inbound: one-way audio; outbound:two-way audio

Discussion in 'General' started by benign, May 1, 2010.

  1. benign

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    What would cause inbound calls to not be able to hear the caller, but outbound calls have two-way comm?

    Setup is like this:

    Phone - > Elastix -> Pfsense -> internet

    I have the same setup at another location, and everything works fine both ways. Same provider, same phones. Same phone config files, etc.. etc..

    I have every single port opened and going to elastix from the external interface (for testing). This doesn't look like a NAT issue as there are no remote phones. When we put remote extensions to call into our first site, I had to configure everything for NAT to get it working. Before that, there was no nat=yes or qualify=no anywhere.

    It seems to not be allowing voice packets passed the firewall, or to elastix, or to the phone if any only if the call is inbound. Outbound calls function 100% correctly.
     
  2. dicko

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    That is ALWAYS a nat/router/firewall misconfiguration in the router or the Elastix box, it can't be anything else, any inbound udp connections on ports 10000-20000 must be forwarded to the Asterisk box, the asterisk box must nat translate to the external IP if necessary. Check the documentation again, PFSense works perfectly apart from tftp traversal (yet)
     
  3. benign

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    Here are details of my setup. Do you see anything out of place?

    Telephone -> Elastix -> Firewall -> Internet

    Firewall: NAT: Port forward (I've also tried this with *)

    If ------- Proto -------- Ext. Port ------ NAT IP ------- Int. Port
    WAN TCP/UDP 5004-5082 <elastix box> 5004-5082
    WAN TCP/UDP 10000-65535 <elastix box> 10000-65535
    WAN UDP 4569 <elastix box> 4569

    Firewall: Rules (I've tried this with specific source ports)

    Proto --- Source -- Port -- Destination ----- Port ------ GW -- Sch.
    TCP/UDP * * <elastix box> 5004-5082 * *
    TCP/UDP * * <elastix box> 10000-65535 * *
    UDP * * <elastix box> 4569 * *

    Elastix Trunk Details:
    (These don't require any username/auth). They are set up the same way on another working box.
    ---------------------
    PEER:
    host=38.xxx.225.xxx
    type=peer
    disallow=all
    allow=ulaw
    dtmfmode=rfc2833

    USER:
    context=ext-did
    host=38.xxx.225.xxx
    insecure=very
    type=friend

    ---------------
    Extension: 1000

    secret: <secret>
    dtmfmode: auto
    canreinvite: no
    context: from-internal
    host: dynamic
    type: friend
    nat: no
    port: 5060
    qualify: yes
    deny: 0.0.0.0/0.0.0.0
    permit 0.0.0.0/0.0.0.0

    sip_nat.conf (not using NAT)

    externip=72.xx.202.xxx
    localnet=10.20.0.0/255.255.255.0

    elastix*CLI> sip show peer 1000
    elastix*CLI>

    * Name : 1000
    Secret : <Set>
    MD5Secret : <Not set>
    Context : from-internal
    Subscr.Cont. : <Not set>
    Language :
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Mailbox : 1000@default
    VM Extension : *97
    LastMsgsSent : 0/0
    Call limit : 50
    Dynamic : Yes
    Callerid : "device" <1000>
    MaxCallBR : 384 kbps
    Expire : 2035
    Insecure : no
    Nat : RFC3581
    ACL : Yes
    T38 pt UDPTL : No
    CanReinvite : No
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Trust RPID : No
    Send RPID : No
    Subscriptions: Yes
    Overlap dial : Yes
    DTMFmode : auto
    LastMsg : 0
    ToHost :
    Addr->IP : <phone IP> Port 5060
    Defaddr->IP : 0.0.0.0 Port 5060
    Def. Username: 1000
    SIP Options : (none)
    Codecs : 0xc (ulaw|alaw)
    Codec Order : (ulaw:20,alaw:20)
    Auto-Framing: No
    Status : OK (88 ms)
    Useragent : Cisco-CP7960G/8.0
    Reg. Contact : sip:1000@<phone IP>:5060;transport=udp


    elastix*CLI> sip show settings
    elastix*CLI>

    Global Settings:
    ----------------
    SIP Port: 5060
    Bindaddress: 0.0.0.0
    Videosupport: No
    AutoCreatePeer: No
    Allow unknown access: Yes
    Allow subscriptions: Yes
    Allow overlap dialing: Yes
    Promsic. redir: No
    SIP domain support: No
    Call to non-local dom.: Yes
    URI user is phone no: No
    Our auth realm asterisk
    Realm. auth: No
    Always auth rejects: Yes
    Call limit peers only: Yes
    Direct RTP setup: No
    User Agent: Asterisk PBX
    MWI checking interval: 10 secs
    Reg. context: (not set)
    Caller ID: Unknown
    From: Domain:
    Record SIP history: Off
    Call Events: Off
    IP ToS SIP: CS3
    IP ToS RTP audio: EF
    IP ToS RTP video: AF41
    T38 fax pt UDPTL: No
    RFC2833 Compensation: No
    SIP realtime: Disabled

    Global Signalling Settings:
    ---------------------------
    Codecs: 0xc (ulaw|alaw)
    Codec Order: ulaw:20,alaw:20
    T1 minimum: 100
    Relax DTMF: No
    Compact SIP headers: No
    RTP Keepalive: 0 (Disabled)
    RTP Timeout: 0 (Disabled)
    RTP Hold Timeout: 0 (Disabled)
    MWI NOTIFY mime type: application/simple-message-summary
    DNS SRV lookup: Yes
    Pedantic SIP support: No
    Reg. min duration 60 secs
    Reg. max duration: 3600 secs
    Reg. default duration: 120 secs
    Outbound reg. timeout: 20 secs
    Outbound reg. attempts: 0
    Notify ringing state: Yes
    Notify hold state: Yes
    SIP Transfer mode: open
    Max Call Bitrate: 384 kbps
    Auto-Framing: No
    elastix*CLI>
    Default Settings:
    -----------------
    Context: from-sip-external
    Nat: RFC3581
    DTMF: rfc2833
    Qualify: 0
    Use ClientCode: No
    Progress inband: Never
    Language: (Defaults to English)
    MOH Interpret: default
    MOH Suggest:
    Voice Mail Extension: *97
     
  4. dicko

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    I think if your localnet is 10.20.0.0/24 and your externip is static you probably are NATTED and SHOULD be using it in sip*.conf

    check with

    sip set debug ip 72.xx.202.xxx

    to make sure the headers are rewritten properly.
     
  5. benign

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    I did a sip debug on a peer and called it. This is what their elastix box CLI gives from the time I call, until I hang up. I can hear them, they can not hear me.

    I don't see anything for the external address here, just the address of the phone (.53), and the address of the elastix box (.29)




    I did a sip debug on a peer and called it. This is what their elastix box CLI gives from the time I call, until I hang up. I can hear them, they can not hear me.

    I don't see anything for the external address here, just the address of the phone (.53), and the address of the elastix box (.29)




    -- Executing [81xxxx6086@ext-did:1] Set("SIP/from-321-170-0926c3c8", "__FROM_DID=81xxxx6086") in new stack
    -- Executing [81xxxx6086@ext-did:2] Gosub("SIP/from-321-170-0926c3c8", "app-blacklist-check|s|1") in new stack
    -- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/from-321-170-0926c3c8", "") in new stack
    -- Executing [s@app-blacklist-check:2] GotoIf("SIP/from-321-170-0926c3c8", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:3] Return("SIP/from-321-170-0926c3c8", "") in new stack
    -- Executing [81xxxx6086@ext-did:3] ExecIf("SIP/from-321-170-0926c3c8", "1 |Set|CALLERID(name)=+1xxx373xx52") in new stack
    -- Executing [81xxxx6086@ext-did:4] Set("SIP/from-321-170-0926c3c8", "FAX_RX=") in new stack
    -- Executing [81xxxx6086@ext-did:5] Set("SIP/from-321-170-0926c3c8", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [81xxxx6086@ext-did:6] SetCallerPres("SIP/from-321-170-0926c3c8", "allowed_not_screened") in new stack
    -- Executing [81xxxx6086@ext-did:7] Goto("SIP/from-321-170-0926c3c8", "from-did-direct|1001|1") in new stack
    -- Goto (from-did-direct,1001,1)
    -- Executing [1001@from-did-direct:1] Macro("SIP/from-321-170-0926c3c8", "exten-vm|1001|1001") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/from-321-170-0926c3c8", "user-callerid") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/from-321-170-0926c3c8", "AMPUSER=+1xxx373xx52") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/from-321-170-0926c3c8", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/from-321-170-0926c3c8", "1|Set|REALCALLERIDNUM=+1xxx373xx52") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/from-321-170-0926c3c8", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/from-321-170-0926c3c8", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/from-321-170-0926c3c8", "1?report") in new stack
    -- Goto (macro-user-callerid,s,10)
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/from-321-170-0926c3c8", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/from-321-170-0926c3c8", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/from-321-170-0926c3c8", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/from-321-170-0926c3c8", "Using CallerID "+1xxx373xx52" <+1xxx373xx52>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/from-321-170-0926c3c8", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/from-321-170-0926c3c8", "VMBOX=1001") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/from-321-170-0926c3c8", "EXTTOCALL=1001") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/from-321-170-0926c3c8", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("SIP/from-321-170-0926c3c8", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("SIP/from-321-170-0926c3c8", "RT=15") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("SIP/from-321-170-0926c3c8", "record-enable|1001|IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/from-321-170-0926c3c8", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/from-321-170-0926c3c8", "recordingcheck|20100506-112119|1273159279.923") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
    recordingcheck|20100506-112119|1273159279.923: Inbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/from-321-170-0926c3c8", "") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("SIP/from-321-170-0926c3c8", "dial|15|tTrwW|1001") in new stack
    -- Executing [s@macro-dial:1] GotoIf("SIP/from-321-170-0926c3c8", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("SIP/from-321-170-0926c3c8", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
    dialparties.agi: Starting New Dialparties.agi
    == Parsing '/etc/asterisk/manager.conf': Found
    == Parsing '/etc/asterisk/manager_additional.conf': Found
    == Parsing '/etc/asterisk/manager_custom.conf': Found
    == Manager 'admin' logged on from 127.0.0.1
    dialparties.agi: Caller ID name is '+1xxx373xx52' number is '+1xxx373xx52'
    dialparties.agi: USE_CONFIRMATION: 'FALSE'
    dialparties.agi: RINGGROUP_INDEX: ''
    dialparties.agi: Methodology of ring is 'none'
    -- dialparties.agi: Added extension 1001 to extension map
    > dialparties.agi: Extension 1001 has call screening off
    -- dialparties.agi: Extension 1001 cf is disabled
    -- dialparties.agi: Extension 1001 do not disturb is disabled
    > dialparties.agi: extnum 1001 has: cw: 0; hascfb: 0 [] hascfu: 0 []
    dialparties.agi: ExtensionState: 0
    dialparties.agi: Extension 1001 has ExtensionState: 0
    -- dialparties.agi: Checking CW and CFB status for extension 1001
    -- dialparties.agi: dbset CALLTRACE/1001 to +1xxx373xx52
    -- dialparties.agi: Filtered ARG3: 1001
    == Manager 'admin' logged off from 127.0.0.1
    -- AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:7] Dial("SIP/from-321-170-0926c3c8", "SIP/1001|15|tTrwW") in new stack
    Audio is at 10.xx.x.29 port 13516
    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 10.xx.x.53:5060:
    INVITE sip:1001@10.xx.x.53:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.xx.x.29:5060;branch=z9hG4bK199ee84f;rport
    From: "+1xxx373xx52" <sip:+1xxx373xx52@10.xx.x.29>;tag=as2c40a970
    To: <sip:1001@10.xx.x.53:5060;transport=udp>
    Contact: <sip:+1xxx373xx52@10.xx.x.29>
    Call-ID: 42bf4e872bb883d9628cff865315cb8a@10.xx.x.29
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Date: Thu, 06 May 2010 15:21:20 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: 258

    v=0
    o=root 3066 3066 IN IP4 10.xx.x.29
    s=session
    c=IN IP4 10.xx.x.29
    t=0 0
    m=audio 13516 RTP/AVP 0 8 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:eek:ff - - - -
    a=ptime:20
    a=sendrecv

    ---
    -- Called 1001
    -- SIP/1001-092f3cd0 is ringing
    Found RTP audio format 0
    Peer audio RTP is at port 10.xx.x.53:25046
    Found audio description format PCMU for ID 0
    Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
    Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
    Peer audio RTP is at port 10.xx.x.53:25046
    list_route: hop: <sip:1001@10.xx.x.53:5060;transport=udp>
    set_destination: Parsing <sip:1001@10.xx.x.53:5060;transport=udp> for address/port to send to
    set_destination: set destination to 10.xx.x.53, port 5060
    Transmitting (no NAT) to 10.xx.x.53:5060:
    ACK sip:1001@10.xx.x.53:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.xx.x.29:5060;branch=z9hG4bK5e476eba;rport
    From: "+1xxx373xx52" <sip:+1xxx373xx52@10.xx.x.29>;tag=as2c40a970
    To: <sip:1001@10.xx.x.53:5060;transport=udp>;tag=000f8f7608ac048a21c17a9d-33e5dd6f
    Contact: <sip:+1xxx373xx52@10.xx.x.29>
    Call-ID: 42bf4e872bb883d9628cff865315cb8a@10.xx.x.29
    CSeq: 102 ACK
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Content-Length: 0


    ---
    -- SIP/1001-092f3cd0 answered SIP/from-321-170-0926c3c8
    -- Executing [h@macro-dial:1] Macro("SIP/from-321-170-0926c3c8", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/from-321-170-0926c3c8", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/from-321-170-0926c3c8", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/from-321-170-0926c3c8", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/from-321-170-0926c3c8", "") in new stack
    == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/from-321-170-0926c3c8' in macro 'hangupcall'
    == Spawn h extension (macro-dial, h, 1) exited non-zero on 'SIP/from-321-170-0926c3c8'
    Scheduling destruction of SIP dialog '42bf4e872bb883d9628cff865315cb8a@10.xx.x.29' in 6400 ms (Method: INVITE)
    set_destination: Parsing <sip:1001@10.xx.x.53:5060;transport=udp> for address/port to send to
    set_destination: set destination to 10.xx.x.53, port 5060
    Reliably Transmitting (no NAT) to 10.xx.x.53:5060:
    BYE sip:1001@10.xx.x.53:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.xx.x.29:5060;branch=z9hG4bK1f8edd1a;rport
    From: "+1xxx373xx52" <sip:+1xxx373xx52@10.xx.x.29>;tag=as2c40a970
    To: <sip:1001@10.xx.x.53:5060;transport=udp>;tag=000f8f7608ac048a21c17a9d-33e5dd6f
    Call-ID: 42bf4e872bb883d9628cff865315cb8a@10.xx.x.29
    CSeq: 103 BYE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    X-Asterisk-HangupCause: Normal Clearing
    X-Asterisk-HangupCauseCode: 16
    Content-Length: 0


    ---
    == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/from-321-170-0926c3c8' in macro 'dial'
    == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/from-321-170-0926c3c8' in macro 'exten-vm'
    == Spawn extension (from-did-direct, 1001, 1) exited non-zero on 'SIP/from-321-170-0926c3c8'
    Really destroying SIP dialog '42bf4e872bb883d9628cff865315cb8a@10.xx.x.29' Method: INVITE
     
  6. dicko

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    Did you try either in sip*.conf

    nat=1

    or
    nat=route

    ?
     
  7. benign

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    I've tried nat=yes, it show in the sip debug that it is transmitting with nat after a sip reload. I just tried nat = route. Sip details show nat: always instead of rfc3581. With nat=route or nat=yes, there is no audio both ways :(
     
  8. dicko

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    try the nat=route

    also

    rtp debug ip <voip provider>

    To watch the rtp traffic in asterisk (It will presumably be one way) and at the same time

    tcpdump -nn host <voip provider>

    from bash to see where the missing rtp packts are being sent under the various nat=? settings
     
  9. dicko

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    I have no problems with pfsense, but use 1:1 you might want to set your NAT outbound route to manual and use static port mapping
     
  10. benign

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    [root@elastix ~]# tcpdump -nn host 38.xxx.xxx.170
    tcpdump: verbose output suppressed, use -v or -vv for full protocol decode
    listening on eth0, link-type EN10MB (Ethernet), capture size 96 bytes
    13:55:27.218598 IP 38.xxx.xxx.170.5060 > 10.xx.x.29.5060: SIP, length: 1109
    13:55:27.219325 IP 10.xx.x.29.5060 > 38.xxx.xxx.170.5060: SIP, length: 699
    13:55:27.432059 IP 10.xx.x.29.5060 > 38.xxx.xxx.170.5060: SIP, length: 715
    13:55:30.796670 IP 10.xx.x.29.5060 > 38.xxx.xxx.170.5060: SIP, length: 1005
    13:55:30.977776 IP 38.xxx.xxx.170.5060 > 10.xx.x.29.5060: SIP, length: 680
    13:55:38.587758 IP 38.xxx.xxx.170.5060 > 10.xx.x.29.5060: SIP, length: 672
    13:55:38.587988 IP 10.xx.x.29.5060 > 38.xxx.xxx.170.5060: SIP, length: 564

    This is weird because the trunk is actually set up to be 38.xxx.xxx.190. It's not showing
    anything going out to .190 so I assume whatever is sent to 190 just comes back through 170.

    I testing changing the incoming peer settings to 170, which had no effect.

    RTP Debug and SIP headers information for a phone call to the server is this:







    INVITE sip:1001@10.xx.x.53:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.xx.x.29:5060;branch=z9hG4bK3248a0d8;rport
    From: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xx.x.29>;tag=as35702444
    To: <sip:1001@10.xx.x.53:5060;transport=udp>
    Contact: <sip:+1xxxxxxxxxx@10.xx.x.29>
    Call-ID: 4368d3db6f38a5d20ffbcec65f2abf16@10.xx.x.29
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Date: Thu, 06 May 2010 18:04:49 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: 258

    v=0
    o=root 3066 3066 IN IP4 10.xx.x.29
    s=session
    c=IN IP4 10.xx.x.29
    t=0 0
    m=audio 13722 RTP/AVP 0 8 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:eek:ff - - - -
    a=ptime:20
    a=sendrecv

    ---
    -- Called 1001
    -- SIP/1001-092b3200 is ringing
    Found RTP audio format 0
    Peer audio RTP is at port 10.xx.x.53:17922
    Found audio description format PCMU for ID 0
    Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
    Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
    Peer audio RTP is at port 10.xx.x.53:17922
    list_route: hop: <sip:1001@10.xx.x.53:5060;transport=udp>
    set_destination: Parsing <sip:1001@10.xx.x.53:5060;transport=udp> for address/port to send to
    set_destination: set destination to 10.xx.x.53, port 5060
    Transmitting (no NAT) to 10.xx.x.53:5060:
    ACK sip:1001@10.xx.x.53:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.xx.x.29:5060;branch=z9hG4bK4f3d44ad;rport
    From: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xx.x.29>;tag=as35702444
    To: <sip:1001@10.xx.x.53:5060;transport=udp>;tag=000f8f7608ac05476eb02cc5-1c7876ab
    Contact: <sip:+1xxxxxxxxxx@10.xx.x.29>
    Call-ID: 4368d3db6f38a5d20ffbcec65f2abf16@10.xx.x.29
    CSeq: 102 ACK
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Content-Length: 0

    ---
    -- SIP/1001-092b3200 answered SIP/from-321-b6a4ee38
    Got RTP packet from 10.xx.x.53:17922 (type 00, seq 003330, ts 533312, len 000160)
    Sent RTP packet to 199.173.105.141:31934 (type 00, seq 021818, ts 533312, len 000160)
    Got RTP packet from 10.xx.x.53:17922 (type 00, seq 003331, ts 533472, len 000160)
    Sent RTP packet to 199.173.105.141:31934 (type 00, seq 021819, ts 533472, len 000160)

    199.173.105.141 is some verizon equipment.. which is the ISP the site I'm working with is using.
     
  11. dicko

    Joined:
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    I would try something else but pfsense at this time, if it works go to the pfsense forum with this problem.

    sorry I can't be of more help

    dicko
     
  12. benign

    Joined:
    Mar 10, 2010
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    Ok, well I appreciate the help.
     

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