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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
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