Ofloo

Forum Replies Created

Viewing 15 replies - 1 through 15 (of 37 total)
  • Replies
  • Ofloo
    Participant
    Up
    0
    Down

    Indeed dtmf mode was wrong on the phone it was rfc2833 + sip info and on the server it was rfc2833, changing them to match each made it work.

    0
    Ofloo
    Participant
    Up
    0
    Down

    @ing-joserivera26

    euhm it’s not a soft phone it’s a yealink t41p

    0
    Ofloo
    Participant
    Up
    0
    Down

    Found out what the problem was, .. seems like tls setup had a bad certificate removed the certificate, regenerated it, .. and it stopped restarting and coring.

    0
    Ofloo
    Participant
    Up
    0
    Down

    Just thought i’d post so other people can take advantage of it. I’m glad I found it to ? 

    0
    Ofloo
    Participant
    Up
    0
    Down

    I finally figured out what was wrong, ..

    and this seems to be the case for all betamax voip providers, ..

    don’t set the fromuser to anything just leave it empty, and caller id works ..!?

    0
    Ofloo
    Participant
    Up
    0
    Down

    I think I know what was wrong, the did was replaced by the trunk I’d replacing it with the did with to, fixed it I think at least getting ring tone.

     

    Still need to test further.

    0
    Ofloo
    Participant
    Up
    0
    Down

    Yes

    0
    Ofloo
    Participant
    Up
    0
    Down
    # asterisk -rvvv
    Asterisk 13.23.1, Copyright (C) 1999 - 2014, Digium, Inc. and others.
    Created by Mark Spencer <markster@digium.com>
    Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
    This is free software, with components licensed under the GNU General Public
    License version 2 and other licenses; you are welcome to redistribute it under
    certain conditions. Type 'core show license' for details.
    =========================================================================
    Connected to Asterisk 13.23.1 currently running on pbx (pid = 2940)
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    -- Executing [SIPIDSRV@trk-6-in:1] NoOp("SIP/SIPIDSRV-00000045", "Incoming call through: PROVIDERNAME") in new stack
    -- Executing [SIPIDSRV@trk-6-in:2] Set("SIP/SIPIDSRV-00000045", "DID=SIPIDSRV") in new stack
    -- Executing [SIPIDSRV@trk-6-in:3] Goto("SIP/SIPIDSRV-00000045", "default-trunk,SIPIDSRV,1") in new stack
    -- Goto (default-trunk,SIPIDSRV,1)
    -- Executing [SIPIDSRV@default-trunk:1] Gosub("SIP/SIPIDSRV-00000045", "sub-check-blacklist,s,1(2a4e48aa120a8d8c,SOURCEPHONE)") in new stack
    -- Executing [s@sub-check-blacklist:1] NoOp("SIP/SIPIDSRV-00000045", "Testing if SOURCEPHONE is in Black List") in new stack
    -- Executing [s@sub-check-blacklist:2] GotoIf("SIP/SIPIDSRV-00000045", "0?banned") in new stack
    -- Executing [s@sub-check-blacklist:3] Return("SIP/SIPIDSRV-00000045", "") in new stack
    -- Executing [SIPIDSRV@default-trunk:2] Gosub("SIP/SIPIDSRV-00000045", "sub-setup-call-type,s,1(incoming)") in new stack
    -- Executing [s@sub-setup-call-type:1] NoOp("SIP/SIPIDSRV-00000045", "Determinating Call Type") in new stack
    -- Executing [s@sub-setup-call-type:2] GotoIf("SIP/SIPIDSRV-00000045", "0?return") in new stack
    -- Executing [s@sub-setup-call-type:3] Gosub("SIP/SIPIDSRV-00000045", "s-incoming,1()") in new stack
    -- Executing [s-incoming@sub-setup-call-type:1] NoOp("SIP/SIPIDSRV-00000045", "Incoming Call") in new stack
    -- Executing [s-incoming@sub-setup-call-type:2] Set("SIP/SIPIDSRV-00000045", "__CALL_TYPE=2") in new stack
    -- Executing [s-incoming@sub-setup-call-type:3] Return("SIP/SIPIDSRV-00000045", "") in new stack
    -- Executing [s@sub-setup-call-type:4] Set("SIP/SIPIDSRV-00000045", "__CALL_TYPE_CONFIGURED=yes") in new stack
    -- Executing [s@sub-setup-call-type:5] Set("SIP/SIPIDSRV-00000045", "CDR(calltype)=2") in new stack
    -- Executing [s@sub-setup-call-type:6] Return("SIP/SIPIDSRV-00000045", "") in new stack
    -- Executing [SIPIDSRV@default-trunk:3] Gosub("SIP/SIPIDSRV-00000045", "dynamic-routing-in,s,1(SOURCEPHONE)") in new stack
    -- Executing [s@dynamic-routing-in:1] NoOp("SIP/SIPIDSRV-00000045", "Test if must to apply dynamic routing") in new stack
    -- Executing [s@dynamic-routing-in:2] Set("SIP/SIPIDSRV-00000045", "EXTERNAL_CALLER=SOURCEPHONE") in new stack
    -- Executing [s@dynamic-routing-in:3] GotoIf("SIP/SIPIDSRV-00000045", "1?gd") in new stack
    -- Goto (dynamic-routing-in,s,5)
    -- Executing [s@dynamic-routing-in:5] GotoIf("SIP/SIPIDSRV-00000045", "0?:rb") in new stack
    -- Goto (dynamic-routing-in,s,10)
    -- Executing [s@dynamic-routing-in:10] Return("SIP/SIPIDSRV-00000045", "") in new stack
    -- Executing [SIPIDSRV@default-trunk:4] Goto("SIP/SIPIDSRV-00000045", "incoming-calls,SIPIDSRV,1") in new stack
    -- Goto (incoming-calls,SIPIDSRV,1)
    -- Channel 'SIP/SIPIDSRV-00000045' sent to invalid extension: context,exten,priority=incoming-calls,SIPIDSRV,1
    -- Executing [i@incoming-calls:1] NoCDR("SIP/SIPIDSRV-00000045", "") in new stack
    -- Executing [i@incoming-calls:2] Goto("SIP/SIPIDSRV-00000045", "invalid-dest,s,1") in new stack
    -- Goto (invalid-dest,s,1)
    -- Executing [s@invalid-dest:1] NoOp("SIP/SIPIDSRV-00000045", "Invalid Route to Dial") in new stack
    -- Executing [s@invalid-dest:2] Playback("SIP/SIPIDSRV-00000045", "im-sorry&no-route-exists-to-dest&vm-goodbye") in new stack
    -- <SIP/SIPIDSRV-00000045> Playing 'im-sorry.ulaw' (language 'en')
    -- <SIP/SIPIDSRV-00000045> Playing 'no-route-exists-to-dest.ulaw' (language 'en')
    -- <SIP/SIPIDSRV-00000045> Playing 'vm-goodbye.ulaw' (language 'en')
    0
    Ofloo
    Participant
    Up
    0
    Down

    I’ll look into it.

    0
    Ofloo
    Participant
    Up
    0
    Down

    Yeah i thought that too but then I reran the elastix vm and it just worked. So if if elastix was able to overwrite/set the Caller ID why can’t vitalpbx.

    0
    Ofloo
    Participant
    Up
    0
    Down

    Extension => Call to External Number

    0
    Ofloo
    Participant
    Up
    0
    Down

    Is there an ability to forward the added extensions? I mean now I used speed dial, but then i won’t be able to add them as emergency numbers, .. so .. how would I add them as emergency numbers and yet have a speed dial like feature.

    I mean I can’t remember 003290342704 112 is a lot easier.

    0
    Ofloo
    Participant
    Up
    0
    Down

    No CID overwrite isn’t active tried with and without, doesn’t matter, .. this issue is been going on for a while.

    I used to run elastix and there it always worked fine but for this trunk I wasn’t able to set it right for some reason, not on ombutel nor on vitalpbx. I’m not sure why that is though.

    I’m sure its some configuration, that needs to be done but can’t put my finger on what it is.

    *not sure but I believe it did work in one of the first releases of ombutel, .. have been using it since it released version 1 or something, I did make a post about it, but  no one ever really to responded to it.

    It’s a betamax sip provider, .. the numbers have been verified, .. I can set them statically and if I would have one number it wouldn’t matter, .. but that’s not the case.

    0
    Ofloo
    Participant
    Up
    0
    Down

    The reason i had to reinstall vitalpbx to make it work was due to the fact that asterisk wasn’t running, I noticed after the upgrade that asterisk doesn’t automatically starts up, so ..

    systemctl enable asterisk.service
    0
    Ofloo
    Participant
    Up
    0
    Down

    After upgrade i noticed

    [2019-01-28 09:01:47] ERROR[6979] config.c: The file ‘ombutel/http__*.conf’ was listed as a #include but it does not exist.

    0
Viewing 15 replies - 1 through 15 (of 37 total)