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FOOD FOR THOUGHT Wazo 17.12 issues

Discussion in 'Open Discussion' started by DavidBer, Sep 13, 2017.

  1. DavidBer

    DavidBer Member

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    My configuration is simple:

    4 grandstream phones, one grandstream ATA.

    The four phones are in a ring group. The ATA is outside it. There are 5 incoming lines. Two numbers go to the ring group, two go straight to a voice mail and one goes to the ATA.

    The system is at RentPBX and I am using flowroute for sip provider. (I have another PIAF (bronze?) running at RentPBX also connected to flowroute)

    The wazo box is doing three odd things.

    1. I cannot make any outbound tollfree calls.

    [​IMG]

    When I dial outbound to the 18XX I get a fast busy. If I strip the 1 and dial 8XX I get the person you are calling is busy . . .


    2. I cannot *98 into a mailbox

    [​IMG]

    3. I cannot *98(mailbox number)

    [​IMG]

    I have read and re-read http://nerdvittles.com/?p=20675

    I am lost as to how to fix these three. I have checked my older PIAF install for dialplan and this wazo config matches.

    Ideas?

    Thanks in advance.

    dave
     
  2. DavidBer

    DavidBer Member

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    After numerous sessions of pulling my hair out, I figured some of the voice mail out.

    As I mentioned, I have 5 incoming lines.

    3 of the lines look like this:

    DID -> VoiceMail

    1 line looks like this

    DID ->Group Ring (3 extensions)

    1 line looks like this

    DID->Extension

    I just went to users->clicked a user -> voicemail and then enabled the voicemail for the user. I was able to either choose an existing voicemail box or create a new one. After creating one for that user and using that users phone, *98 worked.

    I was not able to *98100 to get to mailbox 100 though. I was not able to *98 and hit 0 to try a different box.

    I really do not need a user based voicemail. I need voicemail boxes that I can assign to a DID and be able to configure them.

    Anyone have ideas/solutions how to do this? Also, still stuck on the tollfree numbers :(
     
  3. DavidBer

    DavidBer Member

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    Now I am even more confused :(

    I pulled asterisk logs.

    This is a call to a non toll free number - notice it goes through flowroute. (IPs,names, numbers changed)

    Code:
    [Sep 14 09:36:55] VERBOSE[31498] chan_sip.c: Registered SIP 'lox4wzzz' at 184.181.181.181:57508
    [Sep 14 09:36:55] VERBOSE[31498] chan_sip.c: Saved useragent "X-Lite release 5.0.1 stamp 86895" for peer lox4wzzz
    [Sep 14 09:37:01] VERBOSE[31498][C-0000018d] netsock2.c: Using SIP RTP CoS mark 5
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [8055551212@default:1] Set("SIP/lox4wzzz-000001f5", "XIVO_BASE_CONTEXT=default") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [8055551212@default:2] Set("SIP/lox4wzzz-000001f5", "XIVO_BASE_EXTEN=8055551212") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [8055551212@default:3] Gosub("SIP/lox4wzzz-000001f5", "outcall,s,1(3,)") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@outcall:1] Set("SIP/lox4wzzz-000001f5", "XIVO_DSTID=3") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@outcall:2] Set("SIP/lox4wzzz-000001f5", "XIVO_PRESUBR_GLOBAL_NAME=OUTCALL") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@outcall:3] Set("SIP/lox4wzzz-000001f5", "XIVO_SRCNUM=799") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@outcall:4] Set("SIP/lox4wzzz-000001f5", "XIVO_DSTNUM=8055551212") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@outcall:5] Set("SIP/lox4wzzz-000001f5", "XIVO_CONTEXT=default") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@outcall:6] Set("SIP/lox4wzzz-000001f5", "WAZO_CHANNEL_DIRECTION=to-wazo") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@outcall:7] Gosub("SIP/lox4wzzz-000001f5", "originate-caller-id,s,1") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@originate-caller-id:1] GotoIf("SIP/lox4wzzz-000001f5", "0?:name") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx_builtins.c: Goto (originate-caller-id,s,3)
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@originate-caller-id:3] GotoIf("SIP/lox4wzzz-000001f5", "0?:fix") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx_builtins.c: Goto (originate-caller-id,s,5)
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@originate-caller-id:5] GotoIf("SIP/lox4wzzz-000001f5", "?:end") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx_builtins.c: Goto (originate-caller-id,s,9)
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@originate-caller-id:9] Return("SIP/lox4wzzz-000001f5", "") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@outcall:8] AGI("SIP/lox4wzzz-000001f5", "agi://127.0.0.1/outgoing_user_set_features") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] res_agi.c: agi://127.0.0.1/outgoing_user_set_features: AGI handler 'outgoing_user_set_features' successfully executed
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] res_agi.c: <SIP/lox4wzzz-000001f5>AGI Script agi://127.0.0.1/outgoing_user_set_features completed, returning 0
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@outcall:9] Gosub("SIP/lox4wzzz-000001f5", "xivo-subroutine,s,1()") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@xivo-subroutine:1] GotoIf("SIP/lox4wzzz-000001f5", "?:nosubroutine") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx_builtins.c: Goto (xivo-subroutine,s,4)
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@xivo-subroutine:4] Return("SIP/lox4wzzz-000001f5", "") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@outcall:10] Gosub("SIP/lox4wzzz-000001f5", "xivo-user_rights_check,s,1") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@xivo-user_rights_check:1] AGI("SIP/lox4wzzz-000001f5", "agi://127.0.0.1/user_set_call_rights") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] res_agi.c: agi://127.0.0.1/user_set_call_rights: AGI handler 'user_set_call_rights' successfully executed
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] res_agi.c: <SIP/lox4wzzz-000001f5>AGI Script agi://127.0.0.1/user_set_call_rights completed, returning 0
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@xivo-user_rights_check:2] GotoIf("SIP/lox4wzzz-000001f5", "ALLOW?:error,1") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@xivo-user_rights_check:3] GotoIf("SIP/lox4wzzz-000001f5", "1?allow,1") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx_builtins.c: Goto (xivo-user_rights_check,allow,1)
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [allow@xivo-user_rights_check:1] NoOp("SIP/lox4wzzz-000001f5", "User allowed to make call") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [allow@xivo-user_rights_check:2] Return("SIP/lox4wzzz-000001f5", "") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@outcall:11] AGI("SIP/lox4wzzz-000001f5", "agi://127.0.0.1/check_schedule") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] res_agi.c: agi://127.0.0.1/check_schedule: AGI handler 'check_schedule' successfully executed
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] res_agi.c: <SIP/lox4wzzz-000001f5>AGI Script agi://127.0.0.1/check_schedule completed, returning 0
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@outcall:12] GotoIf("SIP/lox4wzzz-000001f5", "0?CLOSED,1") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@outcall:13] GotoIf("SIP/lox4wzzz-000001f5", "?:15") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx_builtins.c: Goto (outcall,s,15)
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@outcall:15] GotoIf("SIP/lox4wzzz-000001f5", "SIP/Flowroute?:error,1") in new stack
    
    
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@xivo-subrgbl-outcall:8] Return("SIP/lox4wzzz-000001f5", "") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [s@xivo-global-subroutine:6] Return("SIP/lox4wzzz-000001f5", "") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [dial@outcall:5] CELGenUserEvent("SIP/lox4wzzz-000001f5", "XIVO_OUTCALL") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [dial@outcall:6] Set("SIP/lox4wzzz-000001f5", "CONNECTEDLINE(num,i)=8055551212") in new stack
    [Sep 14 09:37:01] VERBOSE[3353][C-0000018d] pbx.c: Executing [dial@outcall:7] Dial("SIP/lox4wzzz-000001f5", "SIP/Flowroute/18055551212,,o(8055551212)") in new stack
    

    Now this is trying to call a toll free. (IPs,names, numbers changed)

    What is this ideasip issue?

    Code:
    [Sep 14 09:34:08] VERBOSE[2660][C-0000018c] pbx.c: Executing [18889781234@default:1] Dial("SIP/lox4wzzz-000001f3", "SIP/18889781234@proxy.ideasip.com") in new stack
    [Sep 14 09:34:08] VERBOSE[2660][C-0000018c] netsock2.c: Using SIP RTP CoS mark 5
    [Sep 14 09:34:08] VERBOSE[2660][C-0000018c] app_dial.c: Called SIP/18889781234@proxy.ideasip.com
    [Sep 14 09:34:08] VERBOSE[2660][C-0000018c] app_dial.c: SIP/proxy.ideasip.com-000001f4 answered SIP/lox4wzzz-000001f3
    [Sep 14 09:34:08] VERBOSE[2663][C-0000018c] bridge_channel.c: Channel SIP/proxy.ideasip.com-000001f4 joined 'simple_bridge' basic-bridge <8390cefc-7800-4db9-8df4-f658037aed9f>
    [Sep 14 09:34:08] VERBOSE[2660][C-0000018c] bridge_channel.c: Channel SIP/lox4wzzz-000001f3 joined 'simple_bridge' basic-bridge <8390cefc-7800-4db9-8df4-f658037aed9f>
    [Sep 14 09:34:08] VERBOSE[2660][C-0000018c] bridge.c: Bridge 8390cefc-7800-4db9-8df4-f658037aed9f: switching from simple_bridge technology to native_rtp
    [Sep 14 09:34:08] VERBOSE[2660][C-0000018c] bridge_native_rtp.c: Locally RTP bridged 'SIP/lox4wzzz-000001f3' and 'SIP/proxy.ideasip.com-000001f4' in stack
    [Sep 14 09:34:08] VERBOSE[2660][C-0000018c] res_rtp_asterisk.c: 0xb25c54c8 -- Probation passed - setting RTP source address to 184.187.191.174:56372
    [Sep 14 09:34:09] VERBOSE[2663][C-0000018c] res_rtp_asterisk.c: 0xb6832148 -- Probation passed - setting RTP source address to 208.97.25.12:10280
    [Sep 14 09:34:10] VERBOSE[2660][C-0000018c] bridge_channel.c: Channel SIP/lox4wzzz-000001f3 left 'native_rtp' basic-bridge <8390cefc-7800-4db9-8df4-f658037aed9f>
    [Sep 14 09:34:10] VERBOSE[2660][C-0000018c] pbx.c: Spawn extension (default, 18889781234, 1) exited non-zero on 'SIP/lox4wzzz-000001f3'
    [Sep 14 09:34:10] VERBOSE[2660][C-0000018c] app_stack.c: SIP/lox4wzzz-000001f3 Internal Gosub(hangup_handlers,userevent,1) start
    [Sep 14 09:34:10] VERBOSE[2660][C-0000018c] pbx.c: Executing [userevent@hangup_handlers:1] NoOp("SIP/lox4wzzz-000001f3", "Sending Hangup userevent") in new stack
    [Sep 14 09:34:10] VERBOSE[2660][C-0000018c] pbx.c: Executing [userevent@hangup_handlers:2] UserEvent("SIP/lox4wzzz-000001f3", "Hangup,XIVO_USERUUID: d3b7b8e2-f6a7-4517-89e6-a68cf8ee7ae9") in new stack
    [Sep 14 09:34:10] VERBOSE[2663][C-0000018c] bridge_channel.c: Channel SIP/proxy.ideasip.com-000001f4 left 'native_rtp' basic-bridge <8390cefc-7800-4db9-8df4-f658037aed9f>
    [Sep 14 09:34:10] VERBOSE[2660][C-0000018c] pbx.c: Executing [userevent@hangup_handlers:3] Return("SIP/lox4wzzz-000001f3", "") in new stack
    [Sep 14 09:34:10] VERBOSE[2660][C-0000018c] app_stack.c: Spawn extension (default, 18889781234, 1) exited non-zero on 'SIP/lox4wzzz-000001f3'
    [Sep 14 09:34:10] VERBOSE[2660][C-0000018c] app_stack.c: SIP/lox4wzzz-000001f3 Internal Gosub(hangup_handlers,userevent,1) complete GOSUB_RETVAL=
    
    [​IMG]

    Where is the ideasip coming from?
     
  4. DavidBer

    DavidBer Member

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    Jesus christ.

    Is this what is intended for a default install?

    [​IMG]


    I normally do not bash PIAF at all. Ward and the team have done an awesome job.

    But this . . . WOW. At least put this in the soup to nuts discussion. I have spent two days trying to figure out what the heck is going on.
     
  5. mainenotarynet

    mainenotarynet Not really a Guru - Just a long time user

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    You want confused -- try reading the 675 page Wazo PDF file where the first 80 or so pages is how to upgrade from one to another. The manual has ONE chapter (that needs to be fixed - so If I need specifics on something I can skip to the correct chapter (yes it has bookmarks but ONE chapter...Really.

    I am not stupid but I just plain gave up on EVER being able to use this rocket.

    Great anaolgy would be IncrediblePBX with the FreePBX-Like GUI is a Mercedes Benz -- Powerful and easy to drive Wazo is a Boeing 777 and I have no Pilot's license -- Yes it may be more powerful and go farther, but if you can't fly it - You'll crash and burn.

    Hope you have better luck than I do.

    Might I suggest something though: comment out that entire section and go to your outbound routes (if Wazo has such a section) and add the conditions there, then use one of your FlowRoute trunks or even a GoogleVoice to place the 8[03-8][03-8] calls. The reason I wrote it this way instead of 888,877... is it is one line but does not include area codes such as 801, 802, 812, etc which would be a problem.

    Have better luck than I do.
     
  6. DavidBer

    DavidBer Member

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    I spent a lot of time reading that damn 675 page PDF. It is not that good.

    I am going off topic here, cuz you dragged me into it :) But OMG, the older PIAF bronze/gold/purple, whatever were so much more intuitive. I have been using PIAF for the last ~7 years. I came here after doing asterisk/freepbx setups on ClearOs pc based routers. I used to maintain the how-to there!

    I love the work that Ward and the team do. I think they are just at a crossroads because others have not stayed true to the free, as in beer, model. FreePBX has gone with commercial modules. So they have been trying to come up with solutions while staying true to their core beliefs. I appreciate that. They see Wazo as the replacement for Xivo which was the replacement for FreePBX. This also ties into their belief of do nothing unless you tell me to do it security model.

    I think Wazo is interesting, to say the least. It is not that far from FreePBX, but it is enough different to confuse me. It also has some features that put FreePBX to shame.

    I am going to spin up a 3cx install to see if I like it better. I am also going to try this issabel thing as well. Honestly, I want people to earn a living and have no problem buying commercial modules, or even paying an entry fee. Ultimately, I think that is where this is all headed to.

    Now back to the toll frees . . .

    I commented out those lines. I let my dial plan take over.
     
  7. wardmundy

    wardmundy Nerd Uno

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    The latest Incredible PBX 13 build with CentOS 6.9 and the latest Asterisk 13 is as close to "the old days" as you can get... if you don't like change (and many of us don't). We're offering (free) open source, GPL, no gotcha's choices. That's the best we can do.
     
  8. mainenotarynet

    mainenotarynet Not really a Guru - Just a long time user

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    Well now my problem is that Vultr has the CentOS7 that is installed -- if you want the 6.9 it's an ISO that is getting stuck that you have to MANUALLY go through the install as if it were a Physibox. I am trying the 23403 with CentOS7 -- Beginning is throwing a crapload of errors due to touch commands in the /proc/ statements as no such file or directory.

    If this doesn't work I may have to fight with Wazo weathter I want to or not -- won't be doing the HA though as Woothosting was horrible -- DO is 1/2 what Vultr resources for same price and I amm not that affluent to afford 2 Vultr same size instances.

    I'm going to be bald by the end of this migration :)
     
  9. DavidBer

    DavidBer Member

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    Ward, this is why you have my respect. You have shown your integrity and how it drives you and the team. In my book, that is extremely commendable.

    Thank you for the great work and product.


    BTW . . . I assume you are going to patch the installer to not have that tollfree dial plan in the extrafeatures anymore :)
     
  10. wardmundy

    wardmundy Nerd Uno

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    You're making this too hard. Don't use CentOS 7. Use whatever CentOS 6 they have. It will be updated to 6.9 as part of the Incredible PBX install. Many of the errors you are seeing are because the installer tries to set a current timestamp on every directory. Some of them obviously are protected so you can't do it. Nothing to worry about. It just skips them. This procedure is required by Asterisk to get current time stamps, and it's been in place since the very first version of PIAF3.
     
    #10 wardmundy, Sep 14, 2017
    Last edited: Sep 15, 2017
  11. Dave Gray

    Dave Gray Guru

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    Do not worry about anything it complains of in /proc. /proc is an access point for info that lives in the kernel. Those are not real files, they correspond to processes in the system, and some kernel structures that are handy to have (like /proc/partitions, which tell what partitions the kernel sees on your disks, whether or not you're doing anything (mounting) with them.)

    touch is not going to succeed in that directory, period.

    Stupid proc trick:
    cat /proc/<processId>/environ |tr "\0" "\n"
    gets you a list of processId's environment variables, which can be the source of some really crazy bugs, especially when processId is a daemon that you don't know how it's being started.
     
  12. jake372

    jake372 Member

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    wardmundy likes this.
  13. DavidBer

    DavidBer Member

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    @jake372

    I never had an issue with flowroute making calls at all. I just had issues with the dialplan that was hidden in the xivo-extrafeatures.confs file.

    I am not using IP based authentication for outbound, I am using the password based authentication. Thanks for the pointer though if I go there with IP based.

    Right now, the new issue I have is the https://ipaddress/admin to add plugins is giving me an internal server error 500

    I am trying to setup the IVR and would REALLY like to do it via a gui :)
     
  14. jeff.h

    jeff.h Guru

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    If you want to dial 10 digitis and have it work on Flowroute you need to add a 1 in the Extern Prefix field for NXXNXXXXXX because Flowroute only accepts 11 digit numbers.
     
    krakastan likes this.
  15. DavidBer

    DavidBer Member

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    @jeff.h

    I don't think that is true. If you scroll up and look at the logs that I listed in this thread, you will see a 8055551212 as a dial out. That number was an 805 number, I just changed the rest. There was no 1805xxxxxxx. And if you look at the dialplan I have in the first post, there is no adding a 1.

    Like I mentioned, everything was working fine, inbound and outbound except for tollfrees. In the install, in the xivo-extrafeatures.conf was HARDCODED a toll free dialplan to a different sip provider. There is no way I think that was intentional.

    Right now, this is what I am getting:

    [​IMG]

    I have tried clicking official on/off.

    I looked in the /root directory for wazo-plugind-cli and didn't find it. I figured screw it, just install via command line.

    Anyone have any ideas how to solve this? Is this a firewall issue where it is blocked?

    Thanks
     
  16. jeff.h

    jeff.h Guru

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    I can only speak from my experience with Flowroute and my recent trial of Wazo. I use Flowroute almost exclusively at this point and they will not accept a call from any of my PBXs unless I prepend a 1 and send it as 11 digits. Otherwise I get the caller is busy error because its trying to route to an internal extension that does not exist.
     
  17. DavidBer

    DavidBer Member

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    @jeff.h

    Honestly, I'd love to compare notes. I only use flowroute. I have it at roughly a dozen installs I have done. I love them. For me, this is the first time I have used Wazo. So I just cheated and took my dialplan from the other installs and cloned it. I am lazy like that :) I am not professing to be the smartest guy here at all, I probably know more than a newbie and that makes me more dangerous.

    If you want, I can screenshot my flowroute dashboard settings and pm them to you. Perhaps that is where we are different, I dunno.
     
  18. jeff.h

    jeff.h Guru

    Joined:
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    Sure, I can tell you that my outbound dial plan was 1NXXNXXXXXX and extern prefix 1 with NXXNXXXXXX and that worked for me.

    I will also say that I had an issue with inbound calls showing 001XXXXXXXXXX and UNKNOWN as the caller ID.

    That along with the overall lack of intutiveness of the system have caused me to give up on it for now and have actually destroyed my instance of it.
     
  19. DavidBer

    DavidBer Member

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    Ok, bumping this up.

    I am getting internal server error 500s on the /admin side when trying to add plugins.

    I thought maybe something was screwy, so I installed another instance this time on DO instead of RentPBX. I am getting it on that as well.

    Anyone have an idea what is going on here?

    IS THIS IPTABLES RELATED?
     

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