trixbox - The Open Platform For Business Telephony

Nothing in CDR and call records

RichardV — Sun, 08 Nov 2009 10:50:57 +0000

I read a lot about this problem: record recording allways on and no calls are showing in table. And when I look in de CDR no calls are showed.
When I look (with FTP) I see that all of the calls are recorded!
In this forum someone says: It's a bug, it is fixed in the new versions.
So we installed 2.8.0.2 and we have the same problem!
It's a very importante function for us.
Please, who's now the answer....

Sangoma A200 2nd Card & New Server / DAHDI Driver setup

KhunIT — Sun, 08 Nov 2009 08:40:50 +0000

Linksys 3102 not passing my regular tone key presses to Trixbox

kpenner — Sun, 08 Nov 2009 06:10:06 +0000

What setting inside Linksys 3102 firmware do I need to set so that key presses from this regular phone are sent directly to Trixbox?
I have the latest Linksys 5.1.8 firmware.

I want to turn a regular telephone into a SIP phone. I set everything up in Trixbox so that this Linksys 3102 is SIP ext. 109

I can call extension 109 no problem from another SIP phone, it works great. I can answer the call and caller ID works.

If I press **** I get the Linksys Voice menu

When I try to dial from this regular phone, the 3102 is not passing my key presses to Trixbox

In the CLI window I can see that nothing is happening when I make key presses from my 109 phone.

I have a number of SIP ext. on this system, and they all work except for this Linksys problem.

Majic Jack Setup

mst — Sun, 08 Nov 2009 04:08:47 +0000

I am going to pay for helping setup MJ (SIP) on Trixbox.

THX

trixbox on an EEE BOX b202

glendle — Sun, 08 Nov 2009 02:01:29 +0000

Hi there,

Have several EEE BOX b202 's running without issue using 2.8, just got another to do an install, but no network.

After a bit of investigation and googleing, I have found that it doesn't have the same NIC as the others - they have the Realtek NIC, but this one has a JMicron JMC250 Gigabit NIC.

It would appear that the driver for it is enabled in Kernel 2.6.28 and above, but the Kernel Version on tb2.8 is 2.6.18

Is there any way of installing a driver for this hardware or am I flogging a dead Donkey?

Is there any possibility of updating the Kernel to 2.6.31 at all? is this going to monumentally break things?

Thanks in anticipation,
Glenn

search trixbox forums direct from your firefox extra search engine plugin

ahmershuja — Sun, 08 Nov 2009 00:54:52 +0000

Hi guys;

I created "firefox extra search engine plugin" for trixbox forum search, now you can search anything direct from your firefox search plugin,

here is the link http://mycroft.mozdev.org/search-engines.html?name=trixbox

any question please ask here...

Ahmer Shuja

Echo problems using Nettalk please help

rrichiez — Sat, 07 Nov 2009 22:07:11 +0000

Hi all

I just installed a nettalk device which I think its great because I can use it with out a computer on the whole day. Anyway I port it into my digium card and it acts like a pots line, the only thing is that when i call or when recieving a call on my side I hear myself twice. where and how can I tweak the echo problems.

any help or tip will be greatly apreciated.

thanks
Rrichiez

Sharing Voicemail with 2 extensions

vitamin — Sat, 07 Nov 2009 21:46:56 +0000

Hello Guys,

Its that time of year again, when I start to play about with my business phone system lol ;)-

Ideally I would like extensions 200 & 201 to share the same voicemail and also alert both cisco 7940's. Is this possible?

I'm running 2.6.2.3 CE.

Cheers

Sam

qloganalyzer in trixbox

raloheni — Sat, 07 Nov 2009 17:53:26 +0000

how to install the queueloganalyzer in trixbox

i need step by step guide, dont post the readme.txt in the qloganalyzer i didnt suceeeded with the readme.txt file.

so please some one help me

xlite softphone for nokia e90

raloheni — Sat, 07 Nov 2009 17:30:08 +0000

how to use xlite softphone in my nokia e 90 mobile .to intagrate with trixbox server.

is there any mobile softclient which should be rigistrated with trixbox.

help me please .

not a problem, just a solution

skydiverQ — Sat, 07 Nov 2009 08:51:11 +0000

earlier during my trials, I was getting the following error output lines when I hit my trixbox admin from outside my lan:

Warning: gethostbyaddr() [function.gethostbyaddr]: Address is not a valid IPv4 or IPv6 address in /var/www/html/maint/modules/home/includes/os/class.Linux.inc.php on line 68

Warning: Cannot modify header information - headers already sent by (output started at /var/www/html/maint/modules/home/includes/os/class.Linux.inc.php:68) in /var/www/html/maint/modules/home/includes/system_header.php on line 25

I figured out what was causing it was that I had no dns servers specified under System|Network settings. Once I added the dns servers (i just used the opendns ones) then the messages went away.

Just thought I'd post a solution in case anyone else ever gets that problem.

cheers

-sky

Does trixbox support session timer?

daisy5376 — Sat, 07 Nov 2009 06:51:49 +0000

Did somebody know Does trixbox support session timer?

IAX Trunk in now unreachable

888xman — Sat, 07 Nov 2009 03:03:04 +0000

Hello,

I have got 2 trixbox machines connected by IAX in 2 remote locations.

I have been calling between the 2 for about 4 or 5 months, however just in the last week, it stopped working, and am getting the "all circuits are busy" voice).

I am still able to call out (not using IAX)

I haven't changed anything with the Trixbox setup over the last 4 months or so, as everything was working fine.

There appears to be some registration problem

At West Perth:
trixbox1*CLI> iax2 show registry
Host dnsmgr Username Perceived Refresh State
58.6.6.30:4569 N Remote1 <Unregistered> 60 Request Sent

trixbox1*CLI> iax2 show peers
Name/Username Host Mask Port Status
home/Remote1 58.6.6.30 (S) 255.255.255.255 4569 (T) UNREACHABLE
420 (Unspecified) (D) 255.255.255.255 0 UNKNOWN
2 iax2 peers [0 online, 2 offline, 0 unmonitored]
trixbox1*CLI>

At Home:
trixbox1*CLI> iax2 show registry
Host dnsmgr Username Perceived Refresh State

trixbox1*CLI> iax2 show peers
Name/Username Host Mask Port Status
Remote1 (Unspecified) (D) 255.255.255.255 0 UNKNOWN
1 iax2 peers [0 online, 1 offline, 0 unmonitored]
trixbox1*CLI>

When trying to make a call from Home to West Perth, I get:

-- Executing [401@from-internal:1] Set("SIP/500-087d9938", "INTRACOMPANYROUTE=YES") in new stack -- Executing [401@from-internal:2] Macro("SIP/500-087d9938", "user-callerid|SKIPTTL|") in new stack -- Executing [s@macro-user-callerid:1] NoOp("SIP/500-087d9938", "user-callerid: device 500") in new stack -- Executing [s@macro-user-callerid:2] Set("SIP/500-087d9938", "AMPUSER=500") in new stack -- Executing [s@macro-user-callerid:3] GotoIf("SIP/500-087d9938", "0?report") in new stack -- Executing [s@macro-user-callerid:4] ExecIf("SIP/500-087d9938", "1|Set|REALCALLERIDNUM=500") in new stack -- Executing [s@macro-user-callerid:5] NoOp("SIP/500-087d9938", "REALCALLERIDNUM is 500") in new stack -- Executing [s@macro-user-callerid:6] Set("SIP/500-087d9938", "AMPUSER=500") in new stack -- Executing [s@macro-user-callerid:7] Set("SIP/500-087d9938", "AMPUSERCIDNAME=Kelly Home") in new stack -- Executing [s@macro-user-callerid:8] GotoIf("SIP/500-087d9938", "0?report") in new stack -- Executing [s@macro-user-callerid:9] Set("SIP/500-087d9938", "AMPUSERCID=500") in new stack -- Executing [s@macro-user-callerid:10] Set("SIP/500-087d9938", "CALLERID(all)="Kelly Home" <500>") in new stack -- Executing [s@macro-user-callerid:11] Set("SIP/500-087d9938", "REALCALLERIDNUM=500") in new stack -- Executing [s@macro-user-callerid:12] ExecIf("SIP/500-087d9938", "0|Set|CHANNEL(language)=") in new stack -- Executing [s@macro-user-callerid:13] NoOp("SIP/500-087d9938", "TTL: ARG1: SKIPTTL") in new stack -- Executing [s@macro-user-callerid:14] GotoIf("SIP/500-087d9938", "1?continue") in new stack -- Goto (macro-user-callerid,s,23) -- Executing [s@macro-user-callerid:23] NoOp("SIP/500-087d9938", "Using CallerID "Kelly Home" <500>") in new stack -- Executing [401@from-internal:3] Set("SIP/500-087d9938", "_NODEST=") in new stack -- Executing [401@from-internal:4] Macro("SIP/500-087d9938", "record-enable|500|OUT|") in new stack -- Executing [s@macro-record-enable:1] GotoIf("SIP/500-087d9938", "0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] AGI("SIP/500-087d9938", "recordingcheck|20091107-104841|1257562121.1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20091107-104841|1257562121.1: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing [s@macro-record-enable:5] NoOp("SIP/500-087d9938", "No recording needed") in new stack -- Executing [401@from-internal:5] Macro("SIP/500-087d9938", "dialout-trunk|3|401||") in new stack -- Executing [s@macro-dialout-trunk:1] Set("SIP/500-087d9938", "DIAL_TRUNK=3") in new stack -- Executing [s@macro-dialout-trunk:2] ExecIf("SIP/500-087d9938", "0|Authenticate|") in new stack -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/500-087d9938", "0?disabletrunk|1") in new stack -- Executing [s@macro-dialout-trunk:4] Set("SIP/500-087d9938", "DIAL_NUMBER=401") in new stack -- Executing [s@macro-dialout-trunk:5] Set("SIP/500-087d9938", "DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing [s@macro-dialout-trunk:6] Set("SIP/500-087d9938", "GROUP()=OUT_3") in new stack -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/500-087d9938", "1?nomax") in new stack -- Goto (macro-dialout-trunk,s,9) -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/500-087d9938", "1?skipoutcid") in new stack -- Goto (macro-dialout-trunk,s,12) -- Executing [s@macro-dialout-trunk:12] AGI("SIP/500-087d9938", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix > fixlocalprefix: Using pattern 4XX == fixlocalprefix: Dialpattern 4XX matched. 401 -> 401 -- AGI Script fixlocalprefix completed, returning 0 -- Executing [s@macro-dialout-trunk:13] Set("SIP/500-087d9938", "OUTNUM=401") in new stack -- Executing [s@macro-dialout-trunk:14] Set("SIP/500-087d9938", "custom=IAX2/Remote1") in new stack -- Executing [s@macro-dialout-trunk:15] GotoIf("SIP/500-087d9938", "1?gocall") in new stack -- Goto (macro-dialout-trunk,s,17) -- Executing [s@macro-dialout-trunk:17] Macro("SIP/500-087d9938", "dialout-trunk-predial-hook|") in new stack -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/500-087d9938", "0?bypass|1") in new stack -- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/500-087d9938", "0?customtrunk") in new stack -- Executing [s@macro-dialout-trunk:20] Dial("SIP/500-087d9938", "IAX2/Remote1/401|300|tr") in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s@macro-dialout-trunk:21] Goto("SIP/500-087d9938", "s-CHANUNAVAIL|1") in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/500-087d9938", "1?noreport") in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/500-087d9938", "TRUNK Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack -- Executing [401@from-internal:6] Macro("SIP/500-087d9938", "outisbusy|") in new stack -- Executing [s@macro-outisbusy:1] Playback("SIP/500-087d9938", "all-circuits-busy-now|noanswer") in new stack -- <SIP/500-087d9938> Playing 'all-circuits-busy-now' (language 'en') == Spawn extension (macro-outisbusy, s, 1) exited non-zero on 'SIP/500-087d9938' in macro 'outisbusy' == Spawn extension (macro-outisbusy, s, 1) exited non-zero on 'SIP/500-087d9938'

My details at Home:
Dial Rules:
4XX
Trunk Name
Remote1
Peer Details:
Disallow=all
Allow=ulaw&alaw&g729
context=from-internal
host=dynamic
qualify=yes
secret=******
type=friend
jitterbuffer=yes

Details at West Perth:
Dial Rules
5XX
Trunk Name
home
Peer Details:
Disallow=all
Allow=ulaw&alaw&g729
host=58.6.6.30
jitterbuffer=yes
secret=*******
trunk=yes
type=friend
qualify=yes
username=Remote1
context=from-internal
Register String
Remote1:(secret of home)@58.6.6.30 (58.6.6.30 is the IP at Home)

Is someone able to steer me in the right direction with this one?

Thanks for any help.

Kind regards,
Anthony

Connect two Trixbox's together and forward all calls from one.

SteveW — Sat, 07 Nov 2009 00:22:25 +0000

Hey all,

We've currently got a Trixbox with * 1.2 running SCCP with 8 phones and one E1 PRI working a treat. We have added another site and want to migrate the system there but leave the PRI line and card where it is.

What we have at the new site is another (new) Trixbox with * 1.4 running SCCP with the phones from the old site installed with one BRI card. That set-up is working fine it can make and receive calls using its local ISDN card.

What I am after is breaking out from the old box to the new one. Currently all calls come into the context below and are sent to the timeconditions() as shown. The DID's work the same other than going to the extension set-up.

I am after breaking out where the timeconditions() call is and send all calls to the new box. At the XXXXXX I assume would be something like the call to a route and a trunk between the boxes?

The old machine won't have any local extensions or anything it is purely there because it is located where the E1 PRI is terminated. When running a Trixbox and a CCME I know that a small macro to the pilot number of the CCME for a hunt group works fine and we have used this in the past - but I cannot see how I can forward all the calls from one box to another box and "inject" it into the dialplan as-is they were coming from a local trunk? The DID I guess can be fixed by calling the stage 031,n,Goto(from-did-direct,8031#,1) and using the # to donate a non local ext number? I might be looking in the wrong place and assume you don't just create a IAX2 <> IAX2 and then route that? The boxes have a IAX2 trunk between them that works fine (used for testing).

Example code from the PRI Trixbox:

[ext-did]
;Main Number
exten => 486,1,Set(__FROM_DID=${EXTEN})
exten => 486,n,Gosub(app-blacklist-check,s,1)
exten => 486,n,Gosub(cidlookup,cidlookup_2,1)
exten => 486,n,GotoIf($[ "${CALLERID(name)}" != "" ] ?cidok)
exten => 486,n,Set(CALLERID(name)=${CALLERID(num)})
exten => 486,n(cidok),Noop(CallerID is ${CALLERID(all)})
exten => 486,n,Set(FAX_RX=disabled)
exten => 486,n,Set(_ALERT_INFO=feature)
;XXXXXX
exten => 486,n,Goto(timeconditions,1,1)
;DID
exten => 031,1,Set(__FROM_DID=${EXTEN})
exten => 031,n,Gosub(cidlookup,cidlookup_1,1)
exten => 031,n,GotoIf($[ "${CALLERID(name)}" != "" ] ?cidok)
exten => 031,n,Set(CALLERID(name)=${CALLERID(num)})
exten => 031,n(cidok),Noop(CallerID is ${CALLERID(all)})
exten => 031,n,Set(FAX_RX=disabled)
exten => 031,n,Set(_ALERT_INFO=urgent)
exten => 031,n,Goto(from-did-direct,8031,1)

Any help would be very much appreciated.

Thanks

Steve

Can't reach trixbox http properly with new static IP address?

skydiverQ — Fri, 06 Nov 2009 23:24:16 +0000

Hi all, tia for your assistance...

I've recently acquired a static IP from my provider and am in the process of getting everything moved over and reconfigured. I have everything working just fine apart from the trixbox.

I can reach it using putty, I can reach the webadmin of the box via port 10000, but it will not respond when I try to hit port 8080 for the trixbox admin.

when I try to hit the admin locally (http://192.168.2.5:8080) i get some funky error messages at the top of the status screen:

Warning: gethostbyaddr() [function.gethostbyaddr]: Address is not a valid IPv4 or IPv6 address in /var/www/html/maint/modules/home/includes/os/class.Linux.inc.php on line 68

When I try to get the system status locally, I get an XML loading error.

I'm guessing this is just a simple configuration setting somewhere in the network settings or apache settings, but for the life of me I can't find it. Any thoughts?

thanks,

sky

Avaya Partner 18D Phones

miggl2 — Fri, 06 Nov 2009 22:23:27 +0000

I was wondering if it is possible to connect AVAYA Partner 18D phones directly to TrixBox CE? We have a client with these phones and were hoping to get them started on TrixBox. I haven't been able to actually take a look at the phones themselves, since our client is using them for day-to-day business. And so far I have only had experience with GrandStream phones, which have a built-in administration console, ideal for setting them up with any SIP provider.

I'd love to hear any experiences, or suggestions on further resources (a google search brought up very few hits that pertain to this situation).

Thanks!

How do i get Trixbox 2.6.2.3 to recognize more than 4GB of ram

reef — Fri, 06 Nov 2009 22:07:11 +0000

Hi guys,

so i have a dell PE2950 with 16GB of Ram, the install is 2.6.2.3.

Ive got about 200 Extensions and always have about 50 concurrent calls.

Right now i have free about 230k out of 4gb and its hurting my call quality.

I did some research and everyone says to install PAE kernel to get the system to recognize more ram. however...

~]# yum search kernel-PAE
Warning: No matches found for: kernel-PAE
No Matches found

~]# yum search PAE
================================================================================================ Matched: PAE =================================================================================================
kmod-cmirror-PAE.i686 : cmirror kernel module(s)
kmod-drbd-PAE.i686 : drbd kernel module(s)
kmod-drbd82-PAE.i686 : drbd82 kernel module(s)
kmod-drbd83-PAE.i686 : drbd83 kernel module(s)
kmod-gfs-PAE.i686 : gfs kernel module(s)
kmod-gnbd-PAE.i686 : gnbd kernel module(s)
kmod-kvm-PAE.i686 : kvm kernel module(s)
kmod-xenpv-PAE.i686 : xenpv kernel module(s)
kmod-xfs-PAE.i686 : xfs kernel module(s)

so i dont see the PAE Kernel, i also have all the repositories selected in the gui.

any ideas?

thanks for the help in advanced.

Master.csv - File rollover

mikecimi — Fri, 06 Nov 2009 20:15:50 +0000

Hi,
I am attempting to output the cdr records in realtime via a serial port to an external call acounting system. I can do a tail -f Master.csv just fine, bit that only works for 10-20 min. It seems that when the files are pushed or pulled to the trixbox report servers, the file get moved and a new Master.csv file gets started. Does anyone know how this process works? Is there a script that gets executed that I can possibly tweak? I'm running Version 4.1

Thanks
-Mike

Trixbox TLS SRTP + MD5 (Security)

Linuxx — Fri, 06 Nov 2009 20:11:50 +0000

I just had a few questions about Trixbox (more Asterisk) security.
From what I can see (from wireshark) the SIP passwords during the authentication phase are sent encrypted with MD5. Am I correct in this statement? Additionally has anyone had any success with Asterisk TLS encryption + eyeBeam soft phones? I have a self signed certificate installed, and have the box listening on 5061 (TCP). The phone registers fine, but I cant make a call. eyeBeam gives me "beeps" and I see this on Asterisk;

<--- Reliably Transmitting (NAT) to 10.20.122.74:4674 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 10.20.122.74:46119;branch=z9hG4bK-d8754z-0e38d85121741936-1---d8754z-;received=10.20.122.74;rport=4674
From: "Joe Schmo"sip:101@trixbox.corporate.local;tag=2978c14b
To: "500"sip:500@trixbox.corporate.local;tag=as713a8e22
Call-ID: ZWI3ODRjYzIxZjZlZDlmMzAyMWM4YTcxODE5MjM3ZTQ.
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.9-samy-r27
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="725fc8ec"
Content-Length: 0

Add Inbound Route to Reports

trafalger — Fri, 06 Nov 2009 19:05:37 +0000

Hey guys,

Does anyone know if it's possible (and how) to add Inbound Route to the reports. I have a 100 # PRI and I want to be able to monitor for heavy users.

Thanks

/etc/init.d/asterisk restart kill`s Trixbox

raymonvdm — Fri, 06 Nov 2009 18:02:54 +0000

I just found out that if i restart asterisk using:
/etc/init.d/asterisk restart

I get the following error:
sip2 ~]# /etc/init.d/asterisk restart
Stopping PBXtra Core service: Asterisk ended with exit status 0
Asterisk shutdown normally.
[ OK ]
Starting PBXtra Core service: fs.file-max = 262144
[ OK ]
/etc/init.d/asterisk: line 114: /etc/init.d/FONmon: No such file or directory
[sip2 ~]#

The result of this are the following running services:

[sip2 ~]# ps aux | grep asterisk
asterisk 1905 0.0 0.8 27360 4432 ? S 18:37 0:00 /usr/sbin/httpd
asterisk 1906 0.0 0.8 27360 4432 ? S 18:37 0:00 /usr/sbin/httpd
asterisk 1907 0.0 0.8 27360 4432 ? S 18:37 0:00 /usr/sbin/httpd
asterisk 1908 0.0 0.8 27360 4432 ? S 18:37 0:00 /usr/sbin/httpd
asterisk 1914 0.0 0.8 27360 4432 ? S 18:37 0:00 /usr/sbin/httpd
asterisk 1915 0.0 0.8 27360 4432 ? S 18:37 0:00 /usr/sbin/httpd
asterisk 1916 0.0 0.8 27360 4432 ? S 18:37 0:00 /usr/sbin/httpd
asterisk 1917 0.0 0.8 27360 4432 ? S 18:37 0:00 /usr/sbin/httpd
asterisk 2098 0.0 0.0 4488 440 ? S 18:38 0:00 bash -c cd /var/www/html/panel && /var/www/html/panel/safe_opserver &
asterisk 2099 0.0 0.1 4484 1028 ? S 18:38 0:00 sh /var/www/html/panel/safe_opserver
asterisk 2101 0.2 1.6 12512 8644 ? S 18:38 0:02 /usr/bin/perl /var/www/html/panel/op_server.pl
root 2116 0.0 1.7 25408 8896 ? S 18:38 0:00 /usr/bin/php -q /var/www/html/aastra/asterisk/aastra_daemon1
root 2117 0.0 1.7 25408 8984 ? S 18:38 0:00 /usr/bin/php -q /var/www/html/aastra/asterisk/aastra_daemon2
root 2253 0.0 0.1 4484 596 pts/0 S 18:51 0:00 /bin/sh /usr/sbin/safe_asterisk -U nobody -G nobody -g
nobody 2267 25.1 4.5 51060 23420 pts/0 Sl 18:51 0:01 /usr/sbin/asterisk -f -U nobody -G nobody -g -vvvg -c
root 2304 0.0 0.1 3916 664 pts/0 R+ 18:51 0:00 grep asterisk

How ever when i boot the machine the following services are running and everything is fine:

[sip2 ~]# ps aux | grep asterisk
asterisk 1905 0.0 0.8 27360 4432 ? S 18:37 0:00 /usr/sbin/httpd
asterisk 1906 0.0 0.8 27360 4432 ? S 18:37 0:00 /usr/sbin/httpd
asterisk 1907 0.0 0.8 27360 4432 ? S 18:37 0:00 /usr/sbin/httpd
asterisk 1908 0.0 0.8 27360 4432 ? S 18:37 0:00 /usr/sbin/httpd
asterisk 1914 0.0 0.8 27360 4432 ? S 18:37 0:00 /usr/sbin/httpd
asterisk 1915 0.0 0.8 27360 4432 ? S 18:37 0:00 /usr/sbin/httpd
asterisk 1916 0.0 0.8 27360 4432 ? S 18:37 0:00 /usr/sbin/httpd
asterisk 1917 0.0 0.8 27360 4432 ? S 18:37 0:00 /usr/sbin/httpd
root 2024 0.0 0.1 4532 608 ? S 18:38 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk
asterisk 2038 0.7 4.5 52308 23684 ? Sl 18:38 0:05 /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c
asterisk 2098 0.0 0.0 4488 440 ? S 18:38 0:00 bash -c cd /var/www/html/panel && /var/www/html/panel/safe_opserver &
asterisk 2099 0.0 0.1 4484 1028 ? S 18:38 0:00 sh /var/www/html/panel/safe_opserver
asterisk 2101 0.2 1.6 12512 8644 ? S 18:38 0:02 /usr/bin/perl /var/www/html/panel/op_server.pl
root 2116 0.0 1.7 25408 8892 ? S 18:38 0:00 /usr/bin/php -q /var/www/html/aastra/asterisk/aastra_daemon1
root 2117 0.0 1.7 25408 8984 ? S 18:38 0:00 /usr/bin/php -q /var/www/html/aastra/asterisk/aastra_daemon2
root 2210 0.0 0.1 3916 664 pts/0 R+ 18:50 0:00 grep asterisk

I found out that the following rule is bad in /etc/init.d/asterisk

# Start up PBXtra Core
daemon /usr/sbin/safe_asterisk "-U nobody -G nobody -g"
RETVAL=$?

So i replaced it with

# Start up PBXtra Core
#raymon# daemon /usr/sbin/safe_asterisk "-U nobody -G nobody -g"
daemon /usr/sbin/safe_asterisk "-U asterisk -G asterisk -g"
RETVAL=$?

But still /etc/init.d/asterisk restart is not working properly. However asterisk is started fine for now. (Inbound calls are working)

I still get the following error:

/etc/init.d/asterisk: line 114: /etc/init.d/FONmon: No such file or directory

Where did /etc/init.d/FONmon went ?

Trixbox Migration

Skibum — Fri, 06 Nov 2009 16:54:57 +0000

I am activating a new Trixbox appliance running CE 2.6 to replace my system that is running on substandard hardware and would like to just migrate my current configuration over to the new appliance without going through a complete reconfiguration. Is there a known process or a config file that I can copy over to my appliance or do i have to reconfig from scratch.

Any direction would be appriciated,

Jeff

digium wildcard card 400p not able to configure

stefan24 — Fri, 06 Nov 2009 16:18:30 +0000

Hello,

I've been tasked with the job of rebuilding the phone system at work, to be honest I'm not an expert in the area, I have learnt a lot in the past couple of days but I'm still hiting a few walls.

I am at present trying to configure trixbox to reconse the computer's digium wildcard 400p card, it's plugged in correctly with a fresh installation of trixbox, with all the latest updates as well as centos updates.

I am currently following this guide to configure the drivers for the card, http://www.cadvision.com/blanchas/Asterisk/DahdiDrivers.html, I am correct to think that the drivers for this card are installed along with the rest of the system?

I believe this card is deader than Elvis and I just want to check it's not something I've missed!

I have run the following command dahdi_cfg -vv which genorates the following output

DAHDI Version: 2.2.0
Echo Canceller(s):
Configuration
======================

Channel map:

0 channels to configure.

Also if I run the following command dadhi_scan the system returns the following...

[1]
active=yes
alarms=UNCONFIGURED
description=DAHDI_DUMMY/1 (source: Linux26) 1
name=DAHDI_DUMMY/1
manufacturer=
devicetype=DAHDI Dummy Timing
location=
basechan=1
totchans=0
irq=0

What can I do to configure this card I believe that is supported out of the box? http://trixbox.org/wiki/hardware-support

thanks for any help offered :)

new installation cant dial out on opstn

idoorsamy — Fri, 06 Nov 2009 13:43:52 +0000

hi ther , i have recently installe dtrixbox on my pc and have a t4 port analogue pstn interface card , everytime i dial out i get this message , the number you have called is invalid ,check number and dial againa. all internal calls are fine , pls can someone help me

Call waiting stopped on SPA-942 from Queue

samwelle — Fri, 06 Nov 2009 12:38:10 +0000

I have multiple extensions all are members of a queue. Call Waiting is turned on for all extensions which works fine. However this week one ectension has stopped receiving the 2nd call (waiting call) whilst all the others work fine.

I have disabled & re enabled call waiting on the extension but no look, I have also checked the device, it's a Linksys SPA-942 and all is fine there too.

Anyone else seen this problem?

Cheers

Remove authorisation for extensions, how?

smirn0ff — Fri, 06 Nov 2009 11:51:51 +0000

Hello all,

I want to remove the authorisation by Extension on my Trixbox, the caller to be authorised only by IP address.

I want to accept all calls comming from one IP and to get them out from the outbound trunks. The Calling number every time will be different for any call.

Please give a tip or a solution

Thanks!

SIP Trunks advanced config?

smirn0ff — Fri, 06 Nov 2009 11:50:48 +0000

Hi guys,

I want to make the Trixbox to rotate my trunks by minutes that were going trough every one of the trunks.

Example:

100 = 32 min outbound calls
101 = 28 min outbound calls
102 = 44 min outbound calls

when a call is made, to be routed at the first time to 101 (less outbound minutes than other trunks) if it's busy to 100 and etc.

Call monitor wav recordings play back too fast

MercJones — Fri, 06 Nov 2009 10:35:57 +0000

Hi there.

If i log into the portal in user mode, and then have a look at the call monitor page, any recording I play seems to be corrupted, or in the wrong format. The wav files play back too fast and they are un understandable.

I've even tried downloading these, or even getting them directly from /var/spool/.. but the problem still exists.

Is there any way to convert these wavs to a format that will allow me to play them back, and how can i fix this for future wav recordings too?

Thanks for the help

Callback from voicemail

fitzrik — Fri, 06 Nov 2009 09:25:15 +0000

Using the instructions below a user can now automatically call back someone who left a voicemail. It works but does not go out the appropriate trunk OR does not place a 9 which would dial the appropriate trunk. I know it mentions context but I am not sure what to do. Can someone suggest an answer.

Thanks

Adding Automatic Callbacks to Your Asterisk Voice Mail System. Don’t you just love those Baby Bell phone messages that say “Press 1 and, for a charge of 75¢, we will place this call for you”? Well, now you can add similar functionality to your Asterisk Voicemail System minus the 75¢ charge. Sometimes it’s a lot more convenient to have the computer do the dialing after you’ve listened to a voicemail message particularly when you’re zipping down the highway at warp speed. To add the functionality to Asterisk using the Asterisk Management Panel (AMP) or freePBX, here’s how. Open the settings for the voicemail extension you wish to configure with this option: Settings->Extensions->ext#. Now scroll down to the vm options field and add the following: callback=from-internal. Save your changes and click the big Red Bar to update Asterisk. In the future, when you listen to a voicemail message on this extension and want to automatically return the call, choose 3 for Advanced Options and then 2 to Return the Call. Just be sure your Default Asterisk Outbound Route is configured to dial using the same number format as your received CallerID numbers, and you’re all set. And, by the way, this works with Asterisk@Home versions at least as far back as 1.5. Too bad no one ever bothered to document it. Very slick!

Function VOLUME Asterisk 1.6

Faust — Fri, 06 Nov 2009 09:02:18 +0000

Hello!

How to use a new function VOLUME in Asterisk 1.6
I use it:
In extensions_custom.conf after [from-internal-custom]
and before [custom-meetme3] insert:

[from-volume-custom]
exten => _X.,1,Set(VOLUME(rx)=10)
exten => _X.,2,Set(VOLUME(tx)=10)
exten => _X.,3,GoTo(from-internal,${EXTEN},1)

and reload asterisk, but it does not work.
What wrong?