Hi,

I'm running Asterisk 1.8.11.1 @office.

The Broadvoice service work fine with all 1.6 version and early 1.8 behind a NAT but about 2 months ago stop working.

No made changes in the firewall NAT rules. Right now I'm @home via my Xlite softphone working fine without problems

Any suggestions or thoughts?

Alex Celi



This is the info


central*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status 488/488 181.64.96.122 D 11037 OK (182 ms) sip.broadvoice.com/305422 206.15.148.221 5060 OK (131 ms)


sip.conf
    externip=190.12.68.20
    localnet=192.168.20.0/255.255.255.0
    localnet=192.168.10.0/255.255.255.0
    nat=comedia

    pedantic=no
register => 3054221...@sip.broadvoice.com:XXXXXXXXXX:3054221...@sip.broadvoice.com

    [sip.broadvoice.com]
    type=friend
    host=sip.broadvoice.com
    fromdomain=sip.broadvoice.com
    fromuser=3054221494
    defaultuser=3054221494
    authname=3054221494
    secret=XXXXXXXXX
    context=entrantes
    dtmfmode=inband
    dtmf=inband
    nat=comedia
    directmedia=no
    qualify=yes
    callgroup=1
    pickupgroup=1
    disallow=all
    allow=ulaw
    allow=alaw



I turned on sip debug. This is what I received

181.64.96.122: Is my home IP
190.12.68.20 or central.cipher.pe: is office IP
206.15.148.221: Broadvoice Server


    <--- SIP read from UDP:181.64.96.122:11037 --->
    INVITE sip:90018006273...@central.cipher.pe SIP/2.0
Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-81993d517bc9b121-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:488@181.64.96.122:11037>
    To: "90018006273999"<sip:90018006273...@central.cipher.pe>
    From: "488"<sip:4...@central.cipher.pe>;tag=93cce179
    Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
    CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
    Content-Type: application/sdp
    User-Agent: X-Lite release 1014k stamp 56015
    Content-Length: 235

    v=0
    o=- 8 2 IN IP4 192.168.7.33
    s=CounterPath X-Lite 3.0
    c=IN IP4 192.168.7.33
    t=0 0
    m=audio 2424 RTP/AVP 0 8 3 101
    a=fmtp:101 0-15
    a=rtpmap:101 telephone-event/8000
    a=alt:1 1 : hC2wRjti 7Lt7EhaI 192.168.7.33 2424
    a=sendrecv
    <------------->
    --- (12 headers 10 lines) ---
    Sending to 181.64.96.122:11037 (NAT)
Using INVITE request as basis request - ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
    Found peer '488' for '488' from 181.64.96.122:11037

    <--- Reliably Transmitting (no NAT) to 181.64.96.122:11037 --->
    SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-81993d517bc9b121-1---d8754z-;received=181.64.96.122;rport=11037
    From: "488"<sip:4...@central.cipher.pe>;tag=93cce179
    To: "90018006273999"<sip:90018006273...@central.cipher.pe>;tag=as77d2f824
    Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
    CSeq: 1 INVITE
    Server: Asterisk PBX 1.8.11.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0a1fded4"
    Content-Length: 0


    <------------>
Scheduling destruction of SIP dialog 'ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.' in 11648 ms (Method: INVITE)

    <--- SIP read from UDP:181.64.96.122:11037 --->
    ACK sip:90018006273...@central.cipher.pe SIP/2.0
Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-81993d517bc9b121-1---d8754z-;rport
    To: "90018006273999"<sip:90018006273...@central.cipher.pe>;tag=as77d2f824
    From: "488"<sip:4...@central.cipher.pe>;tag=93cce179
    Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
    CSeq: 1 ACK
    Content-Length: 0

    <------------->
    --- (7 headers 0 lines) ---

    <--- SIP read from UDP:181.64.96.122:11037 --->
    INVITE sip:90018006273...@central.cipher.pe SIP/2.0
Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-a8ee0d381f58006a-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:488@181.64.96.122:11037>
    To: "90018006273999"<sip:90018006273...@central.cipher.pe>
    From: "488"<sip:4...@central.cipher.pe>;tag=93cce179
    Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
    CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
    Content-Type: application/sdp
    User-Agent: X-Lite release 1014k stamp 56015
Authorization: Digest username="488",realm="asterisk",nonce="0a1fded4",uri="sip:90018006273...@central.cipher.pe",response="597c1f9bfb78f897ec94139eba9bf061",algorithm=MD5
    Content-Length: 235

    v=0
    o=- 8 2 IN IP4 192.168.7.33
    s=CounterPath X-Lite 3.0
    c=IN IP4 192.168.7.33
    t=0 0
    m=audio 2424 RTP/AVP 0 8 3 101
    a=fmtp:101 0-15
    a=rtpmap:101 telephone-event/8000
    a=alt:1 1 : hC2wRjti 7Lt7EhaI 192.168.7.33 2424
    a=sendrecv
    <------------->
    --- (13 headers 10 lines) ---
    Sending to 181.64.96.122:11037 (no NAT)
Using INVITE request as basis request - ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
    Found peer '488' for '488' from 181.64.96.122:11037
      == Using SIP RTP CoS mark 5
    Found RTP audio format 0
    Found RTP audio format 8
    Found RTP audio format 3
    Found RTP audio format 101
    Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
    Peer audio RTP is at port 192.168.7.33:2424
    Looking for 90018006273999 in gerencia (domain central.cipher.pe)
    list_route: hop: <sip:488@181.64.96.122:11037>

    <--- Transmitting (no NAT) to 181.64.96.122:11037 --->
    SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-a8ee0d381f58006a-1---d8754z-;received=181.64.96.122;rport=11037
    From: "488"<sip:4...@central.cipher.pe>;tag=93cce179
    To: "90018006273999"<sip:90018006273...@central.cipher.pe>
    Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
    CSeq: 2 INVITE
    Server: Asterisk PBX 1.8.11.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Contact: <sip:90018006273999@192.168.10.180:5060>
    Content-Length: 0


    <------------>
-- Executing [90018006273999@gerencia:1] Dial("SIP/488-00000000", "SIP/18006273...@sip.broadvoice.com,,Tt") in new stack
      == Using SIP RTP CoS mark 5
    Audio is at 11220
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP
    Reliably Transmitting (no NAT) to 206.15.148.221:5060:
    INVITE sip:18006273...@sip.broadvoice.com SIP/2.0
    Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK47c45d00
    Max-Forwards: 70
    From: "Celi M Carbajal" <sip:3054221...@sip.broadvoice.com>;tag=as18a86be7
    To: <sip:18006273...@sip.broadvoice.com>
    Contact: <sip:3054221494@192.168.10.180:5060>
    Call-ID: 71e46a1e52ecd53c591f47f12589a...@sip.broadvoice.com
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 1.8.11.1
    Date: Fri, 04 May 2012 06:54:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 209

    v=0
    o=root 1056464358 1056464358 IN IP4 192.168.10.180
    s=Asterisk PBX 1.8.11.1
    c=IN IP4 192.168.10.180
    t=0 0
    m=audio 11220 RTP/AVP 0 8
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=sendrecv

    ---
        -- Called SIP/18006273...@sip.broadvoice.com
    Retransmitting #1 (no NAT) to 206.15.148.221:5060:
    INVITE sip:18006273...@sip.broadvoice.com SIP/2.0
    Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK47c45d00
    Max-Forwards: 70
    From: "Celi M Carbajal" <sip:3054221...@sip.broadvoice.com>;tag=as18a86be7
    To: <sip:18006273...@sip.broadvoice.com>
    Contact: <sip:3054221494@192.168.10.180:5060>
    Call-ID: 71e46a1e52ecd53c591f47f12589a...@sip.broadvoice.com
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 1.8.11.1
    Date: Fri, 04 May 2012 06:54:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 209

    v=0
    o=root 1056464358 1056464358 IN IP4 192.168.10.180
    s=Asterisk PBX 1.8.11.1
    c=IN IP4 192.168.10.180
    t=0 0
    m=audio 11220 RTP/AVP 0 8
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=sendrecv

    ---

    <--- SIP read from UDP:206.15.148.221:5060 --->
    SIP/2.0 100 Trying
    Call-ID: 71e46a1e52ecd53c591f47f12589a...@sip.broadvoice.com
    CSeq: 102 INVITE
    From: "Celi M Carbajal" <sip:3054221...@sip.broadvoice.com>;tag=as18a86be7
    To: <sip:18006273...@sip.broadvoice.com>
Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK47c45d00;received=190.12.68.20;rport=5060
    Content-Length: 0

    <------------->
    --- (7 headers 0 lines) ---

    <--- SIP read from UDP:206.15.148.221:5060 --->
    SIP/2.0 503 Service Unavailable
    Call-ID: 71e46a1e52ecd53c591f47f12589a...@sip.broadvoice.com
    CSeq: 102 INVITE
    From: "Celi M Carbajal" <sip:3054221...@sip.broadvoice.com>;tag=as18a86be7
    To: <sip:18006273...@sip.broadvoice.com>;tag=qrst
Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK47c45d00;received=190.12.68.20;rport=5060
    User-Agent: Asterisk PBX 1.8.11.1
    Content-Length: 171
    Content-Type: application/sdp

    v=0
    o=3232238260 1056464358 1056464358 IN IP4 192.168.10.180
    s=-
    c=IN IP4 192.168.10.180
    t=0 0
    m=audio 11220 RTP/AVP 0 8
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    <------------->
    --- (9 headers 8 lines) ---
-- Got SIP response 503 "Service Unavailable" back from 206.15.148.221:5060
    Transmitting (no NAT) to 206.15.148.221:5060:
    ACK sip:18006273...@sip.broadvoice.com SIP/2.0
    Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK47c45d00
    Max-Forwards: 70
    From: "Celi M Carbajal" <sip:3054221...@sip.broadvoice.com>;tag=as18a86be7
    To: <sip:18006273...@sip.broadvoice.com>;tag=qrst
    Contact: <sip:3054221494@192.168.10.180:5060>
    Call-ID: 71e46a1e52ecd53c591f47f12589a...@sip.broadvoice.com
    CSeq: 102 ACK
    User-Agent: Asterisk PBX 1.8.11.1
    Content-Length: 0


    ---
        -- SIP/sip.broadvoice.com-00000001 is circuit-busy
      == Everyone is busy/congested at this time (1:0/1/0)
-- Executing [90018006273999@gerencia:2] Congestion("SIP/488-00000000", "") in new stack

    <--- Reliably Transmitting (no NAT) to 181.64.96.122:11037 --->
    SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-a8ee0d381f58006a-1---d8754z-;received=181.64.96.122;rport=11037
    From: "488"<sip:4...@central.cipher.pe>;tag=93cce179
    To: "90018006273999"<sip:90018006273...@central.cipher.pe>;tag=as17386e93
    Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
    CSeq: 2 INVITE
    Server: Asterisk PBX 1.8.11.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    X-Asterisk-HangupCause: Circuit/channel congestion
    X-Asterisk-HangupCauseCode: 34
    Content-Length: 0


    <------------>
Really destroying SIP dialog '71e46a1e52ecd53c591f47f12589a...@sip.broadvoice.com' Method: INVITE == Spawn extension (gerencia, 90018006273999, 2) exited non-zero on 'SIP/488-00000000'

    <--- SIP read from UDP:181.64.96.122:11037 --->
    ACK sip:90018006273...@central.cipher.pe SIP/2.0
Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-a8ee0d381f58006a-1---d8754z-;rport
    To: "90018006273999"<sip:90018006273...@central.cipher.pe>;tag=as17386e93
    From: "488"<sip:4...@central.cipher.pe>;tag=93cce179
    Call-ID: ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.
    CSeq: 2 ACK
    Content-Length: 0

    <------------->
    --- (7 headers 0 lines) ---
Really destroying SIP dialog 'ZDk2MDVkY2RhMTE2YjRkMmVhMjliMTRiYWIwOTdiM2M.' Method: ACK

    <--- SIP read from UDP:206.15.148.221:5060 --->
    SIP/2.0 503 Service Unavailable
    Call-ID: 71e46a1e52ecd53c591f47f12589a...@sip.broadvoice.com
    CSeq: 102 INVITE
    From: "Celi M Carbajal" <sip:3054221...@sip.broadvoice.com>;tag=as18a86be7
    To: <sip:18006273...@sip.broadvoice.com>;tag=qrst
Via: SIP/2.0/UDP 192.168.10.180:5060;branch=z9hG4bK47c45d00;received=190.12.68.20;rport=5060
    User-Agent: Asterisk PBX 1.8.11.1
    Content-Length: 171
    Content-Type: application/sdp

    v=0
    o=3232238260 1056464358 1056464358 IN IP4 192.168.10.180
    s=-
    c=IN IP4 192.168.10.180
    t=0 0
    m=audio 11220 RTP/AVP 0 8
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    <------------->
    --- (9 headers 8 lines) ---




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