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Se cortan las llamadas en 6 segundos

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  • #1

    Se cortan las llamadas en 6 segundos

    Hola, tengo un problema con mi PBX Asterisk, tengo extenciones en la red local, otras conectadas por VPN y otras conectadas por tls, con las de la red local y las de VPN no tengo ningun problema pero cuando llamo de extension TLS a extension TLS se corta la llamada en aproximadamente 6 segundos, es curioso porque si marco de una extension de red local o VPN a una TLS entra la comunicacion se establece sin problemas, el problema es solo de TLS a TLS.

    <--- SIP read from TLS:XXX.XXX.XXX.17:49775 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/TLS XXX.XXX.XXX.193:5061;branch=z9hG4bK7f96aa54;rport= 46598
    Contact: <sip:ext9110@XXX.XXX.XXX.17:49775;transport=TLS>
    To: <sip:ext9110@XXX.XXX.XXX.17:49775;transport=TLS;ri nstance=b9de2685fde6800d>;tag=128f2c42
    From: "Usuario1" <sip:110@XXX.XXX.XXX.193>;tag=as39f4055c
    Call-ID: 0f098a901477071830aeab0a28b68c12@XXX.XXX.XXX.193:5061
    CSeq: 102 INVITE
    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
    Content-Type: application/sdp
    User-Agent: Zoiper rv2.8.83-mod
    Allow-Events: presence, kpml, talk
    Content-Length: 324

    v=0
    o=Zoiper 0 1 IN IP4 192.168.0.7
    s=Zoiper
    c=IN IP4 192.168.0.7
    t=0 0
    m=audio 43800 RTP/SAVP 0 3 8 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=sendrecv
    a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:EWdqLv6Hn5l/BpN3dwOJh/m12ydnmw/VWlTbRbY0
    <------------->
    --- (12 headers 13 lines) ---
    Found RTP audio format 0
    Found RTP audio format 3
    Found RTP audio format 8
    Found RTP audio format 101
    Found audio description format PCMU for ID 0
    Found audio description format GSM for ID 3
    Found audio description format PCMA for ID 8
    Found audio description format telephone-event for ID 101
    Capabilities: us - (g729|ulaw|alaw|gsm), peer - audio=(ulaw|gsm|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
    Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
    Peer audio RTP is at port 192.168.0.7:43800
    sip_route_dump: route/path hop: <sip:ext9110@XXX.XXX.XXX.17:49775;transport=TLS>
    Transmitting (NAT) to XXX.XXX.XXX.17:49775:
    ACK sip:ext9110@XXX.XXX.XXX.17:49775;transport=TLS SIP/2.0
    Via: SIP/2.0/TLS XXX.XXX.XXX.193:5061;branch=z9hG4bK002daf1c;rport
    Max-Forwards: 70
    From: "Usuario1" <sip:110@XXX.XXX.XXX.193>;tag=as39f4055c
    To: <sip:ext9110@XXX.XXX.XXX.17:49775;transport=TLS;ri nstance=b9de2685fde6800d>;tag=128f2c42
    Contact: <sip:110@XXX.XXX.XXX.193:5061;transport=tls>
    Call-ID: 0f098a901477071830aeab0a28b68c12@XXX.XXX.XXX.193:5061
    CSeq: 102 ACK
    User-Agent: Asterisk PBX 13.18.5
    Content-Length: 0


    ---
    -- SIP/ext9110-0000000b answered SIP/ext110-0000000a
    Audio is at 47556
    Adding codec g729 to SDP
    Adding non-codec 0x1 (telephone-event) to SDP

    <--- Reliably Transmitting (NAT) to XXX.XXX.XXX.17:3648 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/TLS 192.168.0.3:63999;branch=z9hG4bKPjeb9ca4b9500b4093 b4867e309c519968;alias;received=XXX.XXX.XXX.17;rpo rt=3648
    From: "Usuario1" <sip:ext110@sip.XXXXX.XXX>;tag=0aa4fcfc29b640ca9c0 aba0df28492b9
    To: <sip:9110@sip.XXXXX.XXX>;tag=as3be61f9a
    Call-ID: 98a235fcb7bd4cacbf3d674bf3c72155
    CSeq: 1252 INVITE
    Server: Asterisk PBX 13.18.5
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Session-Expires: 1800;refresher=uas
    Contact: <sip:9110@XXX.XXX.XXX.193:5061;transport=tls>
    Content-Type: application/sdp
    Require: timer
    Content-Length: 360

    v=0
    o=root 899467647 899467647 IN IP4 XXX.XXX.XXX.193
    s=Asterisk PBX 13.18.5
    c=IN IP4 XXX.XXX.XXX.193
    t=0 0
    m=audio 47556 RTP/SAVP 18 101
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=maxptime:230
    a=sendrecv
    a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:Bq++mkq2GvKfe+Da9mDam8gjQwM6kjLxMMUmGoXG

    <------------>
    -- Channel SIP/ext9110-0000000b joined 'simple_bridge' basic-bridge <3ce4a593-3c92-4377-98df-7075f25549c3>
    -- Channel SIP/ext110-0000000a joined 'simple_bridge' basic-bridge <3ce4a593-3c92-4377-98df-7075f25549c3>

    <--- SIP read from TLS:XXX.XXX.XXX.17:3648 --->

    <------------->
    Reliably Transmitting (NAT) to XXX.XXX.XXX.17:3648:
    OPTIONS sip:ext110@192.168.0.3:60335;transport=TLS;ob SIP/2.0
    Via: SIP/2.0/TLS XXX.XXX.XXX.193:5061;branch=z9hG4bK0b1cdbd5;rport
    Max-Forwards: 70
    From: "asterisk" <sip:asterisk@XXX.XXX.XXX.193>;tag=as03844661
    To: <sip:ext110@192.168.0.3:60335;transport=TLS;ob>
    Contact: <sip:asterisk@XXX.XXX.XXX.193:5061;transport=tls >
    Call-ID: 43233c614d0ddf072e55b67325b6307b@XXX.XXX.XXX.193:5061
    CSeq: 102 OPTIONS
    User-Agent: Asterisk PBX 13.18.5
    Date: Thu, 15 Feb 2018 10:18:55 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Content-Length: 0


    ---

    <--- SIP read from TLS:XXX.XXX.XXX.17:3648 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/TLS XXX.XXX.XXX.193:5061;rport=46598;received=XXX.XXX. XXX.193;branch=z9hG4bK0b1cdbd5
    Call-ID: 43233c614d0ddf072e55b67325b6307b@XXX.XXX.XXX.193:5061
    From: "asterisk" <sip:asterisk@XXX.XXX.XXX.193>;tag=as03844661
    To: <sip:ext110@192.168.0.3;ob>;tag=z9hG4bK0b1cdbd5
    CSeq: 102 OPTIONS
    Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
    Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
    Supported: replaces, 100rel, timer, norefersub
    Allow-Events: presence, message-summary, refer
    User-Agent: MicroSIP/3.17.3
    Content-Length: 0

    <------------->
    --- (12 headers 0 lines) ---
    Really destroying SIP dialog '43233c614d0ddf072e55b67325b6307b@XXX.XXX.XXX.193: 5061' Method: OPTIONS
    [Feb 15 04:18:58] WARNING[8918]: chan_sip.c:4072 retrans_pkt: Retransmission timeout reached on transmission 98a235fcb7bd4cacbf3d674bf3c72155 for seqno 1252 (Critical Response) -- See https://wiki.asterisk.org/wiki/displ...etransmissions
    Packet timed out after 6976ms with no response
    [Feb 15 04:18:58] WARNING[8918]: chan_sip.c:4096 retrans_pkt: Hanging up call 98a235fcb7bd4cacbf3d674bf3c72155 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/displ...etransmissions).
    -- Channel SIP/ext110-0000000a left 'simple_bridge' basic-bridge <3ce4a593-3c92-4377-98df-7075f25549c3>
    == Spawn extension (contextoLADA, 9110, 2) exited non-zero on 'SIP/ext110-0000000a'
    Scheduling destruction of SIP dialog '98a235fcb7bd4cacbf3d674bf3c72155' in 6976 ms (Method: INVITE)
    Reliably Transmitting (NAT) to XXX.XXX.XXX.17:3648:
    BYE sip:ext110@192.168.0.3:60335;transport=TLS;ob SIP/2.0
    Via: SIP/2.0/TLS XXX.XXX.XXX.193:5061;branch=z9hG4bK4758b791;rport
    Max-Forwards: 70
    From: <sip:9110@sip.XXXXX.XXX>;tag=as3be61f9a
    To: "Usuario1" <sip:ext110@sip.XXXXX.XXX>;tag=0aa4fcfc29b640ca9c0 aba0df28492b9
    Call-ID: 98a235fcb7bd4cacbf3d674bf3c72155
    CSeq: 102 BYE
    User-Agent: Asterisk PBX 13.18.5
    Proxy-Authorization: Digest username="ext110", realm="asterisk", algorithm=MD5, uri="sips:sip.XXXXX.XXX", nonce="0479c43c", response="4ae226e5307bebd2e8139c2d14da579b"
    X-Asterisk-HangupCause: No user responding
    X-Asterisk-HangupCauseCode: 18
    Content-Length: 0


    ---
    -- Channel SIP/ext9110-0000000b left 'simple_bridge' basic-bridge <3ce4a593-3c92-4377-98df-7075f25549c3>
    Scheduling destruction of SIP dialog '0f098a901477071830aeab0a28b68c12@XXX.XXX.XXX.193: 5061' in 6720 ms (Method: INVITE)
    Reliably Transmitting (NAT) to XXX.XXX.XXX.17:49775:
    BYE sip:ext9110@XXX.XXX.XXX.17:49775;transport=TLS SIP/2.0
    Via: SIP/2.0/TLS XXX.XXX.XXX.193:5061;branch=z9hG4bK04cadd8d;rport
    Max-Forwards: 70
    From: "Usuario1" <sip:110@XXX.XXX.XXX.193>;tag=as39f4055c
    To: <sip:ext9110@XXX.XXX.XXX.17:49775;transport=TLS;ri nstance=b9de2685fde6800d>;tag=128f2c42
    Call-ID: 0f098a901477071830aeab0a28b68c12@XXX.XXX.XXX.193:5061
    CSeq: 103 BYE
    User-Agent: Asterisk PBX 13.18.5
    X-Asterisk-HangupCause: No user responding
    X-Asterisk-HangupCauseCode: 18
    Content-Length: 0


    ---

    <--- SIP read from TLS:XXX.XXX.XXX.17:3648 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/TLS XXX.XXX.XXX.193:5061;rport=46598;received=XXX.XXX. XXX.193;branch=z9hG4bK4758b791
    Call-ID: 98a235fcb7bd4cacbf3d674bf3c72155
    From: <sip:9110@sip.XXXXX.XXX>;tag=as3be61f9a
    To: "Usuario1" <sip:ext110@sip.XXXXX.XXX>;tag=0aa4fcfc29b640ca9c0 aba0df28492b9
    CSeq: 102 BYE
    Content-Length: 0

    <------------->
    --- (7 headers 0 lines) ---
    SIP Response message for INCOMING dialog BYE arrived

    <--- SIP read from TLS:XXX.XXX.XXX.17:49775 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/TLS XXX.XXX.XXX.193:5061;branch=z9hG4bK04cadd8d;rport= 46598
    Contact: <sip:ext9110@XXX.XXX.XXX.17:49775;transport=TLS>
    To: <sip:ext9110@XXX.XXX.XXX.17:49775;transport=TLS;ri nstance=b9de2685fde6800d>;tag=128f2c42
    From: "Usuario1" <sip:110@XXX.XXX.XXX.193>;tag=as39f4055c
    Call-ID: 0f098a901477071830aeab0a28b68c12@XXX.XXX.XXX.193:5061
    CSeq: 103 BYE
    User-Agent: Zoiper rv2.8.83-mod
    Content-Length: 0

    <------------->
    --- (9 headers 0 lines) ---
    Really destroying SIP dialog '98a235fcb7bd4cacbf3d674bf3c72155' Method: INVITE
    Really destroying SIP dialog '0f098a901477071830aeab0a28b68c12@XXX.XXX.XXX.193: 5061' Method: INVITE
    Etiquetas: Ninguno
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