the same as well and it works with Asterisk.
-Bruce
On Fri, Jan 21, 2011 at 7:11 AM, Zeeshan Zakaria zisha...@gmail.com wrote:
Hi list,
For a client I am setting up a system which will use T1 PRI from Primus,
who offer
I have been using:
exec ('mv *.call /var/spool/asterisk/outgoing')
and for a long time it has been working just fine for me on more than one
websites. Just make sure the folder where you create the call files has
correct permissions and ownerships so that the file is successfully moved by
the
a linux
SDK http://www.opto22.com/site/downloads/dl_drilldown.aspx?aid=2890 . You
also can set the Opto hardware to send SNMP messages on certain conditions.
On Wed, Dec 15, 2010 at 4:23 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Hi list,
For a telecom project I need to setup a SCADA
Hi list,
For a telecom project I need to setup a SCADA solution. I don't have any
previous experience in this type of monitoring and automization. I'll be
using SNMP data from asterisk servers and endpoints. If anybody has any
suggestion which SCADA software can fit in such a VoIP solution, your
DTMF sent from cell phones are usually not well recognized at the asterisk
end. The main reason for this is that cell phones transmit out-of-band DTMF,
which by the time reaches an asterisk server traveling through cell towers,
their equipment, various VoIP carriers etc. is usually drifted away
How about 'show channels'.
As for filtering, you'll have to do it separately using a format like:
asterisk -rx 'show channels' | grep 'your filter'
You can filter the output further using awk. But each filtering will take a
second or two based on what you are filtering.
Zeeshan A Zakaria
--
If 1.2 is working fine without any problem then why do you need to upgrade
to any newer version? I would suggest don't do it. If you really want to do
it just for the sake of doing it, upgrade to 1.4 only, which is the most
stable and well tested version of asterisk. Upgrading always causes
Its good to know the MATH function because it can do much more and also deal
with floating point numbers. However in your case a simple addition would be
suffice as other posters posted, or try Danny's GotoIf if it fits your
scenario.
Set(vgLabel=vg${MATH(${vg}+1,i)})
Zeeshan A Zakaria
--
Unsuccessful attempts are recorded, however SIP-s is not easily doable on
asteridk 1.4. I tried once without any success. Maybe somebody who has
successfully implemented it can write a little how-to on it.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-11-01 4:48 AM,
)
On 2010-11-01 12:02 PM, Jamie A. Stapleton
jstaple...@computer-business.com wrote:
Only 100? We had a single server over 300.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria
*Sent:* Saturday, October 30, 2010 9:49
server over 300.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria
*Sent:* Saturday, October 30, 2010 9:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Under heavy attack
tips.
Cary Fitch
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria
*Sent:* Monday, November 01, 2010 12:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re
Too late, now switching to attack level: lethal :)
No, I am not one of these losers, and don't ever plan to be.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-11-01 1:49 PM, Jeff LaCoursiere j...@sunfone.com wrote:
On Mon, 1 Nov 2010, Zeeshan Zakaria wrote:
Hi
Finding and punishing the abusers is the real problem, specially when in my
country (Canada) where we generally don't like punishing people (or they get
away finding loop holes in the law, or thanks to their lawyers), how would
we catch people in other parts of the world and punish them?
My main asterisk server is under unusual heavy attack, and so far Fail2Ban
has blocked about 30 IPs, from various different countries. At this time it
is blocking about 1 IP address every few minutes.
Just wondering if anybody else is also experiencing unusually increased hack
attempts today?
My count has reached 100 for the day. The server serves doesn't serve
international calls anyways, I wonder how would it benefit any hacker in any
way.
--
Zeeshan
Sat, Oct 30, 2010 at 9:33 PM, Joel Maslak jmas...@antelope.net wrote:
No. It seems that opening up some sort of automatic
Yes, it works fine in 1.4.22 and 1.4.27 and 1.4.35.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-10-29 5:15 AM, Asterisk User an.asterisk.u...@gmail.com wrote:
Hello everybody,
does anybody know if BLF is correctly working in Asterisk 1.4.*? I'm
particularly
Two incidents in two weeks is not bad. I get 2-4 a day. There must be many
here with even more than that. You should start considering some safety
practices like disabling long distance and international calls by default,
put a cap on long distance and international calls even for genuine users,
Do you have canreinvite=yes anywhere? If yes, try setting it to no. Also
pasting your sip.conf here would be helpful.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-10-27 6:16 PM, Mike Diehl mdi...@diehlnet.com wrote:
There are NO ACL's in place, either at the
Chapters 4, 5 and 6 is a good start.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-10-25 2:01 PM, Jigar Joshi jiga...@gmail.com wrote:
Ok Thanks Guys.
Can you guyz suggest me upto which chapters orwhat are the chapters I should
cover for my requirement.
Because Its
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-10-23 6:24 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
I am using app_swift.
As a side note, demo on their website also generates sounds which at places
sounds like robotic.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta
I totally agree with Steve's wise advice. One should at least give himself a
week learning asterisk fundamentals and related Linux basics before jumping
into creating dialplans or setting up Telecom systems. Asterisk's official
book's first few chapters cover all the basics which every asterisk
www.pbxforall.com (beta)
On 2010-10-22 6:15 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 10/22/2010 04:05 PM, Zeeshan Zakaria wrote:
Hello list,
(Resending this email due to a typ...
Yes, the cards in question can handle some ports configured as T1 while
others are configured as E1.
--
Kevin
Do you recommend using wav files instead? Will there be any downside of
using wav?
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
--
_
-- Bandwidth and Colocatio...
--
Hello list,
I have been using Cepstral's 8KHz voices for my text-to-speech service for
some time now, and have been noticing that the voice quality is really poor,
doesn't matter what phrase I give it to convert. None of the other 8KHz
voices I have ever used were this bad. It doesn't seem good
?
On Oct 23, 2010, at 5:26 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Hello list,
I hav...
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar
I think you are the first person ever to ask this question. Of course you
can use them, they are royalty free for a purpose.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-22 5:53 AM, Aurimas Skirgaila a.skirga...@gmail.com wrote:
Hi,
I wonder if I may freely use the default soundfiles
I didn't know about Digium's cool case studies. Will my realtime virtual PBX
with partially javascript based GUI and Voice Reminder service fit into cool
case study?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-22 7:17 AM, Andrew Latham lath...@gmail.com wrote:
The sound files for MOH,
On Fri, Oct 22, 2010 at 8:26 AM, Zeeshan Zakaria zisha...@gmail.com wrote:
I didn't know about D...
_
-- Bandwidth and Colocation Pr...
--
_
-- Bandwidth
a lot to provide resilient networks, and as far as any
apps on the server are concerned, you are only talking to a single interface
bond0 instead of eth0 and eth1.
Rob
On Mon, 18 Oct 2010 17:03:45 -0400, Zeeshan Zakaria zisha...@gmail.com
wrote:
I didn't desig
Hello list,
I need to do E1 to T1 conversion for a project, and was wondering if there
exists a card with both E1 and T1 on it. Or is it possible to use two
separate cards in an asterisk box, one for E1 and one for T1?
(Please don't mention aculab or adtran)
Zeeshan A Zakaria
--
Hello list,
(Resending this email due to a typo in previous copy)
I need to do E1 to T1 conversion for a project, and was wondering if there
exists a card with both E1 and T1 on it. Or is it possible to use two
separate cards in an asterisk box, one for E1 and one for T1? (Please don't
mention
://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux
On Wed, Oct 20, 2010 at 10:52 AM, Zeeshan Zakaria zisha...@gmail.com
wrote:
Hello list,
W
Maybe you should post this portion for your dialplan. I have done the same
thing several times and never had this timeout issue.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-21 4:08 AM, GBR Icasiano, Ryan A.
raicasi...@globalbridgeresources.com wrote:
Hi,
Here is the scenario:
1. 1st
I was thinking on the same lines, i.e. setup a server which will be
regularly updated with these bad IP addresses, and anybody looking to block
bad IPs will be able to get this list from here. For example when I get mail
from Fail2Ban (which I am getting more and more everyday now), a copy would
Hello list,
What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and
a not much busy website, i.e. getting 500-1000 hits a day.
Thanks,
Zeeshan A Zakaria
--
www.ilovetovoip.com
--
_
-- Bandwidth and
Any suggestions?
On Wed, Oct 20, 2010 at 9:52 AM, Zeeshan Zakaria zisha...@gmail.com wrote:
Hello list,
What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and
a not much busy website, i.e. getting 500-1000 hits a day.
Thanks,
Zeeshan A Zakaria
How about setting up a high availability cluster using DRBD and Heartbeat?
There is some good info on it on the Internet. In this type of setup you
have two exact same servers running in parallel, and only one has the
required services up. They keep themselves in sync. When the primary one
goes
?
On Mon, Oct 18, 2010 at 9:34 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
How about setting...
--
Best Regards
Rizwan Qureshi
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join
Hello list,
I need to know how to deal with a redundant network with only one asterisk
server, which is receiving registrations from the end points on both of its
ethernet ports. This means extension 201 is registering both from eth0 and
from eth1.
Is there a way/software which can act as a
Will OpenSIPs do the job?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-18 4:43 PM, Paul Belanger paul.belan...@polybeacon.com wrote:
On Mon, Oct 18, 2010 at 3:40 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Is there a way/softwa...
DNS SRV or a SIP proxy.
--
Paul Belanger | dCAP
paul.belan...@polybeacon.com wrote:
On Mon, Oct 18, 2010 at 4:47 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Will OpenSIPs do the ...
Any proxy would work, however I would re think your network design.
Re-registering the same phone, with the same extension, on the same
PBX is asking for trouble
tcpdump to find out the IP of this service. AMI uses TCP port 5038.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-17 3:37 AM, Dan Journo d...@keshercommunications.com wrote:
Nope,
Its a totally normal self-built Asterisk.
Dan
Zeeshan Zakaria zisha...@gmail.com wrote:
Do you use FreePBX
Do you use FreePBX by any chance?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-16 6:38 PM, Dan Journo d...@keshercommunications.com wrote:
Serious answer:
Looks like a process running asterisk -r. Do you have any sort of
AGI, cron j...
Thanks for lightning my day!
Is there any way
I never had this problem, and this is certainly not asterisk's fault.
Probably your conversion is not good. Can you email me a file and I'll do
conversion on my end, and if sounds good, let you know how I did it. Then a
script can be written to convert them all.
Zeeshan A Zakaria
--
For future I would highly recommend to have at least fail2ban installed.
This way sipvicous IPs will be blocked instantly before they could create
any damage. Also I prefer to limit International calling to only certain
limit, e.g. only for $10 per account, but this depends upon how your
business
Check sip_buddies table for the correct context entry.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-13 5:31 AM, Oguzhan Kayhan oguzh...@bilkent.edu.tr wrote:
Hello,
I have a default installation of asterisk 1.6.1.9-2
When i create a user in users.conf via asterisk-gui,
calls, voicemail
It depends upon whether you are receiving DTMF or sending, and whether you
are using a VoIP protocol or using DAHDI/Zaptel.
Could you explain a bit what type of setup you have?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-13 9:15 AM, Dan Journo d...@keshercommunications.com wrote:
Hi,
I would suggest first to make sure that asterisk is receiving DTMF fine from
your IP devices/phones. Do you have a test IVR where you can dial and press
digits and verify that asterisk is responding?
Once you are sure that asterisk is receiving DTMF fine, then you should ask
your provider what
Hi,
(Following is for asterisk 1.4)
For the forwarded calls, you should see two entries in the cdr, and this is
because a forwarded call is actually two separate calls. You have to look in
the channel and dstchannel fields of the cdr to match the call ids of the
calls to figure out which calls
You need to create a dialplan context to achieve it and then access it using
originate.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-11 5:54 AM, 施铁泉 justhin...@gmail.com wrote:
I use the action Originate，i want the called first ringing，the called
answer,callee ringing.it can achieve?
For a production environment, 1.4 is the most stable, and it has everything
one needs to setup a telecom platform. As per my understanding 1.6 never got
the same recognition for stability as 1.4, plus it doesn't have any
significant advantages over 1.4. The newer version 1.8 series might be my
Here is a presentation from Kevin P. Fleming, Director of Software
Technologies at Digium. Information might be old by now still gives a good
overview of what is new in 1.6:
http://www.asterisk-tag.org/2008/slides/Kevin-Fleming-Asterisk-Tag-2008.pdf
Summary of his presentation is as follows:
–
Hi list,
A few times I have been asked if I could do encrypted VoIP development, for
embedded systems, and in C++. And my answer has been in negative.
Now I am thinking I should start learning how to do it, but I have no clue
where to start from. I have been developing in Java for some time now,
You can use proxmox from proxmox.com. I am using it for the same reason you
want to use it. I have been testing it for some time now and it works great.
Proxmox is an excellent hypervisor and it is free. Easy to install and
simple to setup. Install it drom its ISO. Then you can download a OenVZ
Seems like anonymous SIP calls which end up in from-sip-external context
with a dead end. This is usually how hackers start their hack attempts.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-02 3:05 PM, bruce bruce bruceb...@gmail.com wrote:
Hi Everyone,
Like always, here are IPs from
Is your ISP doing DNS resolutions for you? If yes, then I also think it has
something to do with the DNS queries which hangs asterisk. But it should not
bring the server down.
On CentOS, caching name server should be very easy to install by doing:
yum install caching-nameserver
I don't remember
If your sip provider supports gsm, then it is fine to send them your
existing format, but I am sure by the time voice reaches an end user, it is
transcoded at least once or twice again, so you can never guarantee what
quality the end user is getting. I would stay with ulaw, as it has more
chances
Its a long and old thread, haven't read it all, but just to let you know
this happens when there is no reply from the DNS. So change DNS or install
it locally on your asterisk server. At least caching name server should be
installed.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-24 1:51
From what you explained it seems to me that your mobile provider has blocked
your sip communication altogether. Have you tried changing IP address of
your asterisk server? If changing IP works, then probably your provider has
blocked you sip communication by IP only.
Zeeshan A Zakaria
--
Have you tried removing option 'g' from your Dial command?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-20 7:45 AM, Arie Skliarouk sklia...@gmail.com wrote:
Hi,
I use asterisk with sip3000 device with sip-aho connected to PSTN and
sip-ahi connected to a phone.
When call arrives from
with
spa3000 device, not asterisk.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-20 11:02 AM, Arie Skliarouk sklia...@gmail.com wrote:
Hi,
On Mon, Sep 20, 2010 at 16:39, Zeeshan Zakaria zisha...@gmail.com wrote:
Have you tried removi...
Of course, with the same result.
--
Arie
Zeeshan
It means that fail2ban is not configured correctly on your machine. For me
it works fine, and in fact lately these registration/hack attempts have gone
up significantly, thanks to cloud computing I guess.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-17 5:28 PM, dave george
When making an outbound call, if sip peer is not registered, first it
registers itself, and then makes the call. This is why you don't see any
problem dialing out. For receiving, asterisk has to wait until the sip peer
registers, otherwise asterisk has nowhere to send the call.
I know the pain,
I prefer to keep qualify=on for all the extensions, as it gives you an idea
which extensions are going to give you trouble. For extensions with qualify
value greater than 300 ms you should definitely worry. For extensions at
2000ms delay or more, turning qualify off simply means to ignore the
Hi,
I went over your dialplan and though it looks fine at first glance, but
because I have no experience with Asterisk 1.6, so I would like to ask if
commas in mysql query are ok without escape character? In my asterisk 1.4 I
would type it like:
SELECT var1\, var2\, var3 FROM ...
Other things
As Kevin said, you need to define an 's' extension where the calls will be
answered. Seems like you are using default configuration. Open file
'extensions.conf' in /etc/asterisk folder and look for context named
[default]. If it is not there, create one and add something under it, e.g.,
[default]
You can do 'extensions reload' or 'ael reload' if you don't want to lose
real-time sip registrations. I only reload what is needed to be reloaded
instead of reloading everything.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-15 4:28 PM, Leif Madsen leif.mad...@asteriskdocs.org wrote:
On
In theory it should work but in real life it doesn't. Converting reliably
half an hour of speech into text is simply a dream.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-14 4:52 AM, Nickolay V. Shmyrev nshmy...@nexiwave.com wrote:
В Втр, 14/09/2010 в 14:00 +0530, DHAVAL INDRODIYA
voice
recognition engine should do it.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-14 7:09 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote:
is it possible with lumenvox i will purchase liceance
regards
Dhaval
On Tue, Sep 14, 2010 at 4:20 PM, Zeeshan Zakaria zisha...@gmail.com wrote
I use ${UNIQUEID} with ${CDR(accountcode)} and it works great. But this is
when you have an accountcode for each user. As the last poster suggest, you
can append it with date and time and it'll be truly unique and also help you
keep track of the recording.
Zeeshan A Zakaria
--
...@polybeacon.com wrote:
On Sat, Sep 11, 2010 at 9:40 PM, Zeeshan Zakaria zisha...@gmail.com
wrote:
Poster is having problem when he disallows anonymous sip peers. Do you
know
at all how FreePBX deals with anonymous sip peers? Obviously you haven't
yet
seen the dialplan for FreePBX.
It's very simple
Can you post what are you doing to see UNIQUEID? And also what version of
Asterisk you are using?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-14 12:11 PM, Dan Journo d...@keshercommunications.com wrote:
${UNIQUEID} is going to be realtivly unique certnely in the short term
I dont
Or
1234 = {
Verbose (ID is ${UNIQUEID});
};
:)
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-14 12:27 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria
*Sent
Do a 'show channels'. You can also do 'show channels concise' or 'show
channels verbose' for more details. In any case, it'll show you number of
active calls at the end of output.
Now some may point out to prepend 'core' before issuing these commands. I
prefer to be brief, and used to this
True, that is even better.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-14 12:49 PM, Steve Howes steve-li...@geekinter.net wrote:
On 14 Sep 2010, at 17:32, Dan Journo wrote:
I'm trying to view a list of the active calls to see i...
Don't?. 'core restart when convenient' will wait
Hello list,
Slightly off the list topic, but I hope I'll get some help here. Somebody
wants me to implement for his project a Cisco based VoIP system. I told him
that I specialize in Asterisk based systems, but he is not even aware of
Asterisk. The requirement of project is such that chances are
do it, it will only cause headaches”. It is
completely different than * with different terminology, design
considerations, etc.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria
*Sent:* Tuesday, September 14, 2010 2:56
--
www.ilovetovoip.com
On 2010-09-14 4:23 PM, David Backeberg dbackeb...@gmail.com wrote:
On Tue, Sep 14, 2010 at 3:56 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Now I have no previous experience with Cisco systems and don't want to
screw
up anything. Are th...
sometimes. Cisco supports SIP, but depending
It is simply not possible, though it might be in the distant future.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-14 1:50 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote:
Thanks Paul,
i think still i have some problem to understand , i mean to say that i have
30 minutes audio file
This is not elastix or FreePBX forum and asking non-asterisk related
questions here is misusing this mailing list. Allow anonymous sip is not an
asterisk feature. Look in the code in extensions.conf what it is programmed
to do and you'll figure out why it is happening. Or maybe post the code and
Mr. John,
This is not about policing and this is asterisk-user mailing list. Poster is
a FreePBX user. I am very well aware of Asterisk IS involved, but the fact
is this is not a FreePBX mailing list. If the poster examines the problem
code from extensions.conf, or post it here, it'll made him
I think this may be because ...
So you think, don't know. Maybe you knew if you knew the FreePBX code, or
bothered to look into it.
j
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
whatever you like.
--
Zeeshan
On Sat, Sep 11, 2010 at 2:43 PM, Jeff LaCoursiere j...@sunfone.com wrote:
On Sat, 2010-09-11 at 03:53 -0400, Zeeshan Zakaria wrote:
This is not elastix or FreePBX forum and asking non-asterisk related
questions here is misusing this mailing list. Allow anonymous
Actually it is a very easy to understand and fix issue, but looking at the
code taking care of anonymous sip calls is the key. Those who post third
party GUI related issues should at least post the underlying asterisk config
or code here, so the asterisk part of the problem can be fixed.
Zeeshan
On 2010-09-11 7:22 PM, Paul Belanger
paul.belan...@polybea...
[un top posting]
On Sat, 2010-09-11 at 19:30 -0400, Zeeshan Zakaria wrote:
Actually it is a very easy to understan...
Its not that he isn't receiving a response - its that his peer debug
statement isn't getting tripped because the peer
Some days ago in my lab I setup Proxmox, installed a CentOS 5.2 appliance on
OpenVZ, installed all asterisk related stuff (except dahdi), including php,
mysql, munin, other tools, set it up with a dialplan and it worked just
fine. Then manually made multiple copies of the folder where all this
Hi List,
When doing 'ael reload' on two servers, which are setup with asterisk 1.4.22
and 1.4.35 respectively, I am getting multiple lines of this strange error:
ERROR[15483]: ael.flex:647 ael_yylex: Unhandled char(s):
On three other servers with same versions of asterisk, i.e. 1.4.22, I don't
Thank you list for all the valuable input. Based on your input I have
decided to stay with 1.4 for now for the production systems.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-08-24 2:13 PM, Roderick A. Anderson raand...@cyber-office.net
wrote:
Gordon Henderson wrote:
On Tue, 24 Aug 2010,
is required
somewhere related to AEL, but where, I don't know.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-08-25 11:37 AM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria
Actually I have found the problem, and leanred some new stuff along with it.
Apparently all Linux files have a mime type information stored in them,
which can be checked using command:
file -i filename
For my extensions.ael, which I copied from a different server, the mime type
is 'text/x-c'
This info was useful. So now I have more than a year before I can think
about switching to a newer version.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-08-25 12:34 PM, Leif Madsen leif.mad...@asteriskdocs.org wrote:
On 10-08-24 09:59 AM, Gareth Blades wrote:
Zeeshan Zakaria wrote
wrote:
Use the command aptitude install tofrodos to install dos2unix. This
command get the file and clear the ^M.
Regards,
Rodrigo Lang.
2010/8/25 Zeeshan Zakaria zisha...@gmail.com
Actually I have found the problem, and leanred some new stuff along with
it.
Apparently all
Thanks Steve for clearing this confusion.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-08-25 3:16 PM, Steve Edwards asterisk@sedwards.com wrote:
On Wed, 25 Aug 2010, Zeeshan Zakaria wrote:
Apparently all Linux files have a mime type informati...
Linux files are just a byte stream
This is at least the third post under the subject 'Codec Choice' by the same
sender. Why don't you stay within your first thread? Does posting over and
over again increases chances of getting a solution? If so, then maybe I
should try the same, as seems like an increasing trend on this list.
Hi list,
I am planning a migration to virtual machines, and was considering with it
to move from 1.4 to one of the later versions. My and my clients' 1.4 setups
have been rock solid and I don't want to put myself into any unnecessary
trouble. Those of you with solid experience with all these
I think you asked this question earlier and there were good responses to it.
There is nothing more to it than what people already suggested.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-08-24 8:56 AM, Dan Journo d...@keshercommunications.com wrote:
Hi,
I think I already know the answer
the standpoint of moving forward and
reverting back, if necessary.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria
*Sent:* Tuesday, August 24, 2010 6:51 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
, but the result was the
same.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria
*Sent:* Tuesday, August 24, 2010 7:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Should I move
wrote:
On Tue, 24 Aug 2010, Zeeshan Zakaria wrote:
Hi list,
I am planning a migration to virtual mac...
Some of us are still using 1.2 because it's as stable and solid as it
needs to be...
Gordon
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