Unable to dial other Extensions

Discussion in 'General' started by gbrook, May 21, 2008.

  1. gbrook

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    I have installed a number of Trixbox and PIAF with minimum trouble and decided to try Elastix,

    I have programmed a number of SIP Extensions which are all Registered and have Dial Tone.

    No matter what extension I dial I get Busy Tone.
    Example Extn 305 dialling 306 and I get the following captured from the CLI
    ;----------------------------------------------------------
    -- Executing [s@macro-dial:3] AGI("SIP/305-b7700468", "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 'General Office' number is '305'
    dialparties.agi: USE_CONFIRMATION: 'FALSE'
    dialparties.agi: RINGGROUP_INDEX: ''
    dialparties.agi: Methodology of ring is 'none'
    -- dialparties.agi: Added extension 306 to extension map
    -- dialparties.agi: Extension 306 cf is disabled
    -- dialparties.agi: Extension 306 do not disturb is disabled
    > dialparties.agi: extnum 306 has: cw: 0; hascfb: 0 [] hascfu: 0 []
    > dialparties.agi: ExtensionState: 0
    dialparties.agi: Extension 306 has ExtensionState: 0
    -- dialparties.agi: Checking CW and CFB status for extension 306
    -- dialparties.agi: dbset CALLTRACE/306 to 305
    -- dialparties.agi: Filtered ARG3: 306
    == Manager 'admin' logged off from 127.0.0.1
    -- AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:4] NoOp("SIP/305-b7700468", "Returned from dialparties with no extensions to call and DIALSTATUS: ") in new stack
    ;---------------------------------------------------
    Interestingly in a working replica TrixBox for 305 dialling 306 - The control is returned to Executing [s@macro-dial:7] not Executing [s@macro-dial:4]

    Anyone able to help?

    Cheers
    Garry
     
  2. Bob

    Bob

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    Can you explain how you created the extensions under Elastix e.g. Batch Extension module, restored from backup, manually added??

    The reason why I ask these questions is that generally extensions usually pose no issues at all.

    Also what phones are you using??

    Regards

    Bob
     
  3. gbrook

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    Manually through Extensions in FreePBX

    Aastra 57i, Linksys 942 and a Linksys SPA8000

    They have all Registered and are on the same subnet as each other

    I agree - I have never had this in about 30 Installs.

    Cheers
    GB
     
  4. Bob

    Bob

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    What version of Freepbx have you upgraded to???

    Otherwise it is a little baffling as the log looks ok (at least in terms of phone availability).

    Regards

    Bob
     
  5. gbrook

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    The Core is 2.4.0.1

    Regards
    Garry
     
  6. gbrook

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    The Core is 2.4.0.1

    Regards
    Garry
     
  7. Bob

    Bob

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    Whilst I have done it a couple of times (on non-production machines), I found that upgrading to Freepbx 2.4 via the Freepbx upgrade system, causes more issues that what it is worth. I have not spent time to find out what is affected, but needless to say, I have not have the confidence.

    The one system that I do have in production, that is using 2.4 was upgraded using the Freepbx install procedure (script), not upgrading via the GUI.

    I suspect that Elastix has made some changes that are not expected by the 2.4 upgrade, e.g. rights, directories expected, etc, that cause the upgrade to fail.

    You may still be able to perform this update via this method (e.g. installing from the command line), and resolve your issues, but I cannot guarantee it.

    Most users are happy to stay on 2.3.1.5, at least until Elastix comes out with it as the default install.

    Others may have some input into this??

    Regards

    Bob
     
  8. telecomtechnician

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    Try this

    hello

    Create 2 sip extensions through freepbx but register two softphones, and try to dial between them. If you have communication (ringing and voice) that means there is a bad configuration on all the phones or on the csv file for creating the extensions.

    If the problem persists, as I see it there should be a major problem with the elastix installation or something in your lan that is blocking the registrations of the hardphones. This should not be a major issue, because this configuration is seamless and easy.

    Sometimes we forget the basics, first crawl, then walk, and later we can run.
     
  9. gbrook

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    Re:Try this

    OK

    I did a complete reinstall.

    Added the sip_nat.conf file and put the local ip details in.

    Did absolutely nothing else.

    Programmed up 3 extensions and they worked.

    Upgraded the FreePBX Modules to 2.4 from within FreePBX "Module Admin" - back to the same problem - phones can't ring each other.

    So why has no-one else run into this fairly basic problem - very frustrating. This would seem to me to be very fundamental!

    How do you go about doing it with scripts (updating FreePBX that is) instead of through the GUI?

    Cheers
    Garry<br><br>Post edited by: gbrook, at: 2008/05/22 16:20
     
  10. Bob

    Bob

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    Re:Try this

    TCTechnician,

    I appreciate that you are trying to help, but he has already stated that the phones are registered. He has also stated that he has done thirty installs, so I suspect that he has done the crawling bit...

    The problem he is describing occurs on several occasions, one which is when 2.4 has been installed on Elastix, this one I can vouch for. Another is when the Extensions batch is used (which requires going into each extension and pressing submit).

    There are also a few others that I can recall, in particular with the AAstra phones, but normally they show the extension state of something other than 0. If you look at his log, none of the CF's are on, and the extension state is 0....Another time this has occurred is when trying to do restore, in particular a cross system restore (e.g. Freepbx on Elastix/Freepbx on trixbox).

    I strongly suggest that the 2.4 GUI upgrade has impacted the system.

    Again, not trying to knock you for replying, but read his postings....
     
  11. Bob

    Bob

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    Re:Try this

    As per the freepbx upgrade page

    cd /usr/src/
    wget http://mirror.freepbx.org/freepbx-2.4.0.tar.gz
    tar zxvf freepbx-2.4.0.tar.gz
    cd freepbx-2.4.0
    ./start_asterisk start # for upgrades use: ./amportal start
    ./install_amp

    As I mentioned, I have not tried this for a couple of months now, as I generally do not need to go to 2.4, but have used this upgrade path for a year or so now without any issues....

    Regards

    Bob
     
  12. gbrook

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    Re:Try this

    Thanks for that Bob

    Should have checked there myself for that info.

    Seems to be a pretty basic issue though and noone else seems to run in to it on the forum

    Cheers
    Garry<br><br>Post edited by: gbrook, at: 2008/05/22 17:46
     
  13. gbrook

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    Re:Try this

    Does exactly the same thing using the script - once done no calls possible.

    This is a big showstopper for not allowing me to use Elastix - shame, the rest of it seems a step forward from as far as I got with it.

    Garry
     
  14. Bob

    Bob

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    Re:Try this

    Garry,

    Just one quick question. Is there anything in 2.4 that you really need???

    In general for most people, there is nothing that is really needed in 2.4 for most users.

    Freepbx 2.4 was a change in some of the directory structure plus a few extra features. In general there is not huge improvement in the Macros that handle the calls, and 2.3 continues to get those macro changes.

    This is not to say that 2.4 is not going to happen on Elastix, it needs to the developers to look at what Freepbx has done, and translate. They have spent more time on getting 1.0 out and improving the integration functionality (which I assuming that you may be referrng to, as a "step forward".

    The main issue is that many of the parts, are inter-related, such as conferencing, call centre, billing, op panel.

    All systems that we implement work perfectly well with 2.3.1.5, with the only advantage is the DID handling of PSTN lines in 2.4 (which feels better than writing individual Zap channels)....

    Regards

    Bob
     
  15. gbrook

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    Re:Try this

    Hi Bob,

    I have thought about it and you are probably right in my real need fro 2.4 but I always try to have systems current at time of installation and it seems rather ironic that I am looking at Elastix because I am unhappy with trixbox and the "perceived" treatment of FreePBX (and I am having trouble with FreePBX in this install, rather ironic).

    I was looking at moving to Elastix with new systems and the one I am currently looking at is a change out from trixbox with FreePBX 2.4 installed.

    It does concern me when there is no warnings about not being able to upgrade FreepBX to 2.4 let alone breaking your system completely and what else there is that will break it so badly.

    I do have ZAP PSTN channels on this system using a Rhino Card - is 2.3 much different in the DID handling?

    I will need to think long and hard as to the philosophy behind this distro.

    Cheers
    Garry<br><br>Post edited by: gbrook, at: 2008/05/22 18:12
     
  16. Bob

    Bob

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    Re:Try this

    Garry,

    I agree with you, it is always nice to know that the latest is installed. As mentioned, the 2.4 was a change to the underlying system, and may not occur again until 2.5, maybe 2.6 etc...

    It is possible that the fix for it may be a couple of lines, in fact I know BenS and Cleve were working on getting it sorted, and I suspect it was just copying a couple of files (Symlinks), which may correct your issue.

    If you have a little bit of time, I was going to look at Freepbx on the weekend, for a couple of issues, and will try and return with some news.

    I will be honest and say that I am not sure how the rhino is treated in Freepbx, but the main difference is that in 2.4 you could assign a DID number to each PSTN line (as PSTN lines in general do not provide DID information like a ISDN/E1/T1 line). This move is to make it easier to maintain one set of Inbound flow, instead of having one way for ISDN/E1/T1 and another for PSTN Channels. It makes good sense and I like the thinking.

    Freepbx 2.3 - Historically and for laziness, PSTN was quite often setup using a group G0 or G1 etc...meaning that you setup an inbound route based on a Group, further meaning that you basically did the same thing without regard to which line was actually ringing. Alternatively, you could assign each of your PSTN lines individually e.g. ZAP/1 or ZAP/2, and base the inbounds on what line they were coming in on. this can become messy, on what line was what, not a big issue, but an issue.

    So basically that is the difference in the PSTN/ZAP implementation.

    Last bit, have a look around, I do believe others may have posted about the 2.4 upgrade, and I have a suspicion that it is just copying and symlinking a couple of files, in particular, I remember an extensions.conf in the mix somewhere....

    Hope this helps....

    Regards

    Bob
    P.S. The direction and philosophy of Elastix is in my opinion heading in the right way. It is by no means perfect, it doesn't have the dollars behind it like Trixbox, and they are also not restricting certain parts, but it does have a good development team, new ideas, a better business focus e.g. real business tools....but you can only form your own opinion....<br><br>Post edited by: Bob, at: 2008/05/22 18:36
     
  17. gbrook

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    Re:Try this

    Hi Bob,

    Thanks for the info, I will try to delay the install and see if anything surfaces in the next few days.

    You put it very well.

    I do very much appreciate that efforts that people go to with Elastix, PIAF abd Trixbox to name a couple but when what you think are fairly basic things break it is very frustating.

    Cheers
    Garry
     
  18. roque.m

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  19. Bob

    Bob

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    Re:Try this

    That was what I remembered someone posted, just couldn't locate it.

    However, I think that there was a symlink that had to be corrected as referenced in the post.

    Roque.m, did you perform this Symlink in the end??

    Regards

    Bob
     
  20. gbrook

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    Re:Try this

    Yeah!!

    Thanks for that now works

    Cheers
    Garry
     

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