-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Qual g729 você tem? Se você não tem g729 com licença da digium tire ele da lista do allow, tire também o g723.
Silvio Marçal escreveu: > Ele disse que os codes abaixo soa o compativeis? > AGORA ele toca mas quando atende fica mudo. > Att > Silvio > > G729 for ID 18 > PCMU for ID 0 > G723 for ID 4 > PCMA for ID 8 > GSM for ID 3- > > > -- (10 headers 15 lines) --- > Sending to 201.12.106.139 : 5060 (no NAT) > Using INVITE request as basis request - > [EMAIL PROTECTED] > Found peer 'trunk_2' > Found RTP audio format 18 > Found RTP audio format 0 > Found RTP audio format 4 > Found RTP audio format 8 > Found RTP audio format 3 > Found RTP audio format 101 > Peer audio RTP is at port 201.12.106.141:7350 > Found description format G729 for ID 18 > Found description format PCMU for ID 0 > Found description format G723 for ID 4 > Found description format PCMA for ID 8 > Found description format GSM for ID 3 > Found description format telephone-event for ID 101 > Capabilities: us - 0x406 (gsm|ulaw|ilbc), peer - audio=0x10f > (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) > Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 > (telephone-event), combined - 0x1 (telephone-event) > Peer audio RTP is at port 201.12.106.141:7350 > Looking for 627424201 in DID_trunk_2 (domain 200.206.100.61) > list_route: hop: <sip:[EMAIL PROTECTED]:5060> > > <--- Transmitting (no NAT) to 201.12.106.139:5060 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 201.12.106.139:5060;branch=z9hG4bK45ec3eea-26f9-1;received=201.12.106.139 > From: <sip:[EMAIL PROTECTED]:5060>;tag=fff874e6-ensOBJ09977 > To: <sip:[EMAIL PROTECTED]:5060> > Call-ID: [EMAIL PROTECTED] > CSeq: 59350 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:[EMAIL PROTECTED]> > Content-Length: 0 > > > <------------> > -- Executing [EMAIL PROTECTED]:1] Goto("SIP/627424201-08248998", > "default|2002|1") in new stack > -- Goto (default,2002,1) > -- Executing [EMAIL PROTECTED]:1] Dial("SIP/627424201-08248998", > "SIP/2002&IAX2/2002") in new stack > Audio is at 192.168.0.99 port 11324 > Adding codec 0x2 (gsm) to SDP > Adding codec 0x4 (ulaw) to SDP > Adding codec 0x8 (alaw) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > Reliably Transmitting (no NAT) to 192.168.0.19:5060: > INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.99:5060;branch=z9hG4bK3ebd787e;rport > From: "1639116887" <sip:[EMAIL PROTECTED]>;tag=as1f27360e > To: <sip:[EMAIL PROTECTED]:5060> > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Mon, 05 Mar 2007 16:16:58 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 285 > > v=0 > o=root 3207 3207 IN IP4 192.168.0.99 > s=session > c=IN IP4 192.168.0.99 > t=0 0 > m=audio 11324 RTP/AVP 3 0 8 101 > a=rtpmap:3 GSM/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > ----- Original Message ----- > From: "Junior Polegato - Asterisk" <[EMAIL PROTECTED]> > To: <asteriskbrasil@listas.asteriskbrasil.org> > Sent: Saturday, March 03, 2007 6:34 AM > Subject: Re: [AsteriskBrasil] Problemas - TMAIS > > > Silvio Marçal escreveu: >> está assim: >> [TMAIS] >> secret = XXXX >> provider = >> trunkstyle = customvoip >> username = XXXXXXXXXX >> trunkname = Custom - tmais >> callerid = >> hasexten = no >> hassip = yes >> hasiax = no >> registeriax = no >> registersip = yes >> host = sip.tmais.com.br >> dialformat = ${EXTEN:1} >> context = DID_trunk_2 >> group = >> insecure = very >> fromuser = XXXXXXXX >> fromdomain = sip.tmais.com.br >> type = friends >> disallow = all >> allow = g729 >> Silvio >> Silvio Marçal Orlandini >> Jeito Tecnologia [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> >> 16 3911 6887 >> Rua Joaquim Antonio do Nascimento, 108 – CJ 2 >> CEP: 14 024-180 – Jardim Canadá >> Ribeirão Preto – SP >> www.jeito.com.br <http://www.jeito.com.br> >> >> São Paulo >> 11 2626 3957 > > Olá Silvio, > > > No prompt do CLI, sem receber nem fazer nenhuma ligação, digite "sip > debug" (sem aspas), depois faça um ligação, do seu celular por exemplo, > para esse número da TMais e, e quanto notar que recebeu o chamado, > desligue. Verá logo no começos os codes a TMais reconhece e os que você > está querendo usar... Para usar o g729, tem que ter ele licenciado no > Asterisk ou um de teste que tem pela net (não sei como funciona este > último). Então pode ir trocando o "allow=g729" por cada um dos que a > TMais reconhece e testar. Escolha o que melhor lhe servir em > custo/benefício. > - -- "What most profoundly divides two men is a different sense and degree of cleanliness. What help is all honesty and mutual utility, what help is all the good will for each other: in the end the fact remains-they can't stand each other’s smell!" - - Nietzsche -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF7Wj/2QVs8jsa1mQRAvyGAJ9biGB7Z0CQKG9u25gIizRRLjShFQCdFvTN 0Zk42EoRUjtiLHaLZRrf41I= =+vMf -----END PGP SIGNATURE----- ---------------------------------------- Estação VoIP 2006 5 e 6 Dezembro Curitiba PR http://www.estacaovoip.com.br _______________________________________________ LIsta de discussões AsteriskBrasil.org AsteriskBrasil@listas.asteriskbrasil.org http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil _______________________________________________ Acesse o wiki AsteriskBrasil.org: http://www.asteriskbrasil.org