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telefono que me tiene de los nervios Planet Vip154NT

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  • #1

    telefono que me tiene de los nervios Planet Vip154NT

    hola bueno me facilitaron para poder aprender esto de la telefonia ip un cacharro de la Planet el VIP-154NT, la cosa esta en que todo perfecto pero no se escucha la voz del menu, ni voicemail, en fin ni Playback() ni BAckground() si embargo cuando llamas con el y la persona te pone en HOLD se escucha perfectamente la musica de HOLD pensando que era cosa de formatos probe hace un Playback() con uno de esos track y nada por gusto el contador del telefono siempre comienza a marcar pero nereida se escucha, eso si las llamadas se escuchan perfecto, le instale su ultimo firmware y auentaron la lista de codecs de audio soportados pero al final solo uso en mi sistema ulaw,,,,se probo con softphones y otros aparatos y todo perfecto, el lio es con ese modelo, bueno felicidadea por el 2013 a todos/as

    porcierto en la consola de pbx me sale:
    WARNIGN[360]: rtp.c:462 ast_rtp_read: RTP Read error: Conecction reset by peer. Hanging up now.
    Editado por última vez por venenodcuba; https://asteriskmx.org/foros/member/1647-venenodcuba en 01-03-2013, 08:37 AM. Razón: [Solucionado]
    Etiquetas: Ninguno
  • #2
    Hola todavía crudos por acá. Con el sip debug podríamos saber un poco más de la llamada.

    Comentario

    • #3
      Asegúrate de hacer un Answer() antes de cualquier reproducción de audio para garantizar que el canal de audio se abra antes y no sea esa la causa del problema
      dCAP Christian Cabrera R.
      Para aprender a usar Asterisk, asiste a uno de mis cursos Asterisk
      Si deseas asesoría pagada, por favor contáctame

      Comentario

      • #4
        siempre utilizo Answer(), resultado del debug:
        Código:
        <-- SIP read from 10.90.87.205:5060: 
ACK sip:113@10.90.87.122 SIP/2.0
Via: SIP/2.0/UDP 10.90.87.205:5060;branch=z9hG4bKafb115a26e
From: "Laboratorio" <sip:lab@10.90.87.122>;tag=31407a8e
To: <sip:113@10.90.87.122>;tag=as1220ce93
Call-ID: 190fce105ca1894a713c88c83d383c8c@10.90.87.205
Contact: <sip:lab@10.90.87.205:5060>
CSeq: 801 ACK
Max-Forwards: 70
Content-Length: 0


Jan  1 23:08:20 VERBOSE[4924] logger.c: --- (9 headers 0 lines) ---
Jan  1 23:08:20 DEBUG[4924] chan_sip.c: Stopping retransmission on '190fce105ca1894a713c88c83d383c8c@10.90.87.205' of Response 801: Match Found
Jan  1 23:08:20 VERBOSE[4924] logger.c: 
<-- SIP read from 10.90.87.205:5060: 
INVITE sip:113@10.90.87.122 SIP/2.0
Via: SIP/2.0/UDP 10.90.87.205:5060;branch=z9hG4bKb5fbc3ce4c
From: "Laboratorio" <sip:lab@10.90.87.122>;tag=31407a8e
To: <sip:113@10.90.87.122>
Call-ID: 190fce105ca1894a713c88c83d383c8c@10.90.87.205
Contact: <sip:lab@10.90.87.205:5060>
CSeq: 802 INVITE
Proxy-Authorization: Digest username="lab",realm="asterisk",nonce="652e7059",response="3fa8f8cde179cc779ef674f4cdbba2f4",uri="sip:113@10.90.87.122",algorithm=MD5
Max-Forwards: 70
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
User-Agent: VIP-154NT
Content-Length: 195

v=0
o=- 10952 0 IN IP4 10.90.87.205
s=SIP CALL
c=IN IP4 10.90.87.205
t=0 0
m=audio 60000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

Jan  1 23:08:20 VERBOSE[4924] logger.c: --- (13 headers 10 lines) ---
Jan  1 23:08:20 VERBOSE[4924] logger.c: Using INVITE request as basis request - 190fce105ca1894a713c88c83d383c8c@10.90.87.205
Jan  1 23:08:20 VERBOSE[4924] logger.c: Sending to 10.90.87.205 : 5060 (non-NAT)
Jan  1 23:08:20 DEBUG[4924] chan_sip.c: Setting NAT on RTP to 0
Jan  1 23:08:20 VERBOSE[4924] logger.c: Found user 'lab'
Jan  1 23:08:20 VERBOSE[4924] logger.c: Found RTP audio format 0
Jan  1 23:08:20 VERBOSE[4924] logger.c: Found RTP audio format 101
Jan  1 23:08:20 VERBOSE[4924] logger.c: Peer audio RTP is at port 10.90.87.205:60000
Jan  1 23:08:20 DEBUG[4924] chan_sip.c: Peer audio RTP is at port 10.90.87.205:60000
Jan  1 23:08:20 VERBOSE[4924] logger.c: Found description format PCMU
Jan  1 23:08:20 VERBOSE[4924] logger.c: Found description format telephone-event
Jan  1 23:08:20 VERBOSE[4924] logger.c: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Jan  1 23:08:20 VERBOSE[4924] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Jan  1 23:08:20 DEBUG[4924] chan_sip.c: Checking SIP call limits for device lab
Jan  1 23:08:20 VERBOSE[4924] logger.c: Looking for 113 in users2 (domain 10.90.87.122)
Jan  1 23:08:20 DEBUG[4924] chan_sip.c: build_route: Contact hop: <sip:lab@10.90.87.205:5060>
Jan  1 23:08:20 VERBOSE[4924] logger.c: list_route: hop: <sip:lab@10.90.87.205:5060>
Jan  1 23:08:20 VERBOSE[4924] logger.c: Transmitting (no NAT) to 10.90.87.205:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.90.87.205:5060;branch=z9hG4bKb5fbc3ce4c;received=10.90.87.205
From: "Laboratorio" <sip:lab@10.90.87.122>;tag=31407a8e
To: <sip:113@10.90.87.122>
Call-ID: 190fce105ca1894a713c88c83d383c8c@10.90.87.205
CSeq: 802 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:113@10.90.87.122>
Content-Length: 0


---
Jan  1 23:08:20 DEBUG[4924] channel.c: Avoiding initial deadlock for 'SIP/lab-009b7b60'
Jan  1 23:08:20 DEBUG[4924] pbx.c: Expression result is '0'
Jan  1 23:08:20 DEBUG[4924] pbx.c: Expression result is '0'
Jan  1 23:08:20 DEBUG[4924] channel.c: Avoiding initial deadlock for 'SIP/lab-009b7b60'
Jan  1 23:08:20 VERBOSE[4924] logger.c: We're at 10.90.87.122 port 11090
Jan  1 23:08:20 VERBOSE[4924] logger.c: Adding codec 0x4 (ulaw) to SDP
Jan  1 23:08:20 VERBOSE[4924] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Jan  1 23:08:20 DEBUG[4924] channel.c: Avoiding initial deadlock for 'SIP/lab-009b7b60'
Jan  1 23:08:20 VERBOSE[4924] logger.c: Reliably Transmitting (no NAT) to 10.90.87.205:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.90.87.205:5060;branch=z9hG4bKb5fbc3ce4c;received=10.90.87.205
From: "Laboratorio" <sip:lab@10.90.87.122>;tag=31407a8e
To: <sip:113@10.90.87.122>;tag=as39c5c910
Call-ID: 190fce105ca1894a713c88c83d383c8c@10.90.87.205
CSeq: 802 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:113@10.90.87.122>
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 4924 4924 IN IP4 10.90.87.122
s=session
c=IN IP4 10.90.87.122
t=0 0
m=audio 11090 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Jan  1 23:08:20 WARNING[4924] rtp.c: RTP Read error: Connection reset by peer.  Hanging up now.
Jan  1 23:08:20 DEBUG[4924] chan_sip.c: update_call_counter(lab) - decrement call limit counter
Jan  1 23:08:20 VERBOSE[4924] logger.c: Scheduling destruction of call '190fce105ca1894a713c88c83d383c8c@10.90.87.205' in 32000 ms
Jan  1 23:08:20 DEBUG[4924] pbx.c: Function result is '"Lab" <30>'
Jan  1 23:08:20 DEBUG[4924] pbx.c: Function result is '30'
Jan  1 23:08:20 DEBUG[4924] pbx.c: Function result is '113'
Jan  1 23:08:20 DEBUG[4924] pbx.c: Function result is 'users'
Jan  1 23:08:20 DEBUG[4924] pbx.c: Function result is 'SIP/lab-009b7b60'
Jan  1 23:08:20 DEBUG[4924] pbx.c: Function result is '(null)'
Jan  1 23:08:20 DEBUG[4924] pbx.c: Function result is 'Playback'
Jan  1 23:08:20 DEBUG[4924] pbx.c: Function result is 'mios/cali'
Jan  1 23:08:20 DEBUG[4924] pbx.c: Function result is '2013-01-01 23:08:20'
Jan  1 23:08:20 DEBUG[4924] pbx.c: Function result is '2013-01-01 23:08:20'
Jan  1 23:08:20 DEBUG[4924] pbx.c: Function result is '2013-01-01 23:08:20'
Jan  1 23:08:20 DEBUG[4924] pbx.c: Function result is '0'
Jan  1 23:08:20 DEBUG[4924] pbx.c: Function result is '0'
Jan  1 23:08:20 DEBUG[4924] pbx.c: Function result is 'ANSWERED'
Jan  1 23:08:20 DEBUG[4924] pbx.c: Function result is 'DOCUMENTATION'
Jan  1 23:08:20 DEBUG[4924] pbx.c: Function result is '(null)'
Jan  1 23:08:20 DEBUG[4924] pbx.c: Function result is '1357099700.2'
Jan  1 23:08:20 DEBUG[4924] pbx.c: Function result is '(null)'
Jan  1 23:08:20 VERBOSE[4924] logger.c: 
<-- SIP read from 10.90.87.205:5060: 
ACK sip:113@10.90.87.122 SIP/2.0
Via: SIP/2.0/UDP 10.90.87.205:5060;branch=z9hG4bKb36597f434
From: "Laboratorio" <sip:lab@10.90.87.122>;tag=31407a8e
To: <sip:113@10.90.87.122>;tag=as39c5c910
Call-ID: 190fce105ca1894a713c88c83d383c8c@10.90.87.205
Contact: <sip:lab@10.90.87.205:5060>
CSeq: 802 ACK
Proxy-Authorization: Digest username="lab",realm="asterisk",nonce="652e7059",response="a5b898447944ed6b613374dbff129526",uri="sip:113@10.90.87.122",algorithm=MD5
Content-Length: 0


Jan  1 23:08:20 VERBOSE[4924] logger.c: --- (9 headers 0 lines) ---
Jan  1 23:08:20 DEBUG[4924] chan_sip.c: Stopping retransmission on '190fce105ca1894a713c88c83d383c8c@10.90.87.205' of Response 802: Match Found
Jan  1 23:08:20 VERBOSE[4924] logger.c: set_destination: Parsing <sip:lab@10.90.87.205:5060> for address/port to send to
Jan  1 23:08:20 VERBOSE[4924] logger.c: set_destination: set destination to 10.90.87.205, port 5060
Jan  1 23:08:20 VERBOSE[4924] logger.c: Reliably Transmitting (no NAT) to 10.90.87.205:5060:
CANCEL sip:113@10.90.87.122 SIP/2.0
Via: SIP/2.0/UDP 10.90.87.122:5060;branch=z9hG4bK715e211e;rport
From: <sip:113@10.90.87.122>
To: "Laboratorio" <sip:lab@10.90.87.122>;tag=31407a8e
Call-ID: 190fce105ca1894a713c88c83d383c8c@10.90.87.205
CSeq: 101 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Jan  1 23:08:20 VERBOSE[4924] logger.c: Scheduling destruction of call '190fce105ca1894a713c88c83d383c8c@10.90.87.205' in 32000 ms
Jan  1 23:08:21 VERBOSE[4924] logger.c: Retransmitting #1 (no NAT) to 10.90.87.205:5060:
CANCEL sip:113@10.90.87.122 SIP/2.0
Via: SIP/2.0/UDP 10.90.87.122:5060;branch=z9hG4bK715e211e;rport
From: <sip:113@10.90.87.122>
To: "Laboratorio" <sip:lab@10.90.87.122>;tag=31407a8e
Call-ID: 190fce105ca1894a713c88c83d383c8c@10.90.87.205
CSeq: 101 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Jan  1 23:08:22 VERBOSE[4924] logger.c: 
<-- SIP read from 10.90.87.204:5060: 


Jan  1 23:08:22 VERBOSE[4924] logger.c: --- (0 headers 0 lines) Nat keepalive ---
Jan  1 23:08:22 VERBOSE[4924] logger.c: Retransmitting #2 (no NAT) to 10.90.87.205:5060:
CANCEL sip:113@10.90.87.122 SIP/2.0
Via: SIP/2.0/UDP 10.90.87.122:5060;branch=z9hG4bK715e211e;rport
From: <sip:113@10.90.87.122>
To: "Laboratorio" <sip:lab@10.90.87.122>;tag=31407a8e
Call-ID: 190fce105ca1894a713c88c83d383c8c@10.90.87.205
CSeq: 101 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

        Comentario

        • #5
          SOLUCIONADO !!!!!!! bueno parece que ellos son lentos o algo porque le agregue:
          Código:
          exten => 111,1,Answer()
same => n,Wait(1) ; >>> sin esto no hubo forma que funcionara
same => n,Playback(mios/test1&mios/test2) ; >> gracias cristian por lo del &
          su PM los telefonitos esos pero bueno ahi queda por si alguien algun dia se cruza con ellos
          Editado por última vez por venenodcuba; https://asteriskmx.org/foros/member/1647-venenodcuba en 01-03-2013, 08:36 AM.

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