No imbound calls "the number you have dialed has b

Discussion in 'General' started by slamman, Dec 3, 2008.

  1. slamman

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    I need help.

    I have been fiddling with this for a couple of weeks and I am getting nowhere. I have both broadvoice and FWD set up and on both trunks (sip and iax2) I can make calls to the outside world. I can call between extensions just fine. However I do not get any inbound calls. I have gone through every post I can find on the subject both here and on the web at large and can't find a solution. I don't think is is necessarily an elastix or asterisk setup issue. It is most likely a networking problem but I though someone here may have more knowledge in tracking down the problem than I.

    I do not see any evidence that the call is hitting my network, although Broadvoice says it is. Is there someway I can definitively tell where the packets are getting lost? I am not that familiar with packet sniffers or how to use them. I see nothing in the logs or on the cli to indicate the inbound call is hitting the elastix box.

    My network is behind an untangle firewall with ports 5060-5080 an 10,000-20,000 open and forwarded. I also have the IAX2 port opened and forwarded. Before the I switched to untangle I was behind a netgear firewall router and had the same issue. Very strange that 2 different firewalls are give the same result, I originally thought it was a netgear problem.

    I am about to give up on this cause I have looked at the same things for weeks and can't come up with any new ideas, please help. I am happy to post setting, but am not sure what would be helpful since I dont think this is a problem with asterisk configuration.

    HELP!!!
     
  2. Stinger81

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    First things first:

    Are the trunks registered? Run these commands from the CLI:
    sip show registry
    iax2 show registry

    If they are registered, did you try to set "Allow Anonymous Inbound SIP Calls?" to "Yes" in "General Settings"?
     
  3. slamman

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    Yes to all.



    Host Username Refresh State Reg.Time
    sip.broadvoice.com:5060 xxxxxxxxxx@s 23 Registered Wed, 03 Dec 2008 09:28:05
    sip.broadvoice.com:5060 xxxxxxxxxx@s 23 Registered Wed, 03 Dec 2008 09:28:05



    Host dnsmgr Username Perceived Refresh State
    192.246.69.186:4569 N xxxxxx xxx.xxx.xxx.xxx:4569 60 Registered
    192.246.69.186:4569 N xxxxxx xxx.xxx.xxx.xxx:4569 60 Registered

    General settings show "allow annonymous"
     
  4. Stinger81

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    If you monitor the CLI and make an inbound call, do you see any output?
     
  5. slamman

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    No. That is the problem. I see no evidence of the call anywhere. I don't see that it ever hits my network, but admittedly I am not that familiar with snooping network traffic.

    I don't think the calls are getting to the astersik box, but I don't see evidence that they are being blocked by the firewall. I have tried opening all ports and still nothing. I have put the elastix box in the dmz and still nothing. I don't know how to verify the packets are even reaching my network, but I suspect they are because 2 providers both have the same issue (broadvoice and FWD). I would suspect my ISP but they claim to not be blocking anything.

    I need help to trace packets or monitor at the firewall to see where they are being lost.

    I hope this makes sense. As I said this is not my strong point, and at this point I am totally lost because elastix appears to be configured properly, the packets just don't seem to get there.
     
  6. slamman

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    I have learned a bit about using wireshark and tcpdump and have found some info. The call does appear to be reaching the pbx, but is being rejected. Here is a line from the tcpdump. When looking at the reports it appears that the call is being originated, and the invite is getting to the pbx but for whatever reason the pbx rejects the call and gives the 404 error. The user is a valid working extension.

    6 1.840015 192.168.1.3 147.135.8.128 SIP Status: 404 Not Found




    Any suggestions
     
  7. slamman

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    Here is a screen shot [​IMG]
     
  8. slamman

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    Ok that shot was terrible, lets try again.
     
  9. slamman

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    Wow the upload feature really sucks. This picture is completely worthless but its the best quality this thing will take.

    [​IMG]
     
  10. slamman

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    I am confused by the fact that while watching the cli nothing happens when calling inbound, but now based on the tcpdump I can see that the invite is getting there. Is there somewhere else I should be looking to see what is happening?
     
  11. Stinger81

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    Seems like something can't be found, when the call hits your pbx.
    Do you have your inbound routes correctly defined?
    Did you try an "Any CID/Any DID"?
     
  12. slamman

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    Yes, I have done the "catch all" DID. I have changed my register => to try different extensions, but the call is always rejected. Are there any specific logs I can look at to get more info on what is not being found? Like I said while watching the CLI I don't see any evidence that the pbx even gets an inbound invite request, or anything else for that matter.

    Desperately seeking an answer.
     
  13. Stinger81

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    Hmm... Strange... Where exactly do you see those packets? Are they going from you router to your pbx?

    Further, did you correctly configure the network settings of your pbx? For example, can it ping hosts outside of your network?

    By the way, I'm a bit confused about what you mean with:
     
  14. slamman

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    Yes I can ping to the internet ex. www.yahoo.com

    What I meant was my provider "broadvoice" recomends the registration line in sip.conf [General] look like this: XXXYYYZZZZ@sip.broadvoice.com:pASSWORD:XXXYYYZZZZ@sip.broadvoice.com/600 where 600 is a valid extension in my dial plan. I switched this extension with other working extensions.

    I finally figured out I was not running in debug mode which is why the event wasn't showing up, even though other calls were showing up. I found the relevant lines in the logs. I bolded what appears to be a problem in the logs and will copy it hear for ease of viewing:

    [Dec 6 08:46:36] NOTICE[2559] chan_sip.c: Call from 'broadvoice#' to extension 'broadvoice#' rejected because extension not found.

    "broadvoice#" is my phone number from broadvoice. I wouldn't think this should be my extension, which would seem to be why it is not found. I am going to look into this further but any advice would be appreciated.


    <------------->
    [Dec 6 08:46:19] VERBOSE[2559] logger.c: --- (12 headers 0 lines) ---
    [Dec 6 08:46:19] VERBOSE[2559] logger.c: Really destroying SIP dialog '18fa4a09493fe21070c5097d1a1db3d1@66.159.252.72' Method: OPTIONS
    [Dec 6 08:46:23] VERBOSE[2559] logger.c: Really destroying SIP dialog '0946ccec4bd270de2c30a84d56fa721c@sip.broadvoice.com' Method: REGISTER
    [Dec 6 08:46:23] VERBOSE[2559] logger.c: Really destroying SIP dialog '1ae4476530c7700b4472cea23ee4ab85@sip.broadvoice.com' Method: REGISTER
    [Dec 6 08:46:36] VERBOSE[2559] logger.c:
    <--- SIP read from 147.135.8.128:5060 --->
    INVITE sip:"broadvoice#"@192.168.1.3:5060 SIP/2.0
    Call-ID: 13f0304-3f@147.135.8.128
    CSeq: 1 INVITE
    From: "Anaheim CA"<sip:"Calling#"@147.135.8.128:5060;user=phone>;tag=ehik
    To: "USER"<sip:100@192.168.1.3:5060>
    Via: SIP/2.0/UDP 147.135.8.128:5060
    Contact: <sip:"Calling#"@147.135.8.128:5060>
    Supported: 100rel
    Content-Length: 309
    Content-Type: application/sdp

    v=0
    o=2475100407 10 10 IN IP4 147.135.8.128
    s=-
    c=IN IP4 147.135.8.128
    t=0 0
    m=audio 26340 RTP/AVP 0 18 8 96 97 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:8 PCMA/8000
    a=rtpmap:96 iLBC/8000
    a=fmtp:96 mode=30
    a=rtpmap:97 t38/8000
    a=rtpmap:101 telephone-event/8000

    <------------->
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: --- (10 headers 14 lines) ---
    [Dec 6 08:46:36] WARNING[2559] rtp.c: Unable to set TOS to 184
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Sending to 147.135.8.128 : 5060 (no NAT)
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Using INVITE request as basis request - 13f0304-3f@147.135.8.128
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Found peer 'sip.broadvoice.com'
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Found RTP audio format 0
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Found RTP audio format 18
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Found RTP audio format 8
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Found RTP audio format 96
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Found RTP audio format 97
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Found RTP audio format 101
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Peer audio RTP is at port 147.135.8.128:26340
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Found audio description format PCMU for ID 0
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Found audio description format G729 for ID 18
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Found audio description format PCMA for ID 8
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Found audio description format iLBC for ID 96
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Found unknown media description format t38 for ID 97
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Found audio description format telephone-event for ID 101
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing)
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Peer audio RTP is at port 147.135.8.128:26340
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Looking for 7145947323 in from-ptsn (domain 192.168.1.3)
    [Dec 6 08:46:36] VERBOSE[2559] logger.c:
    <--- Reliably Transmitting (no NAT) to 147.135.8.128:5060 --->
    SIP/2.0 404 Not Found
    Via: SIP/2.0/UDP 147.135.8.128:5060;received=147.135.8.128
    From: "Anaheim CA"<sip:"Calling#"@147.135.8.128:5060;user=phone>;tag=ehik
    To: "USER"<sip:100@192.168.1.3:5060>;tag=as08c8110c
    Call-ID: 13f0304-3f@147.135.8.128
    CSeq: 1 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Content-Length: 0


    <------------>
    [Dec 6 08:46:36] NOTICE[2559] chan_sip.c: Call from 'broadvoice#' to extension 'broadvoice#' rejected because extension not found.
    [Dec 6 08:46:36] VERBOSE[2559] logger.c: Scheduling destruction of SIP dialog '13f0304-3f@147.135.8.128' in 6400 ms (Method: INVITE)
    [Dec 6 08:46:36] VERBOSE[2559] logger.c:
    <--- SIP read from 147.135.8.128:5060 --->
    ACK sip:"broadvoice#"@192.168.1.3:5060 SIP/2.0
    Call-ID: 13f0304-3f@147.135.8.128
    CSeq: 1 ACK
    From: "Anaheim CA"<sip:"calling#"@147.135.8.128:5060;user=phone>;tag=ehik
    To: "USER"<sip:100@192.168.1.3:5060>;tag=as08c8110c
    Via: SIP/2.0/UDP 147.135.8.128:5060
    Content-Length: 0
     
  15. Stinger81

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    Someone else may say whether that register string either is or isn't correct.

    This, as far as I know, says that it can't find the inbound route. Are you very sure that you have an Any CID/Any DID?
    Try deleting it and recreating it with only changing the destination field.

    Further, are you trying to call from one extension to another through broadvoice? That's what it looks like.


    You might want to try a softphone and see if you can get incoming calls directly on a softphone, bypassing your pbx. If that works, then you can try similar settings in your trunk.
     
  16. slamman

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    Thanks for your help. I found the problem. Somewhere along the line while trying different trunk config settings I copied some from somewhere on the web and there had been a typo in the context. When I started I didn't understand what the settings meant and now I have a basic understanding which allowed me to spot the problem while looking through my extensions.conf.

    Thankfully while trouble shooting this problem I learned alot about the flow of a call and monitoring packets that will be helpful in the future. Now I can move on tho the next problem, which is now that I am getting the call to ring thru the call drops immediately upon answering. I am going to do some research on the issue and if I can't find a solution I will post another question under a separate thread.
     
  17. Stinger81

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    That's great!!!

    The issue you are experiencing now, also occurs with me... But only with one specific cell phone, on certain (random) days. On other days, there isn't a problem at all...
    Did you try multiple lines/phones from outside of your pbx?
     
  18. slamman

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    I have tried 2 different cell phones, but thats it so far. I don't know if this is normal as well but the call also disconnects as soon as it roles to voicemail.

    It is kind of disheartening to hear that this could be a "normal" occurance.
     
  19. Stinger81

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    I'm not saying that this is normal behavior, just that it happens here too.
    You tried only (two) cell phones? You might want to try a land line.

    You could start now by analyzing the log and playing with option in the trunk and inbound route.
     
  20. slamman

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    It looks like I finally got it working. It is now receiving calls from the cell phones that were dropping previously as well as a land line. Once the calls were ringing the box, I went back to my trunk configurations which I had messed with quite a bit while trying to solve the problem and removed a bunch of test configurations to see what would happen. Once I did the pbx stopped dropping calls. I think one of the codecs was the problem.
     

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