Hi all,
Have anyone implemented freepbx 2.2.0 on Asterisk 1.4.0? If yes, what and what step is needed, as there are lot of changes in the Asterisk 1.4.0. Do we just fellow the same procedures as with 1.2.14 and the likes?
Please advice?
Cheers to all
[quote=“p_lindheimer”]schreyack,
as I mentioned over on the trixbox forum, SLA that was hacked together for 1.4.1 is not what described that you need. It is aimed 100% at implementing a key system - not at all concerned with Shared Extensions.
philippe[/quote]
ah… i didn’t quite understand that from your post over there.
Just checking if there is any info on when/if FreePBX will support Asterisk 1.4.1…
I’m looking for the SLA feature, which is supposed to work properly in 1.4.1.
Tim
schreyack,
as I mentioned over on the trixbox forum, SLA that was hacked together for 1.4.1 is not what described that you need. It is aimed 100% at implementing a key system - not at all concerned with Shared Extensions.
philippe
ditto, now trying to resurrect my “old system” I have the source trees for 1.2.x of asterisk, zaptel, and asterisk-addons, and have built them.
asterisk -cvr
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
discussion around the net says to tail /var/log/asterisk/full to see why… THe last entry in there was a day or so ago, and I have rebooted several times since. which seems strange.
It would appear that asterisk is being started…
[root@pqpbx log]# ps -ax | grep asterisk
Warning: bad syntax, perhaps a bogus ‘-’? See /usr/share/doc/procps-3.2.3/FAQ
2699 ? S 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk
2700 ? Sl 0:02 /usr/sbin/asterisk -U asterisk -G asterisk -v -g -p -U asterisk -G asterisk
6131 ? Rs 0:00 [asterisk_channe]
Now, if I kill those two process and amportal start to restart. It appears to start, but as you can see at the end of this, it’s not talking to my ata…
any ideas…
[root@pqpbx log]# amportal start
SETTING FILE PERMISSIONS
Permissions OK
Asterisk Event Logger Started /var/log/asterisk/event_log
Asterisk Dynamic Loader loading preload modules:
Mar 20 16:25:08 NOTICE[6464]: cdr.c:1193 do_reload: CDR simple logging enabled.
Asterisk PBX Core Initializing
Registering builtin applications:
[AbsoluteTimeout]
[Answer]
[BackGround]
[Busy]
[Congestion]
[DigitTimeout]
[Goto]
[GotoIf]
[GotoIfTime]
[ExecIfTime]
[Hangup]
[NoOp]
[Progress]
[ResetCDR]
[ResponseTimeout]
[Ringing]
[SayNumber]
[SayDigits]
[SayAlpha]
[SayPhonetic]
[SetAccount]
[SetAMAFlags]
[SetGlobalVar]
[SetLanguage]
[Set]
[SetVar]
[ImportVar]
[Wait]
[WaitExten]
Asterisk Dynamic Loader Starting:
[res_musiconhold.so] => (Music On Hold Resource)
[res_adsi.so] => (ADSI Resource)
[res_agi.so] => (Asterisk Gateway Interface (AGI))
[res_crypto.so] => (Cryptographic Digital Signatures)
[res_indications.so] => (Indications Configuration)
[res_monitor.so] => (Call Monitoring Resource)
[res_features.so] => (Call Features Resource)
[pbx_config.so] => (Text Extension Configuration)
[pbx_dundi.so] => (Distributed Universal Number Discovery (DUNDi))
[pbx_spool.so] => (Outgoing Spool Support)
[pbx_realtime.so] => (Realtime Switch)
[pbx_loopback.so] => (Loopback Switch)
[pbx_functions.so] => (Builtin dialplan functions)
[pbx_ael.so] => (Asterisk Extension Language Compiler)
Mar 20 16:25:08 WARNING[6464]: pbx.c:6471 ast_context_verify_includes: Context ‘ael-local’ tries includes nonexistent context 'ael-default’
Mar 20 16:25:08 WARNING[6464]: pbx.c:6471 ast_context_verify_includes: Context ‘ael-local’ tries includes nonexistent context ‘ael-parkedcalls’
[chan_phone.so] => (Linux Telephony API Support)
[chan_features.so] => (Feature Proxy Channel)
[chan_oss.so] => (OSS Console Channel Driver)
[chan_agent.so] => (Agent Proxy Channel)
[chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
[chan_sip.so] => (Session Initiation Protocol (SIP))
[chan_zap.so] => (Zapata Telephony w/PRI)
[chan_local.so] => (Local Proxy Channel)
[chan_mgcp.so] => (Media Gateway Control Protocol (MGCP))
[chan_skinny.so] => (Skinny Client Control Protocol (Skinny))
[app_voicemail.so] => (Comedian Mail (Voicemail System))
[app_lookupblacklist.so] => (Look up Caller*ID name/number from blacklist database)
[app_curl.so] => (Load external URL)
[app_meetme.so] => (MeetMe conference bridge)
[codec_zap.so] => (Generic Zaptel Transcoder Codec Translator)
[app_groupcount.so] => (Group Management Routines)
[app_forkcdr.so] => (Fork The CDR into 2 separate entities.)
[app_realtime.so] => (Realtime Data Lookup/Rewrite)
[app_while.so] => (While Loops and Conditional Execution)
[app_directory.so] => (Extension Directory)
[app_authenticate.so] => (Authentication Application)
[app_ices.so] => (Encode and Stream via icecast and ices)
[app_readfile.so] => (Stores output of file into a variable)
[app_sendtext.so] => (Send Text Applications)
[app_parkandannounce.so] => (Call Parking and Announce Application)
[app_dictate.so] => (Virtual Dictation Machine)
[app_enumlookup.so] => (ENUM Lookup)
[app_txtcidname.so] => (TXTCIDName)
[format_g729.so] => (Raw G729 data)
[app_setcdruserfield.so] => (CDR user field apps)
[app_dial.so] => (Dialing Application)
[app_waitforsilence.so] => (Wait For Silence)
[app_image.so] => (Image Transmission Application)
[codec_ulaw.so] => (Mu-law Coder/Decoder)
[format_h263.so] => (Raw h263 data)
[format_g726.so] => (Raw G.726 (16/24/32/40kbps) data)
[app_page.so] => (Page Multiple Phones)
[codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator)
[app_dumpchan.so] => (Dump Info About The Calling Channel)
[cdr_manager.so] => (Asterisk Call Manager CDR Backend)
[app_system.so] => (Generic System() application)
[func_callerid.so] => (Caller ID related dialplan function)
[codec_adpcm.so] => (Adaptive Differential PCM Coder/Decoder)
[app_zapateller.so] => (Block Telemarketers with Special Information Tone)
[app_zapbarge.so] => (Barge in on Zap channel application)
[app_eval.so] => (Reevaluates strings)
[codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator)
[format_ilbc.so] => (Raw iLBC data)
[codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder)
[format_au.so] => (Sun Microsystems AU format (signed linear))
[app_flash.so] => (Flash zap trunk application)
[app_disa.so] => (DISA (Direct Inward System Access) Application)
[app_echo.so] => (Simple Echo Application)
[app_controlplayback.so] => (Control Playback Application)
[app_macro.so] => (Extension Macros)
[func_uri.so] => (URI encode/decode functions)
[app_talkdetect.so] => (Playback with Talk Detection)
[app_queue.so] => (True Call Queueing)
[app_nbscat.so] => (Silly NBS Stream Application)
[format_vox.so] => (Dialogic VOX (ADPCM) File Format)
[app_chanisavail.so] => (Check channel availability)
[app_zapras.so] => (Zap RAS Application)
[format_gsm.so] => (Raw GSM data)
[app_waitforring.so] => (Waits until first ring after time)
[app_setrdnis.so] => (Set RDNIS Number)
[app_alarmreceiver.so] => (Alarm Receiver for Asterisk)
[app_md5.so] => (MD5 checksum applications)
[app_read.so] => (Read Variable Application)
[format_pcm_alaw.so] => (Raw aLaw 8khz PCM Audio support)
[app_getcpeid.so] => (Get ADSI CPE ID)
[cdr_csv.so] => (Comma Separated Values CDR Backend)
[app_cdr.so] => (Tell Asterisk to not maintain a CDR for the current call)
[app_record.so] => (Trivial Record Application)
[app_adsiprog.so] => (Asterisk ADSI Programming Application)
[app_mixmonitor.so] => (Mixed Audio Monitoring Application)
[app_setcidnum.so] => (Set CallerID Number)
[app_chanspy.so] => (Listen to the audio of an active channel)
[app_setcidname.so] => (Set CallerID Name)
[app_sayunixtime.so] => (Say time)
[app_hasnewvoicemail.so] => (Indicator for whether a voice mailbox has messages in a given folder.)
[app_privacy.so] => (Require phone number to be entered, if no CallerID sent)
[codec_a_mu.so] => (A-law and Mulaw direct Coder/Decoder)
[app_exec.so] => (Executes applications)
[app_directed_pickup.so] => (Directed Call Pickup Application)
[app_playback.so] => (Sound File Playback Application)
[app_userevent.so] => (Custom User Event Application)
[app_transfer.so] => (Transfer)
[format_jpeg.so] => (JPEG (Joint Picture Experts Group) Image Format)
[format_wav.so] => (Microsoft WAV format (8000hz Signed Linear))
[func_enum.so] => (ENUM Related Functions)
[format_pcm.so] => (Raw uLaw 8khz Audio support (PCM))
[codec_g726.so] => (ITU G.726-32kbps G726 Transcoder)
[codec_speex.so] => (Speex/PCM16 (signed linear) Codec Translator)
[app_math.so] => (Basic Math Functions)
[app_setcallerid.so] => (Set CallerID Application)
[format_wav_gsm.so] => (Microsoft WAV format (Proprietary GSM))
[app_settransfercapability.so] => (Set ISDN Transfer Capability)
[app_senddtmf.so] => (Send DTMF digits Application)
[app_url.so] => (Send URL Applications)
[cdr_custom.so] => (Customizable Comma Separated Values CDR Backend)
[app_stack.so] => (Stack Routines)
[format_g723.so] => (G.723.1 Simple Timestamp File Format)
[app_verbose.so] => (Send verbose output)
[app_cut.so] => (Cut out information from a string)
[codec_alaw.so] => (A-law Coder/Decoder)
[app_zapscan.so] => (Scan Zap channels application)
[app_festival.so] => (Simple Festival Interface)
[app_test.so] => (Interface Test Application)
[app_externalivr.so] => (External IVR Interface Application)
[app_mp3.so] => (Silly MP3 Application)
[app_random.so] => (Random goto)
[app_softhangup.so] => (Hangs up the requested channel)
[app_db.so] => (Database Access Functions)
[app_sms.so] => (SMS/PSTN handler)
[format_sln.so] => (Raw Signed Linear Audio support (SLN))
[app_milliwatt.so] => (Digital Milliwatt (mu-law) Test Application)
[app_lookupcidname.so] => (Look up CallerID Name from local database)
Asterisk Ready.
Mar 20 16:25:11 NOTICE[6476]: chan_sip.c:11206 handle_request_register: Registration from ‘ATA Line 1 sip:[email protected];user=phone’ failed for ‘192.168.1.25’ - Username/auth name mismatch
Mar 20 16:25:11 NOTICE[6476]: chan_sip.c:11206 handle_request_register: Registration from ‘ATA Line 2 sip:[email protected];user=phone’ failed for ‘192.168.1.25’ - Username/auth name mismatch
Asterisk Started
I have abandoned my attempt to actually get a working system. Even the FreePBX developers say the current version of FreePBX is NOT compatible with Asterisk v1.4.x. Period!
Anyone who says they have it working is either being misleading or has a warped sense of what the definition of “working” is.
Yes, you can get FreePBX to install and get the GUI up, add extensions etc. Just too many little and not so little errors to consider it really “working” though.
Well, I got rid of the console messages, by reducing the verify level (asterisk -cvvvvr) I hope that I am not missing anthing there…
However, on a reboot, asterisk is not restarting, and I get the following in the logs.
(Also, has anyone worked out whether installing asterisk-gui (for 1.4) over a system that is 1.4.1 but with freepbx 2.2.1 - or do I wait for 2.3 for that… looking at things like turning on the jabber support.
Any thoughts on fixes… I have tried a zaptel rebuild.
Mar 19 06:14:56 pqpbx zaptel: Loading zaptel framework: succeeded
Mar 19 06:15:02 pqpbx zaptel: Waiting for zap to come online: succeeded
Mar 19 06:15:02 pqpbx modprobe: FATAL: Error inserting tor2 (/lib/modules/2.6.9-
34.0.2.EL/extra/tor2.ko): Unknown symbol in module, or unknown parameter (see dm
esg)
Mar 19 06:15:02 pqpbx modprobe: FATAL: Error running install command for tor2
Mar 19 06:15:02 pqpbx zaptel: Loading tor2: failed
Mar 19 06:15:02 pqpbx modprobe: FATAL: Error inserting wct4xxp (/lib/modules/2.6
.9-34.0.2.EL/extra/wct4xxp.ko): Unknown symbol in module, or unknown parameter (
see dmesg)
Mar 19 06:15:02 pqpbx modprobe: FATAL: Error running install command for wct4xxp
Mar 19 06:15:02 pqpbx zaptel: Loading wct4xxp: failed
Mar 19 06:15:02 pqpbx modprobe: FATAL: Error inserting wct1xxp (/lib/modules/2.6
.9-34.0.2.EL/extra/wct1xxp.ko): Unknown symbol in module, or unknown parameter (
see dmesg)
Mar 19 06:15:02 pqpbx modprobe: FATAL: Error running install command for wct1xxp
Mar 19 06:15:02 pqpbx zaptel: Loading wct1xxp: failed
Mar 19 06:15:02 pqpbx modprobe: FATAL: Error inserting wcte11xp (/lib/modules/2.
6.9-34.0.2.EL/extra/wcte11xp.ko): Unknown symbol in module, or unknown parameter
(see dmesg)
Mar 19 06:15:02 pqpbx modprobe: FATAL: Error running install command for wcte11x
p
Mar 19 06:15:02 pqpbx zaptel: Loading wcte11xp: failed
Mar 19 06:15:02 pqpbx zaptel: Loading wcfxo: succeeded
Mar 19 06:15:02 pqpbx modprobe: FATAL: Error inserting wctdm (/lib/modules/2.6.9
-34.0.2.EL/extra/wctdm.ko): Unknown symbol in module, or unknown parameter (see
dmesg)
Mar 19 06:15:02 pqpbx modprobe: FATAL: Error running install command for wctdm
Mar 19 06:15:02 pqpbx zaptel: Loading wctdm: failed
Mar 19 06:15:02 pqpbx zaptel: Loading ztdummy: succeeded
Mar 19 06:15:02 pqpbx modprobe: FATAL: Error inserting r4fxo (/lib/modules/2.6.9
-34.0.2.EL/extra/r4fxo.ko): Unknown symbol in module, or unknown parameter (see
dmesg)
Mar 19 06:15:02 pqpbx zaptel: Loading r4fxo: failed
Mar 19 06:15:02 pqpbx modprobe: FATAL: Error inserting r1t1 (/lib/modules/2.6.9-
34.0.2.EL/extra/r1t1.ko): Unknown symbol in module, or unknown parameter (see dm
esg)
Mar 19 06:15:02 pqpbx zaptel: Loading r1t1: failed
Mar 19 06:15:02 pqpbx modprobe: FATAL: Error inserting rxt1 (/lib/modules/2.6.9-
34.0.2.EL/extra/rxt1.ko): Unknown symbol in module, or unknown parameter (see dm
esg)
Thanks,
It works now but not sure why it didn’t the first time. I think I had to rename my /usr/src/ directories
Well that was painful.
Was banging my head and finally figured out I had to change permissions on the amportal.conf file. That should be added to the procedure for FreePBX v2.1.1 with Asterisk v1.4.1.
Now I get a version check error when loading the CID lookup and Asterisk CLI modules in module Admin. Any way to get around that?
Other than that it appears to work
works for me, too —
i do the following in the console log - which are annoying.
q
– Remote UNIX connection disconnected
– Remote UNIX connection
– Remote UNIX connection disconnected
– Remote UNIX connection
– Remote UNIX connection disconnected
– Remote UNIX connection
– Remote UNIX connection disconnected
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘phpagi’ logged on from 127.0.0.1
== Manager ‘phpagi’ logged off from 127.0.0.1
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘phpagi’ logged on from 127.0.0.1
== Manager ‘phpagi’ logged off from 127.0.0.1
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘phpagi’ logged on from 127.0.0.1
== Manager ‘phpagi’ logged off from 127.0.0.1
– Remote UNIX connection
– Remote UNIX connection disconnected
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Par
[quote]Yes, it’s working fine with me. The changes required are:
cd /usr/src/zaptel
./configure
make menuselect
…
cd /usr/src/asterisk
./configure
make menuselect
…
cd /usr/src/asterisk-addons
./configure
make menuselect
…
Change “show version” command:
cd /usr/src/freepbx/amp_conf/htdocs/admin
cp -vf functions.inc.php functions.inc.php-Original
sed -i ‘s/show version/core show version/’ functions.inc.php
Change (Copy) MOH directory:
cp -vR /var/lib/asterisk/moh /var/lib/asterisk/mohmp3
chown -vR asterisk.asterisk /var/lib/asterisk/mohmp3
chmod -v 775 /var/lib/asterisk/mohmp3
chmod -v 664 /var/lib/asterisk/mohmp3/*
[/quote]
Please follow these steps carefully. It worked and still working for me right now.
Cheers.
I got brave and tried this today.
Is this procedure still working with FreePBX 2.2.1 and Asterisk 1.4.1?
I just tried and asterisk-addons complained during make. Amportal came back with “[FATAL] Error executing asterisk: be sure Asterisk is installed and in the path”
There is no moh directory in /var/lib/asterisk either.
I think I have answered my own question on this - I see hints that I have to wait for Freepbx 2.3 (not 2.2.1) for V1.4 support. ah well…
q
Yes, you are quite wrong!
If you take the time to read the documents and examples in the 1.4 branch you will see that it is more than “flashing lights”
Samyoutan’s posting piqued my interest. I am running trixbox 2.0, but have updated the freepbx to 2.2.1 and the asterisk and zaptel etc to 1.2.16 etc per the nerdvittles security fix…
But I have been “itching” to run V1.4 (or course now v1.4.1) I have all of it’s source trees etc on board. and I guess will build in those i.e. /usr/src/asterisk-1.4.1 rather than moving it to /asterisk
Main concern, i guess, is if I build per sam’s instructions, is there a rollback… If it breaks.
I can (and probably will, walk through the build instructions in the meantime, but would be interested in any observations in that regards.
Peter
[quote=“bubba”]“complete rewrite of the Shared Line Appearance (SLA) support that was first released as part of Asterisk 1.4.0”
This is not SLA as we know it…
it is a hack, uses hints I think not a true SLA as with a KSU…
I think all you will get some blinking lights…
I could be wrong…[/quote]
With all due respect, you obviously do not have a clue what you are talking about!
Yes Sir
Could be that I have no clue as I am not a Diguim dev
I only know what I have seen the asterisk 1.4 “SLA” do, and from what I have been told by “those in the know or so they claim to be” (d-CAP tech’s.)
But as I have no firsthand knownledge of the SLA working / not working
that is all I have to base my comments on
The last line said
I could be wrong
And I really hope I am wrong to have a WORKING ZAP CHANNEL “SLA” for asterisk would be great…
“complete rewrite of the Shared Line Appearance (SLA) support that was first released as part of Asterisk 1.4.0”
This is not SLA as we know it…
it is a hack, uses hints I think not a true SLA as with a KSU…
I think all you will get some blinking lights…
I could be wrong…
[quote=“bubba”]No it does not…I really do not think we will see SLA with Asterisk.
It is not on the hot list as not to many of the hardcore DEV care about the
SoHo (from I hear)
Which is why we do not pull the KSU if trhe customer wants SLA.
what they are / where working on for SLA is just some blinking lights
NO real call pickup…As far as I know now.
but as I could care less about it I do not keep up with it.[/quote]
If you say so. BTW, Asterisk v1.4.1 was released today with full SLA support!
http://www.asterisk.org/node/48320
Does that include SLA (Shared Line Appearance)?