Incoming SIP busy

Discussion in 'General' started by Ceephus, Oct 6, 2009.

  1. Ceephus

    Joined:
    Oct 5, 2009
    Messages:
    3
    Likes Received:
    0
    Hello,

    Sorry my first post is a question.. but I've been having this problem for a while now. When I make a call to my incoming DID I can see the traffic coming in but the dialing phone goes to an immediate busy signal. My softphone will show the call and give me the option to accept but no matter how quickly I hit accept I never get a ring on the dialing phone. I can dial internal to internal. I haven't yet setup an outbound DID so I can't test outgoing traffic. Below is the post from asterisk -rvvvvvv:

    Code:
    Verbosity is at least 7
        -- Executing [4018304203@from-pstn:1] NoOp("SIP/64.154.41.100-b756e270", "Catch-All DID Match - Found 4018304203 - You probably want a DID for this.") in new stack
        -- Executing [4018304203@from-pstn:2] Goto("SIP/64.154.41.100-b756e270", "ext-did|s|1") in new stack
        -- Goto (ext-did,s,1)
        -- Executing [s@ext-did:1] Set("SIP/64.154.41.100-b756e270", "__FROM_DID=s") in new stack
        -- Executing [s@ext-did:2] Gosub("SIP/64.154.41.100-b756e270", "app-blacklist-check|s|1") in new stack
        -- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/64.154.41.100-b756e270", "") in new stack
        -- Executing [s@app-blacklist-check:2] GotoIf("SIP/64.154.41.100-b756e270", "0?blacklisted") in new stack
        -- Executing [s@app-blacklist-check:3] Return("SIP/64.154.41.100-b756e270", "") in new stack
        -- Executing [s@ext-did:3] ExecIf("SIP/64.154.41.100-b756e270", "1 |Set|CALLERID(name)=4074883462") in new stack
        -- Executing [s@ext-did:4] SetMusicOnHold("SIP/64.154.41.100-b756e270", "acc_1") in new stack
        -- Executing [s@ext-did:5] Set("SIP/64.154.41.100-b756e270", "__MOHCLASS=acc_1") in new stack
        -- Executing [s@ext-did:6] Set("SIP/64.154.41.100-b756e270", "FAX_RX=disabled") in new stack
        -- Executing [s@ext-did:7] Set("SIP/64.154.41.100-b756e270", "__CALLINGPRES_SV=allowed_not_screened") in new stack
        -- Executing [s@ext-did:8] SetCallerPres("SIP/64.154.41.100-b756e270", "allowed_not_screened") in new stack
        -- Executing [s@ext-did:9] Goto("SIP/64.154.41.100-b756e270", "ext-local|vmb201|1") in new stack
        -- Goto (ext-local,vmb201,1)
        -- Executing [vmb201@ext-local:1] Macro("SIP/64.154.41.100-b756e270", "vm|201|BUSY|") in new stack
        -- Executing [s@macro-vm:1] Macro("SIP/64.154.41.100-b756e270", "user-callerid|SKIPTTL") in new stack
        -- Executing [s@macro-user-callerid:1] Set("SIP/64.154.41.100-b756e270", "AMPUSER=4074883462") in new stack
        -- Executing [s@macro-user-callerid:2] GotoIf("SIP/64.154.41.100-b756e270", "0?report") in new stack
        -- Executing [s@macro-user-callerid:3] ExecIf("SIP/64.154.41.100-b756e270", "1|Set|REALCALLERIDNUM=4074883462") in new stack
        -- Executing [s@macro-user-callerid:4] Set("SIP/64.154.41.100-b756e270", "AMPUSER=") in new stack
        -- Executing [s@macro-user-callerid:5] Set("SIP/64.154.41.100-b756e270", "AMPUSERCIDNAME=") in new stack
        -- Executing [s@macro-user-callerid:6] GotoIf("SIP/64.154.41.100-b756e270", "1?report") in new stack
        -- Goto (macro-user-callerid,s,11)
        -- Executing [s@macro-user-callerid:11] GotoIf("SIP/64.154.41.100-b756e270", "1?continue") in new stack
        -- Goto (macro-user-callerid,s,20)
        -- Executing [s@macro-user-callerid:20] NoOp("SIP/64.154.41.100-b756e270", "Using CallerID "4074883462" <4074883462>") in new stack
        -- Executing [s@macro-vm:2] Set("SIP/64.154.41.100-b756e270", "VMGAIN=""") in new stack
        -- Executing [s@macro-vm:3] GotoIf("SIP/64.154.41.100-b756e270", "1?vmx|1") in new stack
        -- Goto (macro-vm,vmx,1)
        -- Executing [vmx@macro-vm:1] GotoIf("SIP/64.154.41.100-b756e270", "0?s-BUSY|1") in new stack
        -- Executing [vmx@macro-vm:2] Set("SIP/64.154.41.100-b756e270", "MODE=busy") in new stack
        -- Executing [vmx@macro-vm:3] GotoIf("SIP/64.154.41.100-b756e270", "1?notdirect") in new stack
        -- Goto (macro-vm,vmx,5)
        -- Executing [vmx@macro-vm:5] NoOp("SIP/64.154.41.100-b756e270", "Checking if ext 201 is enabled: ") in new stack
        -- Executing [vmx@macro-vm:6] GotoIf("SIP/64.154.41.100-b756e270", "1?s-BUSY|1") in new stack
        -- Goto (macro-vm,s-BUSY,1)
        -- Executing [s-BUSY@macro-vm:1] NoOp("SIP/64.154.41.100-b756e270", "BUSY voicemail") in new stack
        -- Executing [s-BUSY@macro-vm:2] Macro("SIP/64.154.41.100-b756e270", "get-vmcontext|201") in new stack
        -- Executing [s@macro-get-vmcontext:1] Set("SIP/64.154.41.100-b756e270", "VMCONTEXT=default") in new stack
        -- Executing [s@macro-get-vmcontext:2] GotoIf("SIP/64.154.41.100-b756e270", "0?200:300") in new stack
        -- Goto (macro-get-vmcontext,s,300)
        -- Executing [s@macro-get-vmcontext:300] NoOp("SIP/64.154.41.100-b756e270", "") in new stack
        -- Executing [s-BUSY@macro-vm:3] VoiceMail("SIP/64.154.41.100-b756e270", "201@default|b") in new stack
        -- <SIP/64.154.41.100-b756e270> Playing 'vm-theperson' (language 'en')
        -- <SIP/64.154.41.100-b756e270> Playing 'digits/2' (language 'en')
        -- <SIP/64.154.41.100-b756e270> Playing 'digits/0' (language 'en')
        -- <SIP/64.154.41.100-b756e270> Playing 'digits/1' (language 'en')
        -- <SIP/64.154.41.100-b756e270> Playing 'vm-isonphone' (language 'en')
        -- <SIP/64.154.41.100-b756e270> Playing 'vm-intro' (language 'en')
    
    Any help will be much appreciate!

    Thanks!
     
  2. jessie

    Joined:
    Sep 17, 2008
    Messages:
    124
    Likes Received:
    0
    Hi Ceephus,

    Is this a fresh install Elastix? If so, try this work around.

    1. In your Elastix PBX Configuration, go to General Settings and change the option "no" into "yes" in Allow Anonymous Inbound SIP Calls?.

    Make a test call first if that resolved your incoming call. If not,

    2. Go to Inbound Routes. Add Incoming Route But dont put anything on DID Number parameter. You may select any option in Set Destination, then click on Submit. Just don't forget to click on Apply Configuration Changes.

    Now make another test call.

    That should be working. If not, please post another debug result.



    Regards,

    Jessie
     
  3. Ceephus

    Joined:
    Oct 5, 2009
    Messages:
    3
    Likes Received:
    0
    Thanks for the response. Anonymous SIP was turned on. The inbound route is set to the destination extension and when I dial the inbound DID it will ring my phone but the dialing phone goes busy right away...
     
  4. jessie

    Joined:
    Sep 17, 2008
    Messages:
    124
    Likes Received:
    0
    It seems we have to check all posible aspects then.

    1. Does your Elastix runs behind the NAT?

    If not, allow in your router or firewall to forward port 5060-5061 and 10000-20000 into your Elastix server.

    2. Is your DID a SIP trunk or a PSTN trunk?

    If it is a SIP trunk, what is the codec used against your local extensions?

    3. Have you made some configurations under your sip_general_custom.conf?

    If not yet, put this parameters:

    bindport = 5060 ; Port to bind to (SIP is 5060)
    bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
    insecure=invite
    defaultexpiry=160
    tos=0x68
    srvlookup=no
    callevent=yes
    disallow=all
    allow=ulaw
    allow=alaw
    qualify=400

    externip=XXX.XXX.XXX.XXX --> your public IP
    localhost=xxx.xxx.xxx.xxx/255.255.255.0 --> your local IP
    nat=yes
    canreinvite=no


    Let me know if this part fixed your problem.


    Regards,


    Jessie
     
  5. Ceephus

    Joined:
    Oct 5, 2009
    Messages:
    3
    Likes Received:
    0
    It is NAT'd with ports 10000-20000, 5050-5080 & 4569 open.
    The NAT information is entered in sip_nat.conf with my public and private IP information.

    It is a SIP trunk. I'm not sure what the codec is? Where would I find that?

    I did not make any changes under sip_general_custom.conf.

    If it helps this is for an IPCOMMS incoming SIP trunk. I'm hoping to get this incoming trunk going then after a few weeks of testing paying for a trunk.
     
  6. dwells

    Joined:
    Sep 29, 2009
    Messages:
    127
    Likes Received:
    0
    I didn't read any mentions of changing context. I have experienced some problems with wrong contexts. have you tried "context=from-trunk" or "from-internal" in your trunk settings to you provider?

    and only in the most extreme cases should you ever have annonSIP enabled. I have NEVER used it for any troubleshooting, it's usually something else. By enabling that you just gave the world the keys to your * castle!!!

    any help?
    -dwellsy
     

Share This Page