# ifconfig xl0
xl0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500
address: 00:01:02:78:11:e8
media: Ethernet autoselect (10baseT)
status: active
inet 203.219.167.126 netmask 0xfffc broadcast
The solution to the problems with the Grandstream 1.0.4.39 firmware is
to use inband (in-audio) DTMF. Neither the RFC2833 nor INFO seem to
work.
Don't the Grandstreams send a DTMF 'F' INFO message on a hookflash?
Shouldn't be that hard to change chan_sip to register an 'F' as an
AST_FLASH
; FXS Port 1
context=local
signalling=fxs_ls
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
;
;FXS Port 2
context=local
signalling=fxs_ls
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
Change the signalling here to fxo_ls. Its gotta match what's in zaptel.com
I'm evaluating * to replace the crap set of peered smart phones we
have now in our small office, but I haven't been able to find out about
this anywhere yet: I need to know if * can discriminate _incoming_ FAX
calls on a voice line and route them to a specific extension?
Yes, it can.
Hello All,
I've mostly solved my DID problem from a few days ago. Apparenly the
lines weren't configured properly. Now heres the next question. 12 EM
wink lines from telco. I have them all plugging into an Adtran 750 with
FXS cards. The Adtran ports are configured DPO. How do I signal
Thanks John,
I think it is not that simple. I am not using a phone but a Cisco ATA.
The scenario: -
User--(Multitech VOIP MVP200 (FXS))--Internet--(Multitect VOIP MVP100
(FXO))--Cisco ATA--Asterisk--Any extension
Any reason you can't use the H.323 load for the MVP200? I've not tried it
exten = _.,1,Dial(Zap/1/$EXTEN)
exten = _.,1,Dial(Zap/1/${EXTEN})
Gotta put the name of the variable in brackets for it to work.
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To
Now, here's the real question: can you install it on a toaster?
It builds and runs on NetBSD, minus the hardware part (for the
moment)...so yeah.
Asterisk on NetBSD/Vax. Hrm.
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On Tue, 3 Feb 2004, Chris Albertson wrote:
Smallest Asterisk server? No. That old Gateway box must
be about 2 cubic feet. 1.5 ft^3 at a minimum. I've got one
that is about 0.2 ft^3 a factor of maybe 10 smaller.
Hehehe.. As far as Form Factor goes, I'm sure there are smaller boxes
out
Hi,
Please excuse me if my question seems too simplistic. I have been reading
the mailing list for some time and I am still a bit confused. Here is the
scenario that I would need to achieve and am wondering if asterisk is the
correct software to use.
(h323) (h323/SIP)
Is it possible to have the system outdial and take surveys. either by
receiving DTMF or voice?
Yup. Just have the system use the outgoing queue (see sample.call) and
have it call an AGI script upon answering.
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Steven,
Perhaps I should have posted my question differently to the list:
After installing the CVS version of Asterisk, I type, modprobe xct1xxp.
The machine accepts the command but the LED on the T100P does not flash.
How do I know that the T100P module has loaded correctly?
Do you see
I am looking for a good case to house my Digium PCI cards, I was hoping to
mount them in the front for cleaner access then in the back. Unfortunately
I
haven't found much, does anyone have a good recommendation for chassis to
use up to six digium cards?
Probably not cost effective, but i had
parkext = #700 ; What ext. to dial to park
Try removing the #
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Make sure you're using fxs_ks signalling for the FXO channels and also
make sure that your incoming lines support disconnect supervision.
Otherwise, * has no idea when the calling party hung up.
Hi Steven,
I have analog lines connected to the fxo lines of the Zhone channel
bank. All of your
The way I've seen it done is that the incoming fax signal is digitized and
compressed, then sent over the IP channel. It is done in real time. You
end up taking up 7k-14kbps instead of the 32/64kbps you'd use to pass high
enough audio quality to not irritate the modems.
Unfortunately, this
Also, it isn't very easy to 'test' either, as the staff at the 911 call
centre won't appreciate your testing, and at least in Australia, it is
some
sort of criminal?/illegal offence to call emergency for non-emergency
situations.
I had much the same thoughts. Currently my 911 code is just
[EMAIL PROTECTED]:~# modprobe wcfxo
/lib/modules/2.4.20/misc/wcfxo.o: init_module: No such device
/lib/modules/2.4.20/misc/wcfxo.o: Hint: insmod errors can be caused by
incorrect module parameters, including invalid IO or IRQ parameters.
You may find more information in syslog or the
For the development team to get * (and the zaptel cards) running on BSD
shouldn't take too much effort. Perhaps it's just a matter of finding the
right incentive? My only request would be that it be installed to match
BSD
filesytem standards (everything in /usr/local).
One of my next
On Fri, 1 Aug 2003 15:25:49 -0500
McAughan, Matt [EMAIL PROTECTED] wrote:
Have you setup the zaptel.conf and zapata.conf configuration files for
how
ever many ports you have on the card and then run the ztcfg -vvvc
command?
Since the module aren't loaded, config zaptel.conf,
There is the other hurdle of clients with existing PBX systems in place.
I've no idea how we'll cover this scenario as I'm sure most clients will
be
reluctant to replace their existing systems, unless of course asterisk can
be plugged into some of these systems?!?
Yes, it can. If the PBX
RE: [Asterisk-Users] newbie question - devicesHi,
So let me understand this better.
Asterisk can use SIP gateways which offer PSTN access. For example
www.iconnecthere.com, can be used?
Is this correct? And if it is, than any incoming calls through that
service, could be redirected by
Is anyone else having trouble accessing it with something besides IE on a
Windows box? Opera on Mac/FreeBSD/Linux just hangs at the login page, IE
on Mac and Netscape on Solaris Linux explode when loading
login_page.php.
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Could you tell me where mysql/errmsg.h is located on your
distribution? We can update the Makefile to look there for that
header.
Can't you use mysql-config to get the include and library paths? Granted,
you still need to make sure that mysql-config is in your $PATH, but it
keeps you from
Its another one of my If I only had time...damn this sleep thing ideas,
but I really wonder how hard/cost effective it would be to build an open
source IP phone or phone adapter (ala ATA).
In about 20 minutes of mulling and research, I figure you could do it for
about $40 in parts plus coding
On Tue, 19 Aug 2003, Michael Sandee wrote:
I guess you will need some software/mem/cpu/flash too? getting it on a
cicuitboard etc?
Software would be opensource...get a couple of people together to write it
RAM I missed, I thought the C400 had onboard ram, but it doesn't...so add
another $10.
Oh, and let's not forget that the traditional carriers are
not ignorant
of what is happening with VoIP or customer interest. There
is no doubt
that they are aware that if they don't find a way to deliver
this service,
someone else will.
No, if they don't find a way to deliver
Mike,
I opted for an integrated T-1 for 1 customer who needed about 12 lines.
I configured it with 12 lines voices and 768k data. Chances are you need
this kind of bandwidth if you need 12 phone lines. Combining it on 1 T-1
can make it a little more cost effective and of course one of
On my SBC phone, I used to hear a high-pitched chirp before the Call
Waiting beep (much like the first chrip of a V.90 modem negotiation tone)
when someone called in and I was on the line. Does this mean SBC was using
FSK to transmit caller ID on my line?
Yup. That's CallerID over Call
Again, not near my asterisk box so I can't check this out,
but is it possible to have the different ports drop into *
in a different context for each line? That way you could
just set up an 's' extension in that context for the
different attendants.
Yup. Set up different contexts in
On Thu, 2003-09-04 at 17:22, Peter Pauly wrote:
Does the Digium FXS card support modems (and Tivo devices)?
If so, to what speed have they been tested?
Assuming that you can do native zaptel bridging (Going from an FXS port to
an FXO port in the same machine), you should be able to get up to
[thread change, different topic]
is
How about a little tiny program that connects to a remote host, grabs
the contents of an MP3 stream, and pushes it into a FIFO locally? It
would be a raw TCP-to-FIFO stream, so mpg123 would be able to digest
it as if it was a local file. The program
allow this to happen. Do you know of any tools that convert ASF to
mp3?
mplayer/mencoder understands ASF, mp3 and lots of other formats.
Wont play an ASF stream, though...which is what he's looking for.
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Wont play an ASF stream, though...which is what he's looking for.
you're sure?
e.g.
mplayer http://live.atlas.cz/radio1/radio1-32.asx
works fine here.
Well, hell. Make a liar out of me. It wouldn't last time I looked.
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Hi all,
I've got myself all confused about the capabilities of *. I somehow
convinced myself (because I see a lot emails flying around about IP
phones)
that Asterisk works as a PBX and trunking gateway, but does not do voice
coding (i.e. TDM in, VoIP out). Does Asterisk work as a VoIP
On Wed, 2003-09-10 at 11:55, James Sharp wrote:
If I have a system with 1 machine to handle incoming H.323 calls and
then
multiple machines to distribute them to T1 ports over TDMoE, where does
the codec translation take place? Does it take place in the master
system
or does it take place
Is the max recommended still 2 cards, even in a Quad Xeon with
superduperwhizbang Hyperthreading? I'll be running incoming G.729 audio
out to TDM.
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So 3 or more TE410Ps in a system?
Is the bus mastering design that much of a significant improvement?
I would strongly consider the TE410P in this configuration and would be
interested in working with you to check scalability.
Mark
On Wed, 10 Sep 2003, James Sharp wrote:
Is the max
On Fri, 12 Sep 2003, Jim Paraschou wrote:
I have problem with a TDM40B installation.
When i modprobe wcfxs the error i get is the
following:
/lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No
such device
Hint: insmod errors can be caused by incorrect module
parameters, including
Interesting that it has 2 ports on it, and a speaker. The picture looks
a whole lot like a modem to me.
The real X100Ps look like a modem too. They have 2 ports and a speaker.
When I misplaced mine, I rummaged around looking for it and kept finding
it but putting it back in the pile thinking
Hi,
Can anyone suggest a good motherboard for the T/E410P cards ? Coz it
doesn't get inserted in the the regular P4 motherboards due to PCI slot
(32 bit) Any suggestions.
I'm an AMD Athlon bigot, I'm using the MSI-6501 dual AMD MB. Its got 2
64-bit PCI slots that'll take a TE410P.
Do they fall under FCC certification if they're built to the same
specifications as the ones from Digium? If I build my own T100Ps from the
schematics and board layouts that are available, are they legal to plug
into the PSTN?
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On Mon, 29 Sep 2003, Bill Leckey wrote:
I've been playing with the outgoing call spooling feature a bit lately
and it all works as it should with the exception of one irritation.
I'm mostly using SIP to talk to the phones and using G.723.1
I copy the call file into the spool/outgoing
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I've searched the mailing list quite extensively, but didn't come up
with anything promising (some things wer helpful, though). Does anyone
know if Nortel M Series (specifically the 2008, 2616, 7208, and 7310)
phones can be made to work with
Actually, if this was to be done, it might be an idea to do it with DNS, so
client machines would just do
Dial(IAX2/[EMAIL PROTECTED]/442071234567) and the DNS
system would resolve which machine is the correct target - no cleverness at
all required at the client end, so implementation would
It is that type of mechanism that enum uses and yes it was to solve a
similar goal, but in this case you need a 'route server' type system - in
particular as this is for IP routing of PSTN end points not on an IP
network.
A discussion about this came up a while ago. I suggested something
Which one would one should I use to solve my problem? Does an loadable
application give you more control than an AGI script?
If you want something that runs continuously (such as a listener process
or control process), it'll have to be a loadable module. AGI scripts only
get run when the
Below you will find, what I believe to be a typical setup with a T100P
card. My question is -
1. Is this correct?
Possibly. Depends on if you use a channel bank that can do add/drop and
you're not using a PRI.
You'll take your incoming T1 and go into 1 T100P and use another T100P to
feed
Exactly.
So...
I would need as you noted two T100P cards or a T400P. The T1 goes into
the * Server and the second port of a T400P goes back to the asterisk
server. Then the extensions get broken out from the Channel bank?
Geoff
___
Either way will work. Getting the T400 four port card gives you room to
grow, but getting 2 T100P single port cards saves you about $500.
Is this the only way to handle extensions... This turns a 4 port T1 card
into a 2 port card... Is this the suggested method?
Geoff
Ok.. Let me pose a question regarding this configuration.
Lets say you have the ISP bring in a full T1 and they split it half
voice half data. They would usually do this in a channel bank on
site... So in this scenrio... You have the Channel Bank from the ISP
where they split the channels.
UnixODBC. No need to rewrite everything for a simple DB change.
In what language is it written in? It would be interesting to at least
look at it and maybe convert it to use MySQL instead.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
Questions ...
OK - So, I've got Asterisk up, a Cisco 7960 talking to it, some mailboxes,
and extensions. All exciting.
Two questions:
I'm in a natted environment and need to utilize a SIP provider to make
calls
in the US. Currently I have Vonage in my natted network and it works
fine,
We have one other error (twice today) we get Out of trunk data space on
call number , dropping
How do I determine what is causing this error? we have a point-to-point
T1
between 2 * boxes, with 3 phone in the remote office. I have no idea how
the trunk could be out of space.
The
Quickie: Does anyone out there have experience with PRI delivery of ANI II
information?
Specifically, I want to know if it's possible from within Asterisk to know
if the inbound call (which may or may not be to an 800 number) came from a
payphone or not. I know with some 800 providers it's
Hi all,
We are thinking of changing our Nortel Meridian PBX to Asterisk. Before
we
jump into this we would like to know if we can support some important for
us
functionalities on Asterisk. We would like to know if we can
1. Have menu based voice mail with Asterisk?
I've got a couple of PRIs coming in from a SUMA 4 switch with some 800
numbers routed through it.
When the calls come in, I get the following message on the console and the
call never makes it through:
(800 number is fake)
Extension '8005551212' in context 'nonauthenticated' from '232102749585'
)
[pbx_config]
check 'show dialplan nonauthenticated'
regards
Martin
On Fri, 21 Nov 2003, James Sharp wrote:
I've got a couple of PRIs coming in from a SUMA 4 switch with some 800
numbers routed through it.
When the calls come in, I get the following message on the console
It seems that there's a non-printable character at the beginning of the
DNIS stream I'm getting from the SUMA 4 switch. Once I chopped that off,
everything works right.
Hi James,
Try to do
exten = _8005095639,1,Agi(ivr-main.pl)
Quoting James Sharp [EMAIL PROTECTED]:
*CLI show dialplan
Case 1 and 2 are ties in my eyes, except the channel bank would
provably be cheaper to upgrade to 8 lines. I am just afraid of the
channel bank. I just don't know anything about them. If I buy the
wrong crap, it gets really expensive fast, plus adds another layer of
complexity.
You could
Can someone give me an idea exactly what things are intended to be tested
via RADIUS, or some other AAA system?
Are we talking about building SIP/IAX/H323 entries from RADIUS?
This is where the PAM system I developed for * comes into play. I've got
most of it working at the moment, but I'm
It can be either.
Does this card only work as PRI or can it be used like a standard T-1
wired
to a PSTN Switch?
TIA
-Seth
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It's just my lowly opinion but I too must agree when it comes to the
consumer/soho (1 to 3 line) markets.
CAUTION!!, DANGER!! Marketing Hat On!!
Vonage, the most visible marketer of a voip consumer product must also
agree. Vontage offers an ip fax line. using cisco's ata. Vontage must
see
Run using a serial console
(http://www.tldp.org/HOWTO/Remote-Serial-Console-HOWTO/). No monitor,
VGA adapter, keyboard etc needed. Use SSH to log into the asterisk box
for any maintenance, etc. If the box gets hosed, connect the serial
port to a working PC and fire up minicom and your all
What about having your VoIP gateway system placing a 911 call to the 911
answering center in the appropriate region and when the 911 operator
answers, have a message say This is a 911 call from 123 Main Street,
Nowhere Nebraska then connect the caller to the 911 operator. Legal?
Maybe. Dunno.
Hi all.
Could it be possible that video frame buffering be causing problems
even if the computer is not running X ?
Yes. There are known problems with systems running with either a frame
buffer console or a serial console. For best results, run a plain VGA
console.
Occasionally I do NPA-NXX lookups for my local exchanges to see what other
carriers have prefixes in my area. I used to use telcodata.us, but they
seem
to have gone offline. Usually, after you find the carrier's name, you can
see info on the location and type of switch being used. I can't say
I am putting together a solution that will employ the Digium TE410P with
one T1 going out the PSTN and the other front-ending a PBX. The idea is
that based on a URL, Asterisk will dial an employee behind the PBX. When
the employee picks up, Asterisk will dial the customer (detailed in the
Andrew Kohlsmith wrote:
I would set the Enterprise Class bar at five 9's reliability
(about 5.25 minutes per year of down time) the same
as a Class 4/5 phone switch. This would require redundant
design considerations in both hardware and software.
To turn around, let's discuss what we need to
1. Moving a physical interface (whether a T1, ethernet or 2-wire pstn) is
mostly trevial, however what signal is needed to detect a system failure
and move the physical connection to a second machine/interface? (If there
are three systems in a cluster, what signal is needed? If a three-way
I had documented the Makefile modification in an email to the list. If you
search for Sparc in the mailing list, you should be able to find it. If
not, drop me a line and I'll see if I still have it.
I've got an Ultra 30 sitting here doing nothing. I'll see what I can come
up with for
FYI there is a way to do 911 its called E-911 enhanced 911
the user has to set it up with the local emergency services
to it and you setup your pbx to xmit the data.
There's PS/ALI (Private Switch Automatic Location Information) that's
quickly becoming state mandated for all PBX systems. The
If some channel banks don't support this, how on earth do they know when
the telco side of the call has hung up ?
They don't. They rely on either a timeout or the called party hanging up
to disconnect the call.
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Can anyone help me with the term that SBC uses to refer to disconnect
supervision? I have an Adit 600 channel bank which has helped improve the
disconnect detection time down to about 8 seconds. This is still causing
some
issues in particular with call progress enabled in * we are having a
I have a little more info on this. Following the suggestion of another
post on
this topic I tracked down an analog phone with lighted buttons powered by
the
phone connection. I directly connected the phone to one of my inbound
lines and
called it with my cell phone. Picked up the analog
,
James Sharp
ja...@fivecats.org
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk
On 10/07/2011 12:27 AM, Nasir Iqbal wrote:
Check firewall and NAT settings!
Monitoring sip and media flow from asterisk cli can help, use sip set
debug on, rtp set debug on and udptl set debug on
No NAT involved and I shut off IPTables. Still no luck. Debug shows
the SIP invite, RTP
On 10/07/2011 04:04 PM, Kevin P. Fleming wrote:
First, we can see that Gafachi's T.38 implementation still has some
breakage in it (although the problems are ones that Asterisk has been
taught to deal with). In its 200 OK to the T.38 re-INVITE, it has
On 10/07/2011 04:42 PM, Kevin P. Fleming wrote:
You shouldn't be *receiving* CNG, as you are the calling endpoint.
You're right. Hadn't even thought about that.
If you are seeing UDPTL packets containing T.38 CED, V.21 preamble, DIS,
etc. then something is badly wrong.
... and, that
On 10/08/2011 02:38 PM, Ryan Wagoner wrote:
I signed up with Gafachi a few weeks ago to use them for T38 as well.
I haven't had any luck getting it to work. I have been mainly trying
to use Asterisk in T38 pass through mode and have tested with a
Linksys SPA2102 and Zoiper. Gafachi basically
On 10/10/2011 05:35 PM, john Millican wrote:
Hello all,
Does anyone know of a good free/inexpensive 3G SIP client for the
iPhone? If anyone is using one that works good for them could you please
let me know.
I use VaxVoip on my 3GS and iPad. It works great over both 802.11 and 3G.
--
On 10/10/2011 10:31 PM, linux guy wrote:
On Mon, Oct 10, 2011 at 8:08 PM, Andresand...@telesip.net wrote:
I would recommend Acrobits. Not free but only a few bucks. It works fine
with ATT 3G.
This begs the question... which is more expensive (and where)...
making a regular cell call or
Then there's also the point where it makes more sense to drop a GSM card into
your Asterisk box and get a cheap unlimited mobile to mobile plan for a SIM and
use that to transit your calls to VoIP.
Although that won't help the original asker, though, since he mentioned
Verizon.
On Oct 11,
On 10/15/2011 05:31 AM, Michael C. Robinson wrote:
[Oct 15 01:48:02] NOTICE[3747] channel.c: Dropping incompatible voice
frame on SIP/2006- of format ulaw since our native format has
changed to 0x8 (alaw)
[Oct 15 01:48:49] WARNING[3750] pbx.c: Channel 'DAHDI/1-1' sent into
invalid
On 11/03/2011 07:20 PM, Nick Khamis wrote:
Hello Everyone,
Unlike going through DIDx, DIDLogic etc.., we have an option of
getting DIDs directly
from local telco Bell Canada. Currently our SIP Trunk provider
assigned a DID to us,
and as you know, they just redirect requests it to our PBX.
On 11/03/2011 09:16 PM, Nick Khamis wrote:
Hello James,
Thank you so much for your response. We just purchased an AudioCodes
MP124 for this job. And setting
up OpenSIPS as the proxy. As I mentioned earlier, Bell Canada is the
Telco here in Toronto. As for other
Telcos around the world, for
You're not going to get a telnet connection on port 5060, since that's tcp and
sip uses UDP.
Use tcpdump/wireshark on your office pbx to see if the packets are getting to
you. If not, then there's something wrong inbetween.
A firewall misconfig, perhaps. Or the unthinkable: your home ISP
On 11/14/2011 09:57 PM, sean darcy wrote:
Unthinkable!! Used wireshark: I can see the REGISTER packets going out
from the home router, but nothing from home:5060 shows up at the office.
Bummer. Now I get to think about how to set up special ports between
home and office. A great evening
On 11/16/2011 10:30 AM, eherr wrote:
But what is the correct physical setup of a CLEC.
Do you get rack space at a carrier hotel and equipment in there?
Do you get rack space at the local ILEC CO?; which is Verizon here.
What are the types of voice platforms used by CLECs?
Just as a point
I check in CLI
[Dec 6 17:46:58] WARNING[16264]: pbx.c:4088 pbx_extension_helper: No
application 'MeetMe' for extension (employees, 777, 1)
== Spawn extension (employees, 777, 1) exited non-zero on
'SIP/phone1-'
Plz tell me , where i am wrong in configuration.
Chances are you
On 12/11/2011 07:22 PM, Mike Diehl wrote:
Hi all,
I've got a customer who is bringing up a second Internet connection for fail-
over. I've configured a WRT54 with 2 LAN ports and arranged for it to fail
over when one of the routes is no longer available. That works just fine at
the IP level.
On 12/12/2011 12:35 AM, Mike Diehl wrote:
Actually, I've configured the phones to use DNS SRV records to find the Asterisk
server, and this works very well. The problem is that when the router fails
over, the phones IP address changes and this causes them to be unavailable
from Asterisk's point
Build Asterisk with ODBC support and then use the ODBC functions to do the
database dips.
On Dec 12, 2011, at 13:44, Douglas Mortensen d...@impalanetworks.com wrote:
Any suggestions from people who have done this before?
Thanks,
-
Doug Mortensen
Network Consultant
Impala Networks
On 12/15/2011 01:33 PM, Tarek Sawah wrote:
Hello List,
I have customer with a 40 Agents call center. and is looking to install a PBX
switch that can serve those agents.
As per my experience i suggested Asterisk as i have tested it with Call
Centers, however he has been advised not to use it
On 12/15/2011 01:43 PM, James Sharp wrote:
On 12/15/2011 01:33 PM, Tarek Sawah wrote:
Hello List,
I have customer with a 40 Agents call center. and is looking to
install a PBX switch that can serve those agents.
As per my experience i suggested Asterisk as i have tested it with
Call Centers
On 12/26/2011 04:15 PM, sean darcy wrote:
Thanks for the response. Home asterisk : 10.0.0 - Office: 1.8.8.0
So I thought I'd leave all the sip providers on udp, and move the
home-office to tcp.
And registration just work Just Worked over the default tcp registry
port - which I was surprised
On 01/03/2012 01:06 PM, Todd Routhier wrote:
Happy New Year to all!
Asterisk 1.8.x
I have a queue to which I add agent channels like SIP/300 dynamically
using the manager interface. Once logged in, there SIP/300 of course
rings when a call is distributed to them.
How can I also get the agents
On 01/03/2012 01:22 PM, Todd Routhier wrote:
Sounds perfect, I will need to look into how to blend them together like
that.
Put them in extensions conf like so:
[agentblends]
exten = bob,1,Dial(SIP/300SIP/12102263232@myprovider)
Then put Local/bob@agentblends into your queue.
I wonder
On 01/03/2012 09:42 PM, Raj Mathur (राज माथुर) wrote:
Hi,
I have a queue with a number of (static) agents. Is there an easy way
for an agent to indicate that she is away from her seat, so that her
phone is not rung when a call comes in? And the converse, of course: be
able to notify Asterisk
On 01/09/2012 02:44 AM, Eyal wrote:
Hi,
I try to create a new table using MYSQL command in asterisk.
This is what i write:
*Query resultid ${connid} CREATE TABLE IF NOT EXISTS conference_600
(id int(11) NOT NULL auto_increment, channel_id varchar(40),
number_in_line int(2), PRIMARY KEY(id))*
and
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