TIPS XiVO RasPi Growing Pains

[SMTC]

AGETTY "nothing to worry about " issue - is this possibly related to the fact that BELL CANADA's BIZ ADSL line here is flapping up and down like a toilet seat this week and I get a new Public IP address each time?

UPDATE: Bell seems to have quietly fixed the faulty DSLAM issues. Stable once again, Warning to Bell subscribers. Getting Business grade of service is actually poorer service than residential. The Biz techs don't work week-ends and holidays on low-end product issues. (but the residential installers do. Go figure). Bell has never met their service level objective with any tickets I have ever opened up in the Muskoka's.

 

[SMTC]

Xivo looks a lot more complicated than PIAF/Incredible PBX 13.2. Finally licked the no audio on my softphone Zoiper app which turned out to be missing mask syntax on Local LAN. (Pop up help not there for those of us that find it not obvious). Also, I have not stumbled on any indication of any of the Incredible PBX features in the Xivo browser panels? No conferences, no Lenny extension, no IVR demo, no other *-code features. Using the Sept 2 img.

Lenny answers but does not seem to hear me very well. Looking at the Asterisk CLI verbose, it appears Lenny is not local but is on rentpbx.mundy.org - say what???? Are my eyes seeing this correct?

Don't see any way to have IAX user extensions either. I commonly had an IAX Zoiper extension for when SIP was getting blocked at certain my traveled destinations.

I love to experiment with the new but I find myself pining for my old PIAF server - which was working at least (while I fumble into Xivo).

 

[wardmundy]

Xivo looks a lot more complicated than PIAF/Incredible PBX 13.2. Finally licked the no audio on my softphone Zoiper app which turned out to be missing mask syntax on Local LAN. (Pop up help not there for those of us that find it not obvious). Also, I have not stumbled on any indication of any of the Incredible PBX features in the Xivo browser panels? No conferences, no Lenny extension, no IVR demo, no other *-code features. Using the Sept 2 img.

All your favorite features are still there. They're just in a different place but not complicated at all. Take a look at the main tutorial. It covers just about everything.

 

[SMTC]

All your favorite features are still there. They're just in a different place but not complicated at all. Take a look at the main tutorial. It covers just about everything.

The tutorial makes reference to installing Incredible PBX with a script. I was under the impression the PI 3 Xivo build pre-installed IncrediblePBX features? Re-running the script probably will break something, right?

Also, have extensions working internal/external and can talk to *98 voicemail but CONF (2663) fails - no sound, no call, just tries and dies.

Also, also - so far unable to get FreePhoneLine SIP working. Appears to be registered (CLI, sip show registry) but no luck inbound/outbound routes. I seem to be really flying blind with Xivo. The Xivo portal seems towhere be devoid of Asterisk info showing registration status and I don't see any evidence that the apparently working Incredible PBX feature set is there. Am I blind?

Also, also, also - WiFi drops and then never tries to reconnect. Have to reboot each time. Moving PI 3 closer to the Router helps it stay up, but why doesn't it reconnect automatically?

I am certainly not clueless with this stuff but am thinking I am nearly out of patience with this Xivo platform and will need to revert to the old platform unless some major light starts to appear at the end of the tunnel...:-(

 

[SMTC]

More regarding the Conference room (2663 & 2664) fails. Asterisk CLI log:

root@xivo:~# asterisk -r
[Sep 12 15:38:50] Asterisk 13.10.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.
[Sep 12 15:38:50] Created by Mark Spencer <[email protected]>
[Sep 12 15:38:50] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
[Sep 12 15:38:50] This is free software, with components licensed under the GNU General Public
[Sep 12 15:38:50] License version 2 and other licenses; you are welcome to redistribute it under
[Sep 12 15:38:50] certain conditions. Type 'core show license' for details.
[Sep 12 15:38:50] =========================================================================
[Sep 12 15:38:50] This package has been modified for the Debian GNU/Linux distribution
[Sep 12 15:38:50] This package has been modified for XiVO, a product of Avencall
[Sep 12 15:38:50] Please report all bugs to https://projects.xivo.io/projects/xivo/issues
[Sep 12 15:38:50] =========================================================================
[Sep 12 15:38:50] Connected to Asterisk 13.10.0 currently running on xivo (pid = 4387)
[Sep 12 15:38:53] == Using SIP RTP CoS mark 5
[Sep 12 15:38:53] -- Executing [2664@default:1] Set("SIP/0in5u7tb-00000004", "XIVO_BASE_CONTEXT=default") in new stack
[Sep 12 15:38:53] -- Executing [2664@default:2] Set("SIP/0in5u7tb-00000004", "XIVO_BASE_EXTEN=2664") in new stack
[Sep 12 15:38:53] -- Executing [2664@default:3] Gosub("SIP/0in5u7tb-00000004", "meetme,s,1(2,)") in new stack
[Sep 12 15:38:53] -- Executing [s@meetme:1] Set("SIP/0in5u7tb-00000004", "XIVO_DSTID=2") in new stack
[Sep 12 15:38:53] -- Executing [s@meetme:2] Set("SIP/0in5u7tb-00000004", "XIVO_PRESUBR_GLOBAL_NAME=MEETME") in new stack
[Sep 12 15:38:53] -- Executing [s@meetme:3] UserEvent("SIP/0in5u7tb-00000004", "Meetme,CHANNEL: SIP/0in5u7tb-00000004,XIVO_USERID: 2,XIVO_DSTID: 2") in new stack
[Sep 12 15:38:53] -- Executing [s@meetme:4] Set("SIP/0in5u7tb-00000004", "XIVO_SRCNUM=750") in new stack
[Sep 12 15:38:53] -- Executing [s@meetme:5] Set("SIP/0in5u7tb-00000004", "XIVO_DSTNUM=2664") in new stack
[Sep 12 15:38:53] -- Executing [s@meetme:6] Set("SIP/0in5u7tb-00000004", "XIVO_CONTEXT=default") in new stack
[Sep 12 15:38:53] -- Executing [s@meetme:7] AGI("SIP/0in5u7tb-00000004", "agi://127.0.0.1/incoming_meetme_set_features") in new stack
[Sep 12 15:38:53] -- AGI Script Executing Application: (MeetMeCount) Options: (2664,MEETMECOUNT)
[Sep 12 15:38:53] == Parsing '/etc/asterisk/meetme.conf': Found
[Sep 12 15:38:53] == Parsing '/var/tmp/exec.1473709133338653.1980511264': Found
[Sep 12 15:38:53] agi://127.0.0.1/incoming_meetme_set_features: AGI handler 'incoming_meetme_set_features' successfully executed
[Sep 12 15:38:53] -- <SIP/0in5u7tb-00000004>AGI Script agi://127.0.0.1/incoming_meetme_set_features completed, returning 0
[Sep 12 15:38:53] -- Executing [s@meetme:8] Gosub("SIP/0in5u7tb-00000004", "xivo-subroutine,s,1()") in new stack
[Sep 12 15:38:53] -- Executing [s@xivo-subroutine:1] GotoIf("SIP/0in5u7tb-00000004", "?:nosubroutine") in new stack
[Sep 12 15:38:53] -- Goto (xivo-subroutine,s,4)
[Sep 12 15:38:53] -- Executing [s@xivo-subroutine:4] Return("SIP/0in5u7tb-00000004", "") in new stack
[Sep 12 15:38:53] -- Executing [s@meetme:9] Gosub("SIP/0in5u7tb-00000004", "xivo-pickup,0,1") in new stack
[Sep 12 15:38:53] -- Executing [0@xivo-pickup:1] Goto("SIP/0in5u7tb-00000004", "s,1") in new stack
[Sep 12 15:38:53] -- Goto (xivo-pickup,s,1)
[Sep 12 15:38:53] -- Executing [s@xivo-pickup:1] Set("SIP/0in5u7tb-00000004", "WAITSEC=1") in new stack
[Sep 12 15:38:53] -- Executing [s@xivo-pickup:2] Goto("SIP/0in5u7tb-00000004", "pickup,1") in new stack
[Sep 12 15:38:53] -- Goto (xivo-pickup,pickup,1)
[Sep 12 15:38:53] -- Executing [pickup@xivo-pickup:1] GotoIf("SIP/0in5u7tb-00000004", "?return") in new stack
[Sep 12 15:38:53] -- Executing [pickup@xivo-pickup:2] Answer("SIP/0in5u7tb-00000004", "") in new stack
[Sep 12 15:38:53] > 0x75b098f0 -- Probation passed - setting RTP source address to 192.168.1.100:41180
[Sep 12 15:38:53] -- Executing [pickup@xivo-pickup:3] Wait("SIP/0in5u7tb-00000004", "1") in new stack
[Sep 12 15:38:54] -- Executing [pickup@xivo-pickup:4] Set("SIP/0in5u7tb-00000004", "XIVO_PICKEDUP=1") in new stack
[Sep 12 15:38:54] -- Executing [pickup@xivo-pickup:5] Return("SIP/0in5u7tb-00000004", "") in new stack
[Sep 12 15:38:54] -- Executing [s@meetme:10] Gosub("SIP/0in5u7tb-00000004", "xivo-global-subroutine,s,1") in new stack
[Sep 12 15:38:54] -- Executing [s@xivo-global-subroutine:1] GotoIf("SIP/0in5u7tb-00000004", "1?:return") in new stack
[Sep 12 15:38:54] -- Executing [s@xivo-global-subroutine:2] GotoIf("SIP/0in5u7tb-00000004", "MEETME?:return") in new stack
[Sep 12 15:38:54] -- Executing [s@xivo-global-subroutine:3] GotoIf("SIP/0in5u7tb-00000004", "xivo-subrgbl-meetme?:return") in new stack
[Sep 12 15:38:54] -- Executing [s@xivo-global-subroutine:4] GotoIf("SIP/0in5u7tb-00000004", "0?:return") in new stack
[Sep 12 15:38:54] -- Goto (xivo-global-subroutine,s,6)
[Sep 12 15:38:54] -- Executing [s@xivo-global-subroutine:6] Return("SIP/0in5u7tb-00000004", "") in new stack
[Sep 12 15:38:54] -- Executing [s@meetme:11] Gosub("SIP/0in5u7tb-00000004", "originate-caller-id,s,1") in new stack
[Sep 12 15:38:54] -- Executing [s@originate-caller-id:1] GotoIf("SIP/0in5u7tb-00000004", "0?:name") in new stack
[Sep 12 15:38:54] -- Goto (originate-caller-id,s,3)
[Sep 12 15:38:54] -- Executing [s@originate-caller-id:3] GotoIf("SIP/0in5u7tb-00000004", "0?:fix") in new stack
[Sep 12 15:38:54] -- Goto (originate-caller-id,s,5)
[Sep 12 15:38:54] -- Executing [s@originate-caller-id:5] GotoIf("SIP/0in5u7tb-00000004", "?:end") in new stack
[Sep 12 15:38:54] -- Goto (originate-caller-id,s,8)
[Sep 12 15:38:54] -- Executing [s@originate-caller-id:8] Return("SIP/0in5u7tb-00000004", "") in new stack
[Sep 12 15:38:54] -- Executing [s@meetme:12] MeetMe("SIP/0in5u7tb-00000004", "2664,icM") in new stack
[Sep 12 15:38:54] == Parsing '/etc/asterisk/meetme.conf': Found
[Sep 12 15:38:55] == Parsing '/var/tmp/exec.1473709134972258.1980511264': Found
[Sep 12 15:38:55] == Spawn extension (meetme, s, 12) exited non-zero on 'SIP/0in5u7tb-00000004'
[Sep 12 15:38:55] -- SIP/0in5u7tb-00000004 Internal Gosub(hangup_handlers,userevent,1) start
[Sep 12 15:38:55] -- Executing [userevent@hangup_handlers:1] NoOp("SIP/0in5u7tb-00000004", "Sending Hangup userevent") in new stack
[Sep 12 15:38:55] -- Executing [userevent@hangup_handlers:2] UserEvent("SIP/0in5u7tb-00000004", "Hangup,XIVO_USERUUID: f7998943-5b4e-4392-a35d-30ed365e6df2") in new stack
[Sep 12 15:38:55] -- Executing [userevent@hangup_handlers:3] Return("SIP/0in5u7tb-00000004", "") in new stack
[Sep 12 15:38:55] == Spawn extension (meetme, s, 12) exited non-zero on 'SIP/0in5u7tb-00000004'
[Sep 12 15:38:55] -- SIP/0in5u7tb-00000004 Internal Gosub(hangup_handlers,userevent,1) complete GOSUB_RETVAL=
xivo*CLI>

 

[wardmundy]

Don't reinstall anything. And, yes, there is a learning curve, and it's more than a day.

I'll take a look at CONF. It's supposed to be 2663 NOT 2664. You can't use MeetMe conferences because there's no timing device on the RasPi.

There are several other threads about SIP registrations with XiVO. Take a look at those to diagnose what's happening.

WiFi sounds like nothing more than a very poor connection. If it's dropping, then the router probably falls off the radar completely.

 

[SMTC]

Don't reinstall anything. And, yes, there is a learning curve, and it's more than a day.

I'll take a look at CONF. It's supposed to be 2663 NOT 2664. You can't use MeetMe conferences because there's no timing device on the RasPi.

There are several other threads about SIP registrations with XiVO. Take a look at those to diagnose what's happening.

WiFi sounds like nothing more than a very poor connection. If it's dropping, then the router probably falls off the radar completely.

So 2663 is created as part of the distro? As in don't try and create a Conf bridge? No timing device, no meet-me - oh-oh. Bummer.

Raspian command: IWCONFIG WLAN0 yields a 37/70 signal quality in the improved spot and its holding, but I also notice power management a being ON. Thought I saw something about needing to be patch it OFF.

 

[SMTC]

I deleted the 2663 and 2664 Conference rooms I had created. Now a call to 2663 plays music on hold. What kind of conf bridge is this hidden in there without any GUI access point? No PIN, no Moderator PIN?

Was not aware there was a timing device issue with the Pi3. I now have a Pi2 and a Pi3 that aren't fitting the desired spec...

...still don't have a working SIP trunk installed..

 

[wardmundy]

You can adjust the conference bridge setup in any way you like. We've just set up a simple one to get folks started. As more people start using it, we'll add features.

There's nothing difficult about it. Syntax is documented here. Lots of examples here. Also take a look at the Conference Bridge documentation and /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf:

;# // BEGIN Conf 2663
exten=2663,1,Answer
same=n,Set(CONFBRIDGE(user,music_on_hold_when_empty)=yes)
same=n,Set(CONFBRIDGE(user,music_on_hold_class)=default)
same=n,ConfBridge(2663)
same=n,Hangup
;# // END Conf 2663

p.s. Lack of a timing source in Raspberry Pi devices is nothing new.