No such number tone instead of Asterisk voice

Discussion in 'General' started by Chilling_Silence, Oct 21, 2008.

  1. Chilling_Silence

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    Hey all,

    I have a small call center that uses a box Ive setup for them. They do a lot of cold-calling.
    When they call a number which doesnt exist, they get "All circuits are busy now, please try your call again later".
    Really it should just give them a message which says "The number you are dialing is not in service, please check the number and try again", because the number doesnt exist.
    I know it doesnt exist because of the message I get when I ring it from my Cellphone.

    Anyway, how can I change it? Heres what a regular call looks like:
    -- Executing [4224604@from-internal:1] Macro("SIP/1207-b760f328", "user-callerid|SKIPTTL|") in new stack
    -- Executing [s@macro-user-callerid:1] NoOp("SIP/1207-b760f328", "user-callerid: device 1207") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/1207-b760f328", "AMPUSER=1207") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("SIP/1207-b760f328", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("SIP/1207-b760f328", "1|Set|REALCALLERIDNUM=1207") in new stack
    -- Executing [s@macro-user-callerid:5] NoOp("SIP/1207-b760f328", "REALCALLERIDNUM is 1207") in new stack
    -- Executing [s@macro-user-callerid:6] Set("SIP/1207-b760f328", "AMPUSER=1207") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/1207-b760f328", "AMPUSERCIDNAME=Tippy") in new stack
    -- Executing [s@macro-user-callerid:8] GotoIf("SIP/1207-b760f328", "0?report") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/1207-b760f328", "AMPUSERCID=1207") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/1207-b760f328", "CALLERID(all)="Tippy" <1207>") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/1207-b760f328", "REALCALLERIDNUM=1207") in new stack
    -- Executing [s@macro-user-callerid:12] ExecIf("SIP/1207-b760f328", "0|Set|CHANNEL(language)=") in new stack
    -- Executing [s@macro-user-callerid:13] NoOp("SIP/1207-b760f328", "TTL: ARG1: SKIPTTL") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("SIP/1207-b760f328", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,23)
    -- Executing [s@macro-user-callerid:23] NoOp("SIP/1207-b760f328", "Using CallerID "Tippy" <1207>") in new stack
    -- Executing [4224604@from-internal:2] Set("SIP/1207-b760f328", "_NODEST=") in new stack
    -- Executing [4224604@from-internal:3] Macro("SIP/1207-b760f328", "record-enable|1207|OUT|") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/1207-b760f328", "0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/1207-b760f328", "recordingcheck|20081021-133147|1224549107.2220") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
    recordingcheck|20081021-133147|1224549107.2220: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] NoOp("SIP/1207-b760f328", "No recording needed") in new stack
    -- Executing [4224604@from-internal:4] Macro("SIP/1207-b760f328", "dialout-trunk|2|4224604||") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/1207-b760f328", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] ExecIf("SIP/1207-b760f328", "0|Authenticate|") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1207-b760f328", "0?disabletrunk|1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/1207-b760f328", "DIAL_NUMBER=4224604") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/1207-b760f328", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/1207-b760f328", "GROUP()=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1207-b760f328", "0?nomax") in new stack
    -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/1207-b760f328", "0?chanfull") in new stack
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1207-b760f328", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/1207-b760f328", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/1207-b760f328", "outbound-callerid|2") in new stack
    -- Executing [s@macro-outbound-callerid:1] GotoIf("SIP/1207-b760f328", "1?start") in new stack
    -- Goto (macro-outbound-callerid,s,3)
    -- Executing [s@macro-outbound-callerid:3] NoOp("SIP/1207-b760f328", "REALCALLERIDNUM is 1207") in new stack
    -- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/1207-b760f328", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,9)
    -- Executing [s@macro-outbound-callerid:9] Set("SIP/1207-b760f328", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:10] Set("SIP/1207-b760f328", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:11] Set("SIP/1207-b760f328", "TRUNKOUTCID=099294928") in new stack
    -- Executing [s@macro-outbound-callerid:12] GotoIf("SIP/1207-b760f328", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,16)
    -- Executing [s@macro-outbound-callerid:16] GotoIf("SIP/1207-b760f328", "0?usercid") in new stack
    -- Executing [s@macro-outbound-callerid:17] Set("SIP/1207-b760f328", "CALLERID(all)=099294928") in new stack
    -- Executing [s@macro-outbound-callerid:18] GotoIf("SIP/1207-b760f328", "1?report") in new stack
    -- Goto (macro-outbound-callerid,s,22)
    -- Executing [s@macro-outbound-callerid:22] NoOp("SIP/1207-b760f328", "CallerID set to "" <099294928>") in new stack
    -- Executing [s@macro-dialout-trunk:12] AGI("SIP/1207-b760f328", "fixlocalprefix") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
    -- AGI Script fixlocalprefix completed, returning 0
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/1207-b760f328", "OUTNUM=4224604") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/1207-b760f328", "custom=SIP/099294928") in new stack
    -- Executing [s@macro-dialout-trunk:15] GotoIf("SIP/1207-b760f328", "1?gocall") in new stack
    -- Goto (macro-dialout-trunk,s,17)
    -- Executing [s@macro-dialout-trunk:17] Macro("SIP/1207-b760f328", "dialout-trunk-predial-hook|") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1207-b760f328", "0?bypass|1") in new stack
    -- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/1207-b760f328", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:20] Dial("SIP/1207-b760f328", "SIP/099294928/4224604|300|") in new stack
    -- Called 099294928/4224604
    -- SIP/099294928-086fe848 is making progress passing it to SIP/1207-b760f328
    -- SIP/099294928-086fe848 is circuit-busy
    == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s@macro-dialout-trunk:21] Goto("SIP/1207-b760f328", "s-CONGESTION|1") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
    -- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/1207-b760f328", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,3)
    -- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/1207-b760f328", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
    -- Executing [4224604@from-internal:5] Macro("SIP/1207-b760f328", "outisbusy|") in new stack
    -- Executing [s@macro-outisbusy:1] Playback("SIP/1207-b760f328", "all-circuits-busy-now|noanswer") in new stack
    -- <SIP/1207-b760f328> Playing 'all-circuits-busy-now' (language 'en')
    -- Executing [s@macro-outisbusy:2] Playback("SIP/1207-b760f328", "pls-try-call-later|noanswer") in new stack
    -- <SIP/1207-b760f328> Playing 'pls-try-call-later' (language 'en')
    -- Executing [s@macro-outisbusy:3] Macro("SIP/1207-b760f328", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/1207-b760f328", "w") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("SIP/1207-b760f328", "") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("SIP/1207-b760f328", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("SIP/1207-b760f328", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("SIP/1207-b760f328", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("SIP/1207-b760f328", "") in new stack
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1207-b760f328' in macro 'hangupcall'
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1207-b760f328' in macro 'outisbusy'
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1207-b760f328'

    I get the feeling the issue is where it says:
    -- Called 099294928/4224604
    -- SIP/099294928-086fe848 is making progress passing it to SIP/1207-b760f328
    -- SIP/099294928-086fe848 is circuit-busy
    == Everyone is busy/congested at this time (1:0/1/0)

    Problem is all the lines arent busy...

    Any ideas on fixing this?

    Thanks


    Chill.
     
  2. Buks

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    I dont know the answer to your problem, but it seems that there is a serious bug in Elastix with outbound routes. I also experience "all circuits are busy' for no reason from time to time.

    Elastix might not be as stable as they say....
     
  3. Chilling_Silence

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    Actually, I dont think its an issue with the stability.
    When you get it next, open an asterisk console and run:
    sip show registry
    See if your trunks are online. It could even be your router...
    Mine is definitely giving me that error for "No such number", I ran a sip debug:
    <--- SIP read from 202.180.76.161:5060 --->
    SIP/2.0 404 Not Found

    Thats the reply it gets when dialing the invalid number, so its definitely not an asterisk "problem". Same deal happens with trixbox even.
    Just investigating changing the message played now :)
     
  4. Buks

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    It is NOT my router. I have been using this router (Linksys WRT54GL) for at least 17 months with trixbox with no problems. I have good knowledge of networks etc as I own a small IT service company. I have more than a dozen production trixboxes in the field with the normal glitches associated with a VOIP PBX which I can easily address over the net.

    I was persuaded by a supplier to switch to Elastix because of its claim of stability. I have to disagree with this. Just read this forum and see how many people suddenly are experiencing strange behavior with outbound and inbound pstn calls. This happened right after they did a yum -y update. Surely the developers of the "Reliable PBX..." should break their silence now, admit that they shipped a flawed version and give us version 1.4 (or whatever).

    The problem is not with Asterisk or Freepbx. There is a bug in the code of Elastix after the upgrade to version 1.3x. Again - just read all the posts (and no real cure from the developers) and you will see.
     
  5. Chilling_Silence

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    Thats all very well and good, but your issue is completely separate to mine, as I get this same issue with my SIP provider regardless of using Trixbox or Elastix

    Try contributing something useful, or go back to Trixbox if you feel its so much better.
    You're not the only one who owns a small IT company with a good knowledge of networks.
    Start a thread of your own for your separate issue, and begin by looking in the asterisk console, do a SIP debug and see what message you get when it gives you "All Circuits Are Busy", as well as a sip show registry.
     
  6. Buks

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    You are aggressive and obviously just out looking for a fight. It is not necessary and will help no one, least of all yourself. The reason I mentioned trixbox, was because I found it to be more stable than Elastix 1.3x and I am hoping that one of the developers of this product would read this and hasten to support their claim of the "Most Reliable PBX..."

    All I wanted to bring under your attention, was the fact that the "All circuits are busy..." a common error is with people using Elastix version 1.3x, especially after a yum update. Surely, after you have thought of this for a moment and calmed doen you must realize that there is truth in this.

    I merely tried to give you lead - For interest sake - my system gives the "your call cannot be completed as dialed...." if I dial a non-existing number.

    Please, I do not wish to quibble with you and I do wish to use Elstix, as long as it IS the most reliable PBX and I can do a yum update without fear and with the knowledge that the developers will quickly fix a bug that has slipped into the system and not quietly read messages posted on the forum without acting.

    Cheers
     
  7. Chilling_Silence

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    When I dial a number that doesnt exist, it still gives me "All circuits are busy now, please check the number and try again".

    It should be giving me something like "Your call cannot be completed as dialed, please check the number and try again".

    Im using elastix-1.2, but have the same issue if I use trixbox for example...
    asterisk-1.4.21.2 & asterisk-1.4.22

    Any assistance would be greatly appreciated. I have a feeling it has to do with how its handling the SIP 404 response

    Thanks
     
  8. dgordillo

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    Hello friends,

    I have a elastix 1.2 installed on hp proliant ML110 whit a openvox card 4 fxo and 4 fxs ports. All work good but in a random moment the people try make a call using analog line to pstn and they listen "All service are busy now, please try again later".

    I don´t have idea to solve the problem, some one know a solution, tellme. tks:dry:
     
  9. Chilling_Silence

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    Hi dgordillo,

    Your problem is a completely separate issue from this threads topic, I would suggest you start a new thread to get help with it :)
    When you start a new thread, you might want to be specific and tell people when the issue occurs (e.g. after a few calls, first call on boot, always happens etc)

    Welcome to the Elastix Forums

    Cheers


    Chill.
     
  10. Buks

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    I am not sure if this is such a "completely seperate issue", but here is what worked for me:

    I use a somewhat older TDM400 card with FXO ports on it. I also occasionally got the "all circuits are busy" message. This also happened on newer model Digium TDM cards.

    The following COMPLETELY solved my problem:

    I edited the zaptel.conf, zapata.conf as well as zapata-channels.conf and changed the kewlstart settings to loopstart. eg: fxsks to fxsls. Make sure you do all of them and then bounce the box. (restert)

    This seems to be an issue with the later Asterisk versions....
     
  11. Chilling_Silence

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    Just to clarify, yeah, it is totally separate.

    When I dial out, I wanted a "no such number" tone rather than the asterisk voice in a 100% SIP environment (If I had PSTN as a fallback it would not be an issue).

    This user randomly is getting "All circuits are busy now" when dialing out a PSTN card.
     

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