inux Users Mailing List
> Subject: Re: [Astlinux-users] multi-tenant voicemail
> Message-ID: <525f84af-435f-4f71-aa91-86dc186d4...@mksolutions.info>
> Content-Type: text/plain; charset=utf-8
>
>
>> Am 17.08.2018 um 03:15 schrieb Shamus Rask :
>>
>> This is
This is more of an Asterisk question, but I’m hoping someone can share their
experience. I’m setting up a multi-tenant system, and in doing so discovered
the following. When I tried the following in voicemail.conf (note this is
directly from the sample config):
[default]
1234 => 4242,Example Ma
Here is my scenario. I have an Asterisk 13 server with a publicly accessible
IP. On the same server, I’ve installed OpenVPN server. At my office, I have a
Ubiquity router running an OpenVPN client to my Asterisk box. My SIP phones are
behind this router and I’m trying to get them to register.
A
I learnt something… apparently you need to restart Asterisk for it to re-parse
modules.conf. I assumed doing a “core reload” or “module reload” would be
sufficient.
Thanks for your suggestions!
Shamus
--
Check out t
I’m running a default installation of Asterisk 13 on Ubuntu 16.04. Currently
trying to “skinnify” Asterisk to minimize the attack surface and resource usage.
In /etc/asterisk, I currently see the following:
total 16
drwxr-xr-x 3 asterisk asterisk 4096 Apr 14 14:11 .
drwxr-xr-x 96 root root
I’m looking to deploy several SIP phones for a client in remote locations. I’ve
done some Googling, but there isn’t a lot of information out there. Does anyone
have any good recommendations for a SIP phone with built-in VPN client? Looking
to have phone be able connect to any user’s home network
I’ve copied over the default configuration files. It looks like the default are
all commented out (i.e. blank), so they have no real impact. I wish there was a
definitive list of what a minimal configuration of Asterisk looks like. Right
now I’ve got the following config files in /etc/asterisk:
I have just migrated from Asterisk 1.8 to 13. I run AstLinux as a VM, so the
“migration” was simple–download/install a new VM, copy configs and shutdown old
version.
When I reload Asterisk, I now see many errors of the type:
-- Reloading module 'res_pjsip_transport_management.so' (PJSIP Reliabl
Michael,
Apologies… I never thought to look at the AstLinux documentation.
However, and this is for the archives, here are my particulates:
- VM Fusion v7.1.3
- ESXi v5.5
Following your instructions, I was unable to import the .vmdk file into ESXi.
After several frustrating attempts, I instead
I’ve done some searching, but have not found anyone who has done this. I’ve
just received a second-hand Dell server and have installed ESXi on it. I’d like
to be able to install AstLinux as a VM instance on it, but cannot figure out
how to make this happen. I’ve tried creating a bootable-USB, bu
Thank you to all for your suggestions. Based on your input and Google’s I am
now doing the following:
On my Asterisk server to be monitored I am running a script that launches an
Originate through the AMI. This calls my personal PBX which, using the
BLACKLIST function, sees the incoming CID and
Ioan,
How would I detect the non-answer on my personal PBX from the calling Asterisk
server? I was hoping to accomplish this in the dialplan.
> Hello Shamus,
>
> You could not answer the call so you will not be charged at all. Does
> this make sense?
>
> Best regards,
> Ioan
--
I’m posting this here as I find the AstLinux users offer the best and most
responsive insights to all things Asterisk.
I’ve started a SaaS-based project that uses Asterisk to make outbound calls.
I’d like to implement some form of rigorous monitoring to ensure that my 2x
outbound SIP trunks are
Apparently I’m a bit of of an idiot. After following your advice and digging
through countless other forums, turns out I had “fromuser” set in my sip.conf.
That being said, in playing I did discover the following. Not sure if this is
of interest to anyone:
In Asterisk 1.8, you can set sendrpid
My SIP provider confirmed that they have not made any changes and that I should
be allowed to change my callerID on outgoing calls. I have sent them the
following SIP debug output from Asterisk.
In this instance, I used the Dial (no “o”) following CALLERID(num-pres=allowed)
and setting just th
Lonnie,
I removed the “o” option and took a look at the documentation for Asterisk 1.8.
Looks like there are two options that could be of interest:
f - If x is not provided, force the CallerID sent on a call-forward or
deflection to the dialplan extension of this Dial() using a dialplan hint. F
Christopher,
Thanks for the suggestion… tried it, but still no luck.
To be clear, all of the scenarios I’ve tried have still allowed my calls to go
through, they’re just showing my DID as CallerID rather than what I want to set
it to. I’ve verified that my DID’s CallerID feature is set to Dyna
Many thanks for your suggestions; unfortunately none of them worked.
I tried:
(1) setting the following in sip.conf [general]
rpid_update=yes
sendrpid=yes
(2) Set(CALLERID(all)=“S RASK” <777888>)
(3) Set(CALLERID(num-pres)=allowed)
Set(CALLERID(num)=777888)
(4) SipAddHeader(P-Asser
About a year ago, I upgraded my AstLinux install and started using Asterisk 1.8
instead of 1.4. I have only just discovered that my dial plan that had
previously “anonymized” outgoing calls no longer worked… I had to change it to
the following:
exten => _*27.,1,Set(CALLERID(num-pres)=prohib)
Thanks for the responses. I tried Lonnie’s suggestion adding the NAT rules and
it worked. I was hoping for something more elegant.
Just wondering if the following would be possible… On my LAN (192.168.10.0/24)
I have an existing Ubuntu-based server. This is on the same subnet that
AstLinux see
Running the latest version of AstLinux on a box with 2x Ethernet ports. Eth0 is
my external interface and I’ve assigned a static IP, this sits on my LAN. Eth1
is the local port and serves as DHCP/DNS server for all my SIP phones. These
are assigned an address in the 192.168.5.0/24 range and are
I have an existing AstLinux instance running on a motherboard that includes 2x
GigE interfaces. I’m looking to “complicate” my installation and am wondering
if the following is possible from the web-interface?
eth0:
- VLAN60, static IP of 192.168.6.6
- hosts web-interface
- connected to router w
" cause the db file to be
> closed/reopened?
>
> David
>
> On Fri, Dec 5, 2014 at 9:21 AM, Shamus Rask wrote:
>
>> Lonnie, Michael:
>>
>> Thank you for your replies. I had taken a look at the Perl script on
>> void-info, however, as it states, it is
Lonnie, Michael:
Thank you for your replies. I had taken a look at the Perl script on void-info,
however, as it states, it is not reliable. I am currently doing something
similar from the CLI. The issue is that I have no guarantee of proper sync
without writing a whole write-ack layer.
Lonnie–
Would appreciate any input/ideas the community has on the following…
I have 2 servers running Asterisk in a high-availability configuration (i.e.
one on active-standby for backup). I make extensive use of the default AstDB
(/var/lib/asterisk/astdb.sqlite3) in my dialplan and would like to unders
",
>"") in new stack
>
> Green = DID = OK
> Red = Channel Name = Wrong info
>
> In conclusion you should adapt your dialplan to overwrite some CDR
> information (like accountcode) in order to be able to trace/bill
> correctly the call to the right DID.
&g
NS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
<>
== Spawn extension (default, 7057974986, 2) exited non-zero on
'SIP/7057974933-000b'
> Shamus,
>
> 'type=peer' is what you want for each DID&#x
channel limit exists, sometimes that is a
> default sanity setting by the SIP provider and can be raised at no additional
> cost.
>
> Lonnie
>
>
> On May 26, 2014, at 8:30 AM, Shamus Rask wrote:
>
>> I?m posting here as I find the AstLinux community to be the most frien
I’m posting here as I find the AstLinux community to be the most friendly and
knowledgeable about all things Asterisk!
My ITSP offers DID/SIP trunks at a very competitive rate–each DID includes 5
channels. I’m running Asterisk 11.
The ITSP only offers a single server for both incoming and outgo
;1800"
IDS_IPV6_ENABLE=1
Is there anything obvious I should change in either of these files?
thanks,
Shamus
On Sat, 2014-01-04 at 22:44 -0500, Benjamin L. Naber wrote
> you might want to provide the configs you have for adaptive-ban and IDS
> protection
>
>
> On Sat, 2014-01-
I’m running the latest version of AstLinux with adaptive-ban enabled. This
works a charm on blocking ssh login attempts. However, I recently came across
the following in my Asterisk logs (it appears I’m under some sort of attack):
[Jan 4 21:25:49] ERROR[1344]: res_rtp_asterisk.c:570 ast_rtp_new
Lonnie,
Many thanks, that did the trick! I also enable the adaptive-ban on Status page
in the Prefs tab… wonderful!!! One IP banned, and counting…
cheers,
Shamus
Shamus,
No, the Adaptive ban plugin should always look at /var/log/messages, the
default.
The Asterisk Log() command properly
f file to use the
/var/log/messages instead?
cheers,
Shamus
>
>
> Message: 1
> Date: Tue, 13 Aug 2013 21:46:44 -0400
> From: Shamus Rask
> Subject: Re: [Astlinux-users] adaptive-ban for SIP calls
> To: astlinux-users@lists.sourceforge.net
> Message-ID: <43d0b8ee-a
N_REJECT=0
ADAPTIVE_BAN_WHITELIST_INTERNAL=1
ADAPTIVE_BAN_WHITELIST=""
Is there something obvious that I'm missing?
cheers,
S.
On 2013-08-13, at 9:46 PM, Shamus Rask wrote:
> Lonnie,
>
> Many thanks… I had searched through the archives, but was having problems
> finding
(BANIP=${CHANNEL(recvip)})
> exten => s,n,Log(NOTICE,'${BANIP}' - Dialplan Noted Suspicious IP Address)
> --
> Then the adaptive-ban plugin will act on the above generated log. This way
> you have more control over what to ban or not.
>
> Lonnie
>
>
> On A
Currently running the latest (v112) release of Astlinux. I have enabled the
adaptive-ban and ids-protection firewall plugins. My AstLinux box is sitting
behind my router, where I have port-forwaded 5060-5061 for SIP and my RTP ports.
I just took a look in /var/log/asterisk/messages and found the
can figure it out.
David
On Thu, Jun 27, 2013 at 1:27 PM, David Kerr wrote:
> You could use the asterisk FILTER() function to remove everything except
> numbers 0-9 then either goto(FILTER(0-9,${exten})) or
> dial(${trunk}/$FILTER(0-9,${exten}),bla,bla)
>
> David
>
>
>
This isn't AstLinux specific, however I can no longer find any helpful, active Asterisk forums. The digium ones just seem dead…I'm currently running the latest version of AstLinux with Asterisk v1.8.21. I have an existing dialplan that allows for basic NANP dialing in the form of: exten => _1NXXNXX
I would like to start integrating my AstLinux PBX into my Cacti SNMP monitor.
I've done some searching, but there seems to be very little documentation out
there on Asterisk and SNMP. Just wondering if anyone has already done this and
they can share any pointers/suggestions/resources?
cheers,
Many thanks for your quick response. After checking the logs (nothing
suspicious) I tried the "msmtpqueue" command. Sure enough, all of my VMs were
still sitting on the box. I tried to poke around further, but in frustrating
rebooted the box… what do you know, problem fixed!
When all else fails
Hello AstLinux fans…
For some reason, my voicemail to email has stopped working and I'm hoping
someone can help me understand why. I'm running AstLinux v1.1.0 - Asterisk
v1.8.20.1.
From the Network tab, I have verified that my SMTP mail relay is working (sent
test message successfully).
In
> Lonnie
>
> On Feb 10, 2013, at 12:22 PM, "Fernando F." (mailto:digitaldis...@gmail.com)> wrote:
>
> > Shamus,
> >
> > service stop iptables
> > to start
> > service start iptables
> >
> > Thank You,
> >
> > Fernando
I'm running the latest version of AstLinux. A friend of mine recently got
hacked and I've read about the hacking attempts on this list. Based on this, I
decided it was time to enable the firewall.
>From the network tab; I enabled the firewall with all default settings. I am
>no longer able to
That did the trick... thanks!
> Message: 3
> Date: Thu, 21 Jun 2012 21:27:18 -0500
> From: Lonnie Abelbeck
> Subject: Re: [Astlinux-users] hosts file entries
> To: AstLinux Users Mailing List
> Message-ID: <638e1a94-4437-4032-b30f-1c89cd28b...@lonnie.abelbeck.com>
> Content-Type: text/plain; cha
t;
> Lonnie
>
>
> On Jun 19, 2012, at 11:12 AM, Shamus Rask wrote:
>
> > I've recently activated/established an ISN. To test it, I was trying to
> > call myself, but was unable to do so. I have confirmed it is working with
> > other friends, so there is some sor
I've recently activated/established an ISN. To test it, I was trying to call
myself, but was unable to do so. I have confirmed it is working with other
friends, so there is some sort of internal routing problem on my end.
I took a quick look in the /etc/hosts file and found the following entries
What ports would need to be forwarded to an AstLinux box if it is behind a
firewall/NAT device?
--
Shamus Rask
Sent from my mobile
On Wednesday, 25 April, 2012 at 13:04,
astlinux-users-requ...@lists.sourceforge.net wrote:
>
>
> Message: 2
> Date: Wed, 25 Apr 2012 10:17:11
problem no matter how
many extensions I add.
cheers,
S.
--
Shamus Rask
> Message: 3
> Date: Tue, 28 Feb 2012 21:21:37 -0600
> From: Lonnie Abelbeck (mailto:li...@lonnie.abelbeck.com)>
> Subject: Re: [Astlinux-users] RTP ports - number required
> To: AstLinux
I'm currently running AstLinux 1.0.2 (thank you team, flawless upgrade!) with
Asterisk 1.8.x. My SIP provider allows for up to 5 concurrent calls; I had
limited my port range in both rtp.conf and my firewall to 1 - 10020.
However, I recently noticed that one of my 3 SIP extensions stopped ri
James,
I'm running Asterisk 1.8.x. When I was running Asterisk 1.4, I used
NVFaxDetect() in my incoming context as this was the only way I could get
it to detect a fax on a SIP trunk. Then, I passed it to a separate server
running Ubuntu with the following:
> exten => fax,1,Dial(IAX2/iaxmodem)
>
;t tried it, but the "tiff2pdf -j ..." option will definitely reduce
> the file size. Play with some samples manually before changing your dialplan.
>
> $ tiff2pdf -j -o file_name.pdf file_name.tiff
>
> Lonnie
>
>
> On Feb 14, 2012, at 1:26 PM, Shamus Ra
setup commands in here:
> # inadyn --input_file /etc/inadyn2.conf
> # mkdir /root/usb
> /mnt/kd/bin/check_fax
>
> Looking at the check_fax script... it looks for new faxes (.tiff files) in
> /tmp. It converts all .tiff's into .pdf's (and moves the .tiff into
> /mnt/k
I'm running the latest version of AstLinux with Asterisk 1.8.x. Digging through
the files, it looks as though AstLinux uses msmtp for e-mail notifications
(voicemail, safeasterisk, SIP trunk status...).
I was hoping to use it to send received faxes (*.tif files) to email
recipients. The follow
Michael,
That is good news... thanks! I won't ask why it was in the garage, though
given the phone it is certainly understandable!
I look forward to trying it when the next version of AstLinux emerges.
cheers,
Shamus
Message: 1
> Date: Wed, 8 Feb 2012 15:12:41 +0100
> From: Michael Keuter
>
Can I ask how you're using fax on AstLinux: are you using it for both
receive & send? If you are using it for send, how are you "sending" the
fax--e-mail, file upload...?
Just looking for ideas.
cheers,
Shamus
Message: 2
> Date: Mon, 30 Jan 2012 15:45:44 +0100
> From: Michael Keuter
> Subjec
Feb 2012 12:08:40 +0100
> From: Michael Keuter
> Subject: Re: [Astlinux-users] unistim
> To: AstLinux Users Mailing List
> Message-ID:
> Content-Type: text/plain; charset=us-ascii
>
>
> Am 07.02.2012 um 10:28 schrieb Michael Keuter:
>
> >
> > Am 07.02.2012
Now that I'm happily running the latest version of AstLinux with Asterisk
1.8.8.1 I finally have a chance to try to get a Nortel i2004 working with
my system. I managed to configure the phone, register it to Asterisk, but
am finding that on sending/receiving a call Asterisk is actually crashing!
T
A quick update for the record: my Transcend CF card arrived as well as a
CF-SATA adaptor I ordered from e-bay. I'm happy to report all is working
well! Many thanks to Lonnie & Darrick for their suggestions.
... now on to the next set of challenges with Asterisk 1.8 (so many subtle
dialplan changes
I'm looking for advice on how to best manage a multi-NIC server. My current
AstLinux box has 2 NICs; I would like to do the following and would
appreciate any guidance/suggestions:
NIC 1 - WAN
- PPPoE based
- used only for SIP trunk to provider
- only ports to be opened are 5060 + RTP ports
Lonnie,
I wanted to report back on my trial with the USB drive--it works! I did have to
edit the .run.conf. file as you suggested, but once that was done it booted
quickly and cleanly. Clearly I have something screwy with my CF reader/card
combination. I must have been very lucky to get the one
y slow--AstLinux 1.0.0
> To: AstLinux Users Mailing List
> Message-ID: <3536d92d-e023-4685-995a-e2f35678e...@mksolutions.info>
> Content-Type: text/plain; charset=us-ascii
>
>
> Am 22.12.2011 um 19:00 schrieb Shamus Rask:
>
>> I just cloned my working CF card (Lexar
I just cloned my working CF card (Lexar 4GB Platinum II 80x) to one of my newer
CF cards (Lexar 4GB ) with the following:
imac:~ shamus$ dd if=/dev/disk2 of=/dev/disk1
dd: /dev/disk2: Device not configured
6061608+0 records in
6061608+0 records out
3103543296 bytes transferred in 1909.268174 s
> after that?
>
> Very weird... Like the clock speed is 1000x slower.
>
> Lonnie
>
>
>
>
> On Dec 21, 2011, at 9:29 AM, Shamus Rask wrote:
>
>> I have now tried with a Transcend, Lexar and SanDisk Ultra--all 4GB, all
>> producing the same results. B
esh
> 0.7 images for some time. No reports like this.
>
> Question, after syslinux finally boots runnix, do things work as expected
> after that?
>
> Very weird... Like the clock speed is 1000x slower.
>
> Lonnie
>
>
>
>
> On Dec 21, 2011, at 9:29 AM,
December 21, 2011 9:49 AM
> To: AstLinux Users Mailing List
> Subject: Re: [Astlinux-users] runnix very slow--AstLinux 1.0.0
>
>
> Am 21.12.2011 um 16:29 schrieb Shamus Rask:
>
> >> I have now tried with a Transcend, Lexar and SanDisk Ultra--all 4GB,
> all producing the s
ype: text/plain; charset=windows-1252
>
>
> Am 21.12.2011 um 01:31 schrieb Shamus Rask:
>
>> Lonnie,
>>
>> This is on a new CF card; I've (wisely now) kept my "production" card safe.
>> That being said, I've just tried two e
t; I'm equally stumped.
>
> What kind of hardware are you using?
>
> Lonnie
>
>
>
> On Dec 20, 2011, at 6:31 PM, Shamus Rask wrote:
>
> > Lonnie,
> >
> > This is on a new CF card; I've (wisely now) kept my "production" card
> safe.
s. ie...
> --
> label runnix
>kernel runnix
>append initrd=runnix.img root=/dev/ram0 rw init=/runnix runimg=auto
> rootdelay=10
> --
> If that helps, you can do that with your "os/astlinux-1.0.0.run.conf" KCMD
> line as well.
>
> Let us know if that
riable in there. You should
> uncomment it and set it to:
>
> DMA_DEV=/dev/hda
>
> The entire device would be addressed as DMA.
>
> Darrick
>
> -Original Message-
> From: Shamus Rask [mailto:sha...@srask.ca]
> Sent: Tuesday, December 20, 2011 7:57 AM
you 'exactly' follow the 0.7 to 1.0 upgrade process?
>
> http://doc.astlinux.org/userdoc:upgrade-0.7
>
> Or is this a new fresh install?
>
> Lonnie
>
>
>
> On Dec 19, 2011, at 8:51 PM, Shamus Rask wrote:
>
>> First of all... a huge congratulations
Darrick,
My apologies for the mis-spelling earlier!
Doing an fdisk -l, I see that I have 2 partitions:
/dev/hda1 FAT16 *boot
/dev/hda2 Linux
I do not see an rc.conf.d directory in /mnt/kd, I only see a rc.conf file. Is
this where I should be adding DMA...?
thanks,
Shamus
On 2011-12-
First of all... a huge congratulations and thank you to Lonnie, Derek and the
other developers for reaching the 1.0 milestone--thank you!!!
I've downloaded and copied 1.0.0-Asterisk-1.8.7.1 (Generic i586) onto a CF card
and loaded this into my existing PBX. The box is currently running
AstLinux
with a C7).
>
> Also, to be clear, the new runnix-4 files are not usable with the 0.7.x
> branch at all. The 1.0.x branch needs a new version of runnix because the
> version of squashfs changed (and is not backwards compatible).
>
> Darrick
>
> From: Shamus Rask [mailto:sha
I was able to go and purchase a different 4GB CF card yesterday and tried
again--same result. This got me thinking that maybe my motherboard was
having some sort of other issue with the current release. I downloaded
AstLinux 0.7.7 (prior to current RUNNIX release) and tried again... this
time, succ
uld be an issue with the BIOS of your box for the larger sizes.
>
> Lonnie
>
>
> On Oct 29, 2011, at 3:21 PM, Shamus Rask wrote:
>
> > Lonnie,
> >
> > I have a mini-ITX motherboard, the Jetway J7F4K1G2E. It is based on the
> VIA C7 processor.
> >
&g
Lonnie,
I have a mini-ITX motherboard, the Jetway J7F4K1G2E. It is based on the VIA
C7 processor.
I tried both the VIA C7 image as well as the i586 version--both gave me the
same result.
Shamus
--
Get your Android app mor
Michael, Lonnie:
I tried both your suggestions and neither worked... I'm beginning to suspect
there might be a problem with the CF card. It's a SanDisk Ultra, whereas my
existing, functional, card is a Lexmark. I'll try to get another one today and
try again.
The console provided the following
I was able to download, extract (double-clicking in Finder) and copy the latest
version of AstLinux onto a CF card; however when I went to boot, the system
seemed to hang, almost as though the card was not being seen as bootable.
>From the CLI in Mac, I did the following:
imac:$ sudo diskutil
ssage-ID:
> Content-Type: text/plain; charset=iso-8859-1
>
> Am 27.10.2011 um 19:49 schrieb Shamus Rask:
>
>> I'm currently running the latest version of AstLinux, but running with
>> Asterisk v1.4. I would like to start exploring v1.8 and will take advantage
I'm currently running the latest version of AstLinux, but running with
Asterisk v1.4. I would like to start exploring v1.8 and will take advantage
of the fact I simply need to switch CF cards and reboot.
That being said, I have a very basic (and probably simple) question
surrounding AstLinux. If d
Ahhh... as always, the problem existed between the screen and chair. I was
testing by turning off my modem; I didn't leave it off long enough. Sure
enough, leave it off for > 3m and SIP monitor works like a charm.
Many thanks!
Shamus
On Thu, Jul 21, 2011 at 07:15, wrote:
>
> Message: 4
> Dat
I've now upgraded to AstLinux 0.7.9 (Asterisk 1.4.42). I'm trying to get SIP
monitoring to work on my SIP trunk.
Through the GUI (Network-->Safe Asterisk & SIP Monitoring), I've added the
to/from e-mail addreses, including my SIP trunk (no peers) and "enabled" it.
I saved and rebooted.
>From the
I commented out the 2 lines (Answer, NVFaxDetect) and re-faxed
myself... no segfault. It does appear NVFaxDetect is the culprit. Has
anyone else seen a similar issue?
cheers,
S.
> Lonnie,
>
> This segfaults every time the incoming call is a fax. There is no
> problem if it is a regular voice c
Lonnie,
This segfaults every time the incoming call is a fax. There is no
problem if it is a regular voice call... that has been and still is
working normally. I will try commenting out the NVFaxDetect line and
report back...
For reference, here is my extensions.conf:
[incoming]
exten =>
Lonnie,
I modified extensions.conf as suggested, but am still seeing the
segfault error. Below are snippets from /var/log/messages,
/var/log/asterisk/messages and the asterisk CLI. To test this, I used
my own fax machine connected to my WCTDM card and dialed my own
number.
This leads me to suspec
plain; charset=us-ascii
>
> Shamus,
>
> I suppose it could be a dying CF card.
>
> Unless this thread generates other reports with similar segfaults, it seems
> unique to your setup.
>
> In your asterisk config, are you using CURL() or system() calls, or other
> thing
erisk Modules", your asterisk
>>> manager.conf file should contain the following section:
>>> --
>>> [webinterface]
>>> secret = webinterface
>>> deny = 0.0.0.0/0.0.0.0
>>> permit = 127.0.0.1/255.255.255.255
>>> read = command
&g
anager.conf file should contain the following section:
>>> --
>>> [webinterface]
>>> secret = webinterface
>>> deny = 0.0.0.0/0.0.0.0
>>> permit = 127.0.0.1/255.255.255.255
>>> read = command
>>> write = command
>>> --
>>> If
; permit = 127.0.0.1/255.255.255.255
> read = command
> write = command
> --
> If not, you can add it.
> Lonnie
>
> > On Jan 17, 2011, at 10:15 AM, Shamus Rask wrote:
> > I've been using AstLinux for > 1 year now and it has been wonderfully
> > stable
I've been using AstLinux for > 1 year now and it has been wonderfully
stable. Recently, I've started to receive the following error every ~40
hours:
Jan 17 09:54:54 pbx user.info kernel: asterisk[1713]: segfault at 62616237
> ip 400a5625 sp bebfb29c error 4 in libuClibc-0.9.28.so[4006e000+44000]
Lonnie,
Thank you... that did the trick. Much appreciated!
Shamus,
CURL is a function, not an application, hence it returns a string and
is called via ${CURL(...)}
Try this...
[macro-features]
exten =>
s,1,Set(rtn=${CURL(http://ekonserver.lan/aastra/push_notify.php?e
exten => s,n,CURL(
http://ekonserver.lan/cisco/push_notify.php?ec500=${DB(global/ec500)}&dnd=${DB(global/dnd)}
)
cheers,
Shamus
--
Try using CURL as asterisk functions are case sensitive.
Lonnie
On Aug 7,
I'm using AstLinux 0.7 and have tried to included the dialplan application
Curl() in extensions.conf. On execution, I get the following error:
[Aug 7 22:11:46] WARNING[4182]: pbx.c:1849 pbx_extension_helper: No
> application 'Curl' for extension (macro-features, s, 1)
>
I've tried using curl fro
$uri = "Init:AppStatus";
$xml .= "\n";
$xml .= "\n";
}
$xml .= "\n";
push2aastra($aastra, $xml);
push2cisco($cisco, $uri);
?>
I use PHP to drive my aastra phones. I go both ways with them. im using
> mini-httpd so that I can have menus on the
In the past, I have install Asterisk on a full-fledged Ubuntu-based server.
This has allowed me to create several AGI scripts that I use for passing XML
to my Aastra phones to turn on/off/blink lights for different event
notifications. I would like to enable similar functionality on my AstLinux
ins
e original
default name.
Thanks for all of your help,
Shamus
> ps -w
>
>
> On 7/14/10 7:55 PM, Shamus Rask wrote:
> > I updated user.conf and rebooted--I was immediately able to see all of
> > the mDNS advertised services (IAX, SIP, SSH, TFTP...). However, within
> >
u will have three
> occurrences, the user.conf will supersede all others.
>
> If you do enable this, be sure this AstLinux box is behind another router
> to block the mDNS broadcasts.
>
> Lonnie
>
>
>
> On Jul 14, 2010, at 12:21 PM, Shamus Rask wrote:
>
> > Th
ed, add to your user.conf...
>
> Network tab -> Advanced Configuration: User System Variables: {Edit User
> Variables}
>
> ADNAME="AstLinux PBX"
>
> (or whatever name you want to advertise) and reboot.
>
> Lonnie
>
>
>
> On Jul 14, 2010, at 10:34 A
I have AstLinux 0.7.2 running and noticed on first install (prior to using
web-interface to partition CF card) that there was an mDNS responder that
advertised SIP, IAX and SSH among others. I've since used the web interface
to partition the CF card, and now there is no services being advertised.
I've followed the end-user documentation on doc.astlinux.org and also
reviewed the original AstLinux user guide written by Kristian.
The new documentation makes no mention of using a separate USB drive; when
followed, my CF card was partitioned into 3:
> pbx ~ # df -h
FilesystemS
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