what people told me. To asure that I have DirectX on SJPhone (audio setting option enable DirectX 8.1) and I can't run fxotune because I don't use this cards (sorry if I'm wrong). I'm just trying to probe my * box with the voip-info.org tutorials. I turn down the mic gain and the pc's has the enough power. I make the echo test on the 2 clients (dialing 500 on *) and it sounds great (fluid voice). This is my [general]sip.conf format: I omitted other parts which were on comments because are examples from the web site [general] context=default ;allowguest=no ;realm=mydomain.tld bindport=5060 bindaddr=0.0.0.0 srvlookup=yes ;domain=mydomain.tld ;************** Cambio de lineas disallow=all ;allow=g729 allow=gsm allow=ulaw jitterbuffer=yes maxjitterbuffer=800 ;allow=ilbc ;musicclass=default ;language=en ;relaxdtmf=yes rtptimeout=60 ;rtpholdtimeout=300 ;trustrpid = no ;sendrpid = yes ;progressinband=never ;useragent=Asterisk PBX ;promiscredir = no ;usereqphone = no ;*********** Cambio de lineas DTMFMODE estaba en comentarios ******************** dtmfmode = rfc2833 ;compactheaders = yes ;sipdebug = yes ;subscribecontext = default ;notifyringing = yes ;******************** 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 And again thanks for the help! Carlos Bernat -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060725/c49e6f0d/attachment.htm