You don't install it as such, you just include the files from your php
scripts.
On 19/03/2008, Carlos Carvalhar [EMAIL PROTECTED] wrote:
Hello,
How do I install phpagi?
http://phpagi.sourceforge.net/
I couldn't find any info about setup in that site, and I couldn't email
the
I've used several hitachi dmp330's they work great, roam between wireless
access points with no loss of audio or connection for that matter.
it will be a great shame if hitachi stop producing them, they are the most
reliable wireless sip phones i've come accross... stay well away from
pirelli
Hi everyone,
I'm trying to get calls to record with the following setup:
Using phpagi originate is called from a web application:
$asm-originate(Local/ . $row['extension'] . @sip-standard,
$row['phone_number'], sip-standard, 1, , , 7000);
The agent being called is extension Local/[EMAIL
You can get incoming numbers from voipon.co.uk and a load of other companies
in the UK... 0800 is free for them to ring but you have to pay for the call,
you can also get regional numbers which are charged as a local call for them
- stay away from 070 numbers though.
2008/9/29 Babcock, Michael
costing
them much more.
Gordon
On Sep 28, 2008, at 10:26 PM, Geraint Lee wrote:
You can get incoming numbers from voipon.co.uk and a load of other
companies in the UK... 0800 is free for them to ring but you have to pay
for the call, you can also get regional numbers which are charged
Hi all, I've done this a few times with other PBX's but swyx has stumped me!
I'm having some trouble getting Asterisk connected to a Swyx system using a
sangoma A104dx... currently the setup is:
BT - Swyx
The above setup works fine... what i'm trying to achieve is
BT SIP Trunks - Asterisk - Swyx
brilliant idea - except it would be a sunday morning and another problem
the handsets that come with swyx aren't sip compatible :S
Cheers
Geraint
2008/10/6 Gordon Henderson
[EMAIL PROTECTED][EMAIL PROTECTED]
On Mon, 6 Oct 2008, Geraint Lee wrote:
Hi all, I've done this a few times
Linksys SRW248P or something like that... something from linksys anyway are
quite capable of all you mentioned... maximum 24 port powered though iirc.
Geraint
2008/10/6 Ken D'Ambrosio [EMAIL PROTECTED]
Hey, all. We're rolling out VoIP, and I'm wondering about PoE
recommendations, as we're
yes, thats the one i mean, 224p, the one i mentioned isn't capable of vlans
properly (which was strange, since it said it did)... i never had any
problems with them powering phones and cisco access points.
2008/10/6 Chris Bagnall [EMAIL PROTECTED]
We've used Linksys SRW224P units at quite a few
I don't mean to be a pain, but i could really do with a heads up on this...
does anyone have ANY ideas? I've trawled through google and come up with
nothing except for questions with no answers...
Cheers
Geraint
2008/10/6 Geraint Lee [EMAIL PROTECTED]
Hi all, I've done this a few times
instead of
16
job done!
Geraint
2008/10/7 Geraint Lee [EMAIL PROTECTED]
I don't mean to be a pain, but i could really do with a heads up on this...
does anyone have ANY ideas? I've trawled through google and come up with
nothing except for questions with no answers...
Cheers
Geraint
2008/10
Hello there...
Noticed some strangeness going on with mixmonitor and chanspy, the called
(External SIP) party seem to be responding before the calling party
(Internal SIP) on call recordings and also when you listen in using chanspy.
as far as the agent (calling party) is conserned the
Doesn't look like anyone has any suggestions though, guess it's time to play
until it's fixed then :)
2008/12/2 Thomas Kenyon [EMAIL PROTECTED]
Geraint Lee wrote:
Hello there...
Noticed some strangeness going on with mixmonitor and chanspy, the
called (External SIP) party seem
solved!
Cheers
Geraint
2008/12/2 Thomas Kenyon [EMAIL PROTECTED]
Geraint Lee wrote:
Hello there...
Noticed some strangeness going on with mixmonitor and chanspy, the
called (External SIP) party seem to be responding before the calling
party (Internal SIP) on call recordings and also when
use deadagi on the h extension maybe?
Cheers
Geraint
2008/12/10 Martin Tirsel [EMAIL PROTECTED]
Hello,
I am googling for a while but google seems to be broken today, no
solution yet :D I have a PHP script which needs to be started after
Dial() has ended. If there is no answer, busy, etc.,
nload will show you current bandwidth usage, but i guess that isn't what
you're looking for?
http://sourceforge.net/projects/nload/
Cheers
Geraint
2008/12/11 Shaun Wingrin [EMAIL PROTECTED]
Hi,
Would like to run the software to monitor the quality of the bandwidth.
Suggestions welcome?
just an idea, could it have something to do with DNS being unavailable, but
that wouldn't really explain why it would die when ADSL is down... h.
Cheers
Geraint
2008/12/11 Phil Knighton [EMAIL PROTECTED]
Hello
Looking for some help with a rather odd problem. We have Asterisk 1.4.10
2008/12/11 Dave Fullerton [EMAIL PROTECTED]
Brent Davidson wrote:
I have several branch offices all running Asterisk PBX's that register
to each other via SIP so that calls can be transferred from office to
office. Everything is working great on the office to office transfers,
but I'd
learn to use google.
http://tldp.org/LDP/Bash-Beginners-Guide/html/sect_02_02.html
2009/1/25 David @ULC ucoms2...@gmail.com
*1) What name I have to save it.Like what extension ?*
3) How I save it ?
*2) How to run it to execute it ?*
Should i do
vi autobatch
and then type and then
Could you not use some iptables to do this? I don't know the exact command
you'd need but it could work something like...
If the destination port is 5060 and destination ip is xxx then route via the
default ip (so do nothing)
If the destination port is 5061 and destination ip is xxx change the
what about something along the lines of...
iptables -A INPUT -p udp --dport 5060 -j DROP
iptables -A INPUT -p udp -s 192.168.0.0/24 --dport 5060 -j ACCEPT
iptables -A INPUT -p udp -s 10.0.0.0/8 --dport 5060 -j ACCEPT
iptables -A INPUT -p udp -s 66.66.66.66 --dport 5060 -j ACCEPT
Cheers
well, you got the general idea :)
2009/2/9 Tzafrir Cohen tzafrir.co...@xorcom.com
On Mon, Feb 09, 2009 at 11:09:34AM +, Geraint Lee wrote:
what about something along the lines of...
iptables -A INPUT -p udp --dport 5060 -j DROP
iptables -A INPUT -p udp -s 192.168.0.0/24 --dport 5060
Still doesn't work but i'm guessing it's to do with not being friends with
Michael?
2009/2/9 Dean Collins d...@cognation.net
http://tinyurl.com/c4qbcj
is that better for you?
Cheers,
Dean
-Original Message-
From: asterisk-users-boun...@lists.digium.com
.conf all the way, purely because i only noticed that extensions.ael even
existed a couple of months back, i should pay more attention really :p but
until it's broke, i can't be bothered to fix it.
2009/2/10 Alan Lord (News) alansli...@gmail.com
Hi all,
I built my first asterisk using the
Yes it's possible..
When you install use...
./configure --prefix=/usr/local/asterisk2 or something like it.
I had to change astrundir (in asterisk.conf) as well.
One thing to watch out for is that if you run make samples it will overwrite
the ones stored in /etc/asterisk and not where you'd
Almost forgot, you need to make sure you bind each instance to either it's
own IP address or different ports on the same ip, i used 2 IP's for it and
never hda a problem.
2009/2/24 Geraint Lee gera...@gmail.com
Yes it's possible..
When you install use...
./configure --prefix=/usr/local
I do it for CDR, when using the originate command via the manager and
initiate a call to a phone which then connects to an agi script upon answer,
the cdr stops at the point of answer and there is no other created, which of
course is useless for billing customers - there may very well be a way to
should have thought of that one lol
Cheers for the tip... will be changing my setup to this lol
2009/2/24 Klaus Darilion klaus.mailingli...@pernau.at
Rilawich Ango wrote:
Hi all,
Is it possible to install more than 1 asterisk in a single server?
If yes, what do I need to set and take
yes, you need to make sure bindaddr is set correctly in iax.conf, sip.conf,
dundi.conf, manager.conf and any other files that might include bindaddr for
BOTH instances of asterisk, you can't allow one to bind to all ip's and the
other just to bind to one - it won't work.
2009/2/25 Rilawich Ango
If you're using a php i'd take a look at phpagi - there are others around
for various different languages too. our agents use twinkle with
auto-answer, the only reason they need to look at twinkle is if they need to
perform a transfer (that too will soon be done from the web browser), you
can do
any suggestion on that?
Regards,
Kurian Mathew Thayil.
On Tue, Mar 10, 2009 at 7:32 PM, Geraint Lee gera...@gmail.com wrote:
If you're using a php i'd take a look at phpagi - there are others around
for various different languages too. our agents use twinkle with
auto-answer, the only
If it's anything like the UK, it won't make a difference... for example:
o2 mobile number ported to orange mobile...
On most providers you still pay the o2 rate.
three mobile ported to o2...
you still pay the three rate (which isn't so good since it's far more
expensive than o2).
Cheers
are you sure calls from this provider are going to context 'default' ?
sip.conf
[procall]
type=peer
username=XX
secret=XX
context=default
2009/3/16 Bayardo Sanchez bayardo.sanc...@gmail.com
i create inbound number but i calling and send this error:
[Mar 16 11:41:12] NOTICE[30847]:
We can put about 9/10 calls using SIP/gsm through our BT Business Network
ADSL package connection (832kbit upstream, £65/month) before you notice the
quality starting to drop, but you could always get two connections and
bond them together into one using openvpn or some other method if you
wanted
what about relogging the information using:
Set(CDR(customfield)=${CDR(originalfield)})
i think?
who knows, i might be wrong with all of this but i guess it will work...
2009/3/17 Matthias Urlichs matth...@urlichs.de
Hi,
as German phone numbers are variable_length, I need to use direct
We've had no end of trouble with usb headsets on linux (especially cmedia
chipset), as soon as you touch the volume control the sound settings all
mess up... i'm sure there'll be an alsa seting somewhere which would solve
this but i'm not that clued up on alsa so opted for using standard 2
I disagree with your opinion on softphones, i think they're great, saved
thousands in cabling, switch and phone costs.
I've had 50 agents running diskless/pxe linux (fedora 8), firefox,
thunderbird and twinkle and never had any problems, in the next few months i
expect to have at least 250 agents
I think we can conclude that hardphones should be used if you cannot under
any circumstances loose the call (power goes down in building, phones still
powered by PoE switches on UPS) or if you prefer/don't mind spending the
extra on hardphones... and softphones if it doesn't make a difference.
Just noticed you said DECT headsets... so what i wrote had nothing to do
with them, but i've used them too i think, excellent quality, tested them
with an aastra phone and worked great.
2009/3/23 Geraint Lee gera...@gmail.com
hehe, nice.
i've used those headsets hooked up to an old 4400 (well
another reason
why hardphones aren't so good?
2009/3/23 Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
On Mon, 23 Mar 2009, Geraint Lee wrote:
but on that subject... plantronics all the way, they seem to realise
that agents will complain if the headsets hurt (too
Use the Local/ channel type(?)
Local/0123456...@outbound-route
2009/3/24 Nhadie nha...@gmail.com
Hi All,
I'm trying to use the orginate cmd.
I have it working if originate is from a user e.g. SIP/
originate SIP/ extension 987654...@outbound-route
What i'd like to be able to is
sangoma support are amazing, they've solved nearly all the problems i've
experienced with PRI, except for one which turned out to be a bug in SWIX
(some rubbish windows based voip pbx, full of bugs and generally crap!).
there also quite happy to log in to your systems and have a look themselves
i haven't understood any of this thread... but i'm going to throw
busy-level in sip.conf in to the mix... i have no idea if this is a useful
contribution... but i felt i should contribute something :)
2009/4/16 David @ULC ucoms2...@gmail.com
My SIP config is below :
[sip64]
type=peer
I haven't worked with the omnipbx's but i have with an alcatel 4400 and used
a sangoma A108 and A104.. the sangoma cards work perfectly and if you have
nay issues sangoma support are always more than happy to help - and they
actually know what there talking about as apposed to having someone
i'd use mysql... and i do use mysql for this...
2009/4/21 Sriram d_r_sri...@hotmail.com
My setup : Trixbox 2.6.1 TE410P running well .:
1. I need to store the CallerId of the PSTN caller with his language
preference so that next time he is played the prompt in his language that he
chose
Hi Guys,
I have a strong feeling the loads on my servers will be shooting up soon...
anyone got any idea how many calls i can expect to put through a
DL360:
Dual Quad Core 2.33ghz
4gb RAM with 1gb allocated for a ramdisk (call recordings)
This server is recording calls (mixmonitor), codec is gsm
Thanks for that, it's pretty much confirming what i first anticipated... my
intentions are as follows:
agents register with opensips, opensips clusters a set of call recording
servers which then connect to our border servers which will save cdr and
choose the sip/iax provider to send the call to.
Check you can run the script from th ecommand line and successfully send
email... have you considered using phpagi for your scripts?
2009/4/23 James A. Shigley j...@answeringserv.com
I have the below script that doesn’t seem to be working. I don’t know if
I have something in the script wrong
You could use 2 originate commands and connect both of them to a meetme
room?
But surely what you're trying to do is going to confuse the person anyway if
they don't hear anyone when they answer?
Wouldn't it just be better to play a message after party a answers and then
start ringing party b so
you probably don't want to record directly to mp3 as there will be an
overhead in converting the audio on the fly and this will probably break
your call recordings... you should either record in the codec you are using
for phone calls (i think?) or in .wav and then convert afterwards (correct
me
Content-Type: multipart/alternative;
boundary==_Part_39198_10808701.1229015737923
--=_Part_39198_10808701.1229015737923
Content-Type: text/plain; charset=ISO-8859-1
Content-Transfer-Encoding: 7bit
Content-Disposition: inline
2008/12/11 Dave Fullerton
that's a bit lazy isn't it? google and the list archive should reveal all.
2009/4/27 almidos...@gmail.com almidos...@gmail.com
Hi list,
Anyone knows how to get free VoiP-in numbers from USA or Canada, I
have found some links for example sipnumber.com but it does not run.
Also I want to know
have you checked /var/log/maillog to see what the error might be?
2009/5/22 David da...@linuxcrazy.com
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Here is mine if it helps;
[general]
format=wav49|gsm|wav
serveremail=asterisk
attach=yes
skipms=3000
maxsilence=10
silencethreshold=128
ignore me! i've just realised half this thread was deleted :)
2009/5/22 Geraint Lee gera...@gmail.com
have you checked /var/log/maillog to see what the error might be?
2009/5/22 David da...@linuxcrazy.com
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Here is mine if it helps;
[general
There is indeed... well i was about to say there was, but it turns out the
one i've got is an fxo adapter, have a look and see if sangoma have any fxs
adapters in the series, it seems to be called the usbfxo u100
2009/5/26 Diogo Saad diogos...@gmail.com
What is the easiest way to connect my
It might be worth clarifying what the question is, i'm pretty lost.
Cheers
Geraint
2009/5/27 Adrian Marsh adrian.ma...@ubiquisys.com
Noone can give me a clue on this ?
How Domains are used within Asterisk ?
--
*From:* asterisk-users-boun...@lists.digium.com
i quite like the aastra 55i phones, i find the sound quality is better than
the polycom sound stations on loud speaker, and handset quality is perfect.
2009/6/3 Christian Stredicke christian.stredi...@snom.de
Check out the snom 300 or the snom 820...
CS
-Ursprüngliche Nachricht-
Yes, that should work fine, just remember you need a crossover cable to go
from the a102 to the legacy system
2009/6/3 Jim Dickenson dicken...@cfmc.com
I have a potential client that currently has a T1 circuit that feeds into
an
Adtran 750. Their phone sets are connected to the 24 ports on
I don't quite understand what you're trying to achieve, but if it's a
firewall wouldn't using something like iptables make more sense and be far
more secure?
Cheers
2009/6/3 Lee Spenadel spena...@gmail.com
I’m trying to isolate the IP address of inbound calls to my switch over
IAX2. Is the
I personally find the snom phones to be generally ugly and
un-finger-friendly, in terms of reliability and quality, never had any
trouble, good phones all in all, i just can't get past the tacky look and
feel so don't buy them.
2009/6/3 Darrick Hartman dhart...@djhsolutions.com
On 06/03/2009
find the snom phones to look quite good compared to the
american and chinese brands, might be a european thing though :)
Zoa
Geraint Lee wrote:
I personally find the snom phones to be generally ugly and
un-finger-friendly, in terms of reliability and quality, never had
any
twinkle.
2009/6/15 Manoj Panicker - FOES manoj.panic...@emirates.com
Hi Guys,
Any suggestions on any open source soft phones that has IAX and
SIP support.
I would also like to some programming over it and interface it with address
book or LDAP in order to make the call making
is it just me or am i right in thinking this has nothing to do with
asterisk?
2009/6/19 ABBAS SHAKEEL shakeel.abbas@gmail.com
Hello
Actually i am required to make two application
1) that user use
2) that is deployed on server
Application for user will be just like the windows
)
and whatever the equivalent server would be (don't know as i've never done
it).
Good luck
2009/6/19 ABBAS SHAKEEL shakeel.abbas@gmail.com
Geraint lee
I also dont know .what kind of requirements are these :P
i am just looking if it can happen
On Fri, Jun 19, 2009 at 9:33 PM
I have used an nokia n95 with asterisk without any problems (except for the
actual phone deciding to restart itself every few hours - but that's nothing
new with nokias!)
Are you getting anything on the CLI that might point you in the right
direction when the call is attempted?
CHeers
2009/6/29
agreed.
extended o2 coverage would be very useful, especially for Wales!
I like the idea, i don't like the idea of paying, if they want mobile
traffic it should be possible to buy your own hardware controlled in the
same method as wireless AP's allowing you to connect for free to the service
and
or maybe i misread :)
2009/7/1 Mike Dent mcd...@gmail.com
2009/7/1 Geraint Lee gera...@gmail.com:
agreed.
extended o2 coverage would be very useful, especially for Wales!
I like the idea, i don't like the idea of paying, if they want mobile
traffic it should be possible to buy your
search the mailing list, this question has been asked and answered several
times.
But it's all dependent on hardware, codecs, bandwidth.
If you mix the right technologies there is no limit to how many calls you
could handle, you just have to do it in the right way with multiple servers
yes, when done correctly.
2009/7/13 gergis.rasmy gergis.ra...@gmail.com
i am asked to implement a call center of 50 seats for my company , and i
was wondering if Asterisk can fit this as a relaibale and low price system
is it mature enough for this task??
best regards
Gers
Take a look at:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Random
You should be able to do what you want with this, it obviously won't take in
to account the actual amount of people still in the queue (for example if
someone hangs up while on hold). I'm sure there'd be a way of
Good luck with the N95... my experiences of the N95 and SIP haven't been
great... the phone likes to restart... regularly. Nokia may well have fixed
these glitches by now though. Getting it configured was a bit of a mission
too... and as expected the battery life shoots down when it's enabled...
For someone who is developing an 'autodialer' you are asking for an awful
lot! I would recommend getting to grips with asterisk before even
considering developing a dialer...
question 1 - aren't you developing your own so why would you need
documentation for another? or... why not use the other?
MeetMe agreed, but depending on how many people you expect to be listening,
i think you can do this on a virtual server with minimal bandwidth, you
can probably do this very very cheaply, or even find someone that will host
it for free since it's non profit, unless of course you're talking about
Asterisk is perfectly capable of it, your limiting factor will be bandwidth
if you want to do it in-house... you'll obviously need enough bandwidth for
all of your callers to be able to hear... unless of course you'll be using
real phone lines, in which case you'll need to buy the appropriate
On another note... have you considered using a simple shoutcast setup
instead? There will be a way (many ways probably) to hook this in with
asterisk if necessary.
You may have better results if it's simply listening the callers need to do,
and depending on the audience that will be listening may
1) does the file exist
2) is it chmod'd to 755 (not sure if this matters though)
3) do you have something like #!/usr/bin/php at the start of the php file?
Cheers
Geraint
2009/9/17 Anahi Ludueña a_ludu...@hotmail.com
Hi people, I have the following dialplan:
[context]
exten =
look at Random()
2009/12/12 Landy Landy landysacco...@yahoo.com
Hello List.
I would like to know how I can use two or more service providers with
asterisk to be used randomly for ei, if an user tries to make a call I would
like to randomly use a provider. It doesn't matter where the call is
try looking in extensions.ael
On 25 June 2010 12:25, Eyal Goltzman egoltz...@gmail.com wrote:
Hi,
I have a trivial peace of dialplan for exten 100. I try to change it to
_1XX and the asterisk act according to a different (Default??) dial plan and
not the one I want? Is that possible?
it would be far easier to just use the source...
but...
yum search asterisk
might get you on your way, although i can't see anything that looks like
samples in there.
On 13 August 2010 19:08, Albert Bonomo apeto2...@gmail.com wrote:
Hi, I'm trying to install Asterisk with yum.
I have
Right, I'm baffled.
I have:
exten = s,1,MYSQL(Connect DB1 127.0.0.1 geraint xxx amis2)
exten = s,n,MYSQL(Query NORESULT ${DB1} INSERT\ INTO\ recordings\
(caller_number\,called_number\,date_created\,date_started\,in_use\,server_id)\
VALUES\
This is what I ended up doing, working fine now.
Cheers
On 18 August 2010 08:52, Nasir Iqbal na...@ictinnovations.com wrote:
Avoid to use MySQL dialplan application, instead write an AGI script for
this purpose
On Tue, Aug 17, 2010 at 4:59 PM, Geraint Lee gera...@gmail.com wrote:
Right
sherwood.mcgo...@gmail.comwrote:
On Wed, Aug 18, 2010 at 3:59 PM, Geraint Lee gera...@gmail.com wrote:
This is what I ended up doing, working fine now.
Cheers
On 18 August 2010 08:52, Nasir Iqbal na...@ictinnovations.com wrote:
Avoid to use MySQL dialplan application, instead write
i do this by having 2 peers setup, one has a call limit of 10 and uses g729,
the rest of the calls get sent to the second peer which uses ulaw.
all calls attempt peer 1 if there's channels available it uses it if not it
just moves through the dialplan to use the second one.
On 19 August 2010
to get accurate cdr's i just use a border server to send every call
through that logs cdr... doesn't matter how many times it gets transferred
internally the border server still gets accurate records of the whole
call.
On 27 August 2010 21:07, Benny Amorsen
i suppose that depends on the number of eggs and baskets you have... but i'm
guessing not many of either since you're considering using a desktop board
for this...
but, email sangoma support, they will tell you.
On 17 September 2010 12:47, John Novack jnov...@stromberg-carlson.orgwrote:
Is there a way to specify which IP address to originate calls from in a peer
on sip.conf?
I need to send calls from 10.1.3.10 which is a routed network through
openvpn, but it's using 10.39.0.10 which is a vpn IP address - the asterisk
box is the same box as the vpn bridge for the 10.1.3.0/24
86 matches
Mail list logo
