Just this week, I've installed an Asterisk solution for my current job, to
replace a single POTS line which was shared between voice and fax.
Before we go fully live with the solution, we need to come up with a solution
to what to do with the current POTS line. The company is very keen to retain
t
On 12/30/05 06:45 Eck said the following:
Thanks for the reply, I'll give that a try. Does anyone know why the
zaptel drivers insert a 5secs pause before dialing the last digit? there
is a digium bug report about this, but they wrote it off as they rekon
are you sure the pause is not caused
On 01/05/06 18:24 Igor Neves said the following:
Take a look ate pfsense.sf.net, its GPL and its one merge of m0n0.
Much better, take a look. :)
i think you're mistaken. pfsense is not under the GPL, but rather under the
BSD license. it is based on FreeBSD 6.0.
--
Regards,
Douglas Garstang a écrit :
Thanks all for the replies. I started working for a CLEC a few months ago and
we've chosen to implement Asterisk. I'm not sure if the fact that my boss is an
open source advocate is a good thing or not... ie yes it's great to work with
Asterisk and see all the featu
On Sat, 2006-01-07 at 18:45 -0800, Bart Fisher wrote:
> Redhat has a 'Hardware Discovery Utility' called Kudzu.
>
> When I change cards, kudzu pops up and ask to remove/config the card.
> Most of the time kudzu has trouble recognizing the Digium Zaptel cards
> and calls them something wrong, like
Thanks for the suggestion, but I can’t seem to get
this to work for some reason.
When I dial my zap channel it does not seem to go beyond the
first priority.
I have setup the following just as a test, but never see the
output of Noop:
exten => _0.,1,Dial(Zap/g1/${EXTEN:2},,f)
e
Hi all,
We have 1760 working perfectly here with Asterisk for in and outbound calls except for:
1) Outgoing calls sound like they have silence suppression on them
(inbound calls are totally fine though). Have tried "no vad" and and
different VICs.
2) On outgoing calls on the Cisco console I get
Hi all, Newbie quest here. I have 3 PSTN lines (2430-2432) setup by the CLEC in a hunt group coming in to a TDM04B and 3 Grandstream gxp-2000's (say a,b and c) and Asteriskathome installed. I want all phones (a,b,c) to be able to take calls from the hunt group. Does this mean I must make 9
(This is an extension of an email I sent earlier, but I'm not sure if it
made it to the list or not I never saw it!)
We seem to be accumulating Cisco 8XX series ISDN routers as DSL becomes
more and more available in Australia and our clients upgrade.
Does anyone know if those routers can make
On Sunday 08 January 2006 05:48, Joseph Rothstein wrote:
> When I dial my zap channel it does not seem to go beyond the first
> priority.
> exten => _0.,1,Dial(Zap/g1/${EXTEN:2},,f)
> exten => _0.,2,Noop(${DIALSTATUS})
No application continues upon hangup unless there are special conditions which
/bin/sh: -c: line 0: syntax error near unexpected
token `;'/bin/sh: -c: line 0: `if [ -n "" ]; then if [ -f ];
then mv -f .bak ; fi; cat .bak | grep -v "alias char-major-250"
| grep -v "post-install torisa /sbin/ztcfg" | grep -v "post-install
wcfxsusb /sbin/ztcfg" | grep -v "alias wct
Hi Michael,
sorry for the late answer.
On Wed, 28 Dec 2005, Michael J. Tubby G8TIC wrote:
> 1. Call to Orange GSM mobile phone (effectively ISDN all the way) - works as
> expected
Can you please try attached patch 'cc1.patch' and recreate the log.
Because the ringing tone you hear does not come
I just found a Dialogic VFX/41JCT-LS (4 analog ports) in a drawer. I can use it in my house with asterisk at home project.Can I use that with asterisk?Where can I download proper drivers?--
---Erick PerezLinux User 376588http://counter.li.org/ (Get counted!
turn on callwaiting (*70) on the GXP-2000's and it will go to a second line.
On 1/8/06, Al Stery <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> Newbie quest here.
> I have 3 PSTN lines (2430-2432) setup by the CLEC in a hunt group coming in
> to a TDM04B and 3 Grandstream gxp-2000's (say a,b and c) and
Michael Graves wrote:
Surely there's something more to the truly SIP-aware device, such as
the Ingate IX66, that merits their use in some specific circumstances?
I know that I can stay with m0n0. The question still stands; are there
circumstances when something more is required? Would something
JCC wrote:
I don't get it. What is the advantage of using a GSM gateway? VOIP calls are
pretty inexpensive as they are now. Is the use of a gateway intended as a
backup incase a wired network connection goes down? I have being looking
around the net for information on this. Anyone out there using
If you mean the call keeps ringing, then the reason why is you have no
timeout in your dial statement. When there is no timeout the system
will never give up on line one so it can continue to the second
priority.
Thanks,
Steve
_
From: Joseph Rothstein [mailto:[EMAIL PROTECTED]
S
Wouldn't queues be a better approach to your problem?
Thanks,
Steve Totaro
_
From: Al Stery [mailto:[EMAIL PROTECTED]
Sent: Sunday, January 08, 2006 7:09 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] 3 PSTN lines, 3 IP Phones
Hi all,
Newbie quest here.
Dinesh Nair wrote:
we've got a number of TE410P 1st gen firmware cards. could we send them
in to digium for a firmware upgrade ? the cards were purchased in
october 2004 from atp in melbourne.
Yeah, you can. Contact RMA.
Kevin
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On Sun, Jan 08, 2006 at 07:50:53AM -0600, Mike Hammett wrote:
> /bin/sh: -c: line 0: syntax error near unexpected token `;'
> /bin/sh: -c: line 0: `if [ -n "" ]; then if [ -f ]; then mv -f .bak ; fi;
> cat .bak | grep -v "alias char-major-250" | grep -v "post-install torisa
> /sbin/ztcfg" |
> Asterisk drastically lowers barriers of entry in the field of
> commercial telephony systems. Besides, the wiki, the mailing
> list and the IRC channels make it relatively easy to get
> started with the system. This "no-pointy-clicky no-brainer
> interface" actually allows you to gain more
On Sunday January 08 2006 12:23 pm, Kerry Garrison wrote:
> > Asterisk drastically lowers barriers of entry in the field of
> > commercial telephony systems. Besides, the wiki, the mailing
> > list and the IRC channels make it relatively easy to get
> > started with the system. This "no-pointy-clic
Does anyone know if it's possible to set the incoming and or outgoing fax
baud rate in asterisk ?
Mark
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From: http://www.voip-info.org/wiki-Asterisk+Hardware
Dialogic
D/41JCT-LS
Note: The D/41JCT-LS is a full duplex card and is the first of the D/41
family that will work with Asterisk. Older D/41 cards like D/41(E)PCI
are half duplex cards designed for IVR type applications so they won't
work
Hi folks
You are welcome to try our (Xorcom)'s latest debs (for Xorcom Rapid, or
Debian Sarge in general)
"Unstable": Asterisk and AMPortal:
The repository is available at:
deb http://rapid.dotsrc.org/ unstable/
#deb-src http://rapid.dotsrc.org/ unstable/
The commands you are looking
Hi,
Is it possible to monitor conversation of logged in Agents? Currently I
am using ZapScan to monitor incoming calls, but I would like to monitor
individual agents.
raj
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Jean-Michel Hiver wrote:
Douglas Garstang a écrit :
Actually, I've found Asterisk to be a great experience. Not so much
because of the product itself (which is already great), but because of
the level of accessibility and the community around it.
Asterisk drastically lowers barriers o
Have you tried ChanSpy?
On 1/8/06, Rajkumar S <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Is it possible to monitor conversation of logged in Agents? Currently I
> am using ZapScan to monitor incoming calls, but I would like to monitor
> individual agents.
>
> raj
> __
yes i know, it's not
i have been battling with this problem myself
I thought of a theoretical solution (but I have zero programming skills
which means no possibility to code / try it) :
record the sound fax machines make when negotiating (specifically the
part where they try to negotiate anyth
Quoting Bogdan Moldovan <[EMAIL PROTECTED]>:
I have upgraded to Asterisk 1.2.1 and haven't gotten it to work yet.
Does it depend on some options in the Dial command?
I have also got the source now, and would like to know how it can be modified.
There is no documentation on what the structures d
-- Forwarded message --From: David Sarmiento Q. <[EMAIL PROTECTED]>Date: Jan 8, 2006 3:49 PM
Subject: Problems with R2 SupportTo: [EMAIL PROTECTED]
Hello!
I've been trying to compile R2 support into * 1.2 for two days now, and havent been able to do so.
I am using:
* spandsp
I have PolyCom phones in one office working perfectly, but in another
office with a new subnet, new server, new everything, the time does not
work. Everything else about the phones seems fine, but the time. If you
look at the internal webpage in the phone, it shows "clock". Our
server, which is c
If you are on 1.2.1, do:
In features.conf
[featuremap]
automon => *1 ; One Touch Record
atxfer => *2
disconnect => *97 ; this is the line you should add or edit
Bogdan Moldovan
MODULO Consulting
"The Future Is Not What It Used To Be"
http://www.modulo.ro
-Original Message
Hi all,
We are located in Belgium and using an asterisk as internal Pbx.
We have many problems with Zap lines, in fact, very often, Zap doesn't
release the line after a call or an unanswered call.
Any idea is welcome,
Olivier
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You may enable polarity reversal on the line, ask your telco about it
then add the following to zapata.conf
hanguponpolarityswitch=yes
answeronpolarityswitch=yes
you can also use a call progress detector such as
http://www.broadcastboxes.com/products/CP-2_lit.html
Diyanat
From: "Olivier
You can use ChanSpy() module to monitor
Diyanat
From: Rajkumar S <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussion
Errors-To: [EMAIL PROTECTED]
Return-Path: [EMAIL PROTECTED]
X-OriginalArrivalTime: 08 Jan 2006 19:40:13.0831 (UTC)
FILETIME=[58229970:01C61
Try running ngrep or (t)ethereal on your NTP server and see if you are even
getting requests for the time via (S)NTP.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Sun 1/8/2006 1:52 PM
To: asterisk-users@lists.digium.com
Okay then... next question... if I were to come up with a driver for
asterisk (either as hack in chan_capi, an extension to libcapi20, or a
driver for the kernel) to use the rcapi functionality of the cisco (and
other) isdn ta's, would anyone care to try it?
Thanks
James
(ps. Would I get flamed
I would have thought that if the 8XX would send the ISDN channels over as
SIP, that would be the easiest solution...
PaulH
- Original Message -
From: "James Harper" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, January 08, 2006 11:22 PM
On Mon, 2006-01-09 at 08:30 +1100, James Harper wrote:
> Okay then... next question... if I were to come up with a driver for
> asterisk (either as hack in chan_capi, an extension to libcapi20, or a
> driver for the kernel) to use the rcapi functionality of the cisco (and
> other) isdn ta's, would
Have your worked with any other PBX system? Learning Asterisk is extremely
product-centric.
Maybe. But it's out there, it's free. Plus the wiki has lots of
available documentation (including non asterisk stuff) and a great
community.
And actually, I'm not interested in the PBX functionality
I am getting the following error when starting:
loader.c:325 __load_resource: libspandsp.so.0: cannot open
shared object file: No such file or directory
loader.c:499 load_modules: Loading module app_txfax.so
failed!
When I load app_txfax.so and/or app_txfax.so; if these are
comment
On Mon, 9 Jan 2006, James Harper wrote:
> Okay then... next question... if I were to come up with a driver for
> asterisk (either as hack in chan_capi, an extension to libcapi20, or a
> driver for the kernel) to use the rcapi functionality of the cisco (and
> other) isdn ta's, would anyone care to
Make sure that /usr/local/lib is in your lib path.
/etc/ld.conf
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David C.
NicosiaSent: Sunday, January 08, 2006 5:25 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] spandsp
for 1.2.1 - libspandsp
>
> I would suggest extend the libcapi20. I already did such an extension
to
> libcapi20 to support the bintec remote-capi. This means with that
> libcapi20,
> each program (including chan_capi) can do remote-capi without any
> change...
>
That was my preferred option. It sounds like you did mos
On Sun, 2006-01-08 at 13:58 +0400, Jean-Michel Hiver wrote:
> Douglas Garstang a écrit :
> Actually, I've found Asterisk to be a great experience. Not so much
> because of the product itself (which is already great), but because of
> the level of accessibility and the community around it.
>
> As
> I would suggest extend the libcapi20. I already did such an extension
to
> libcapi20 to support the bintec remote-capi. This means with that
> libcapi20,
> each program (including chan_capi) can do remote-capi without any
> change...
>
That was my preferred option. It sounds like you did most
>
> I would suggest extend the libcapi20. I already did such an extension
to
> libcapi20 to support the bintec remote-capi. This means with that
> libcapi20,
> each program (including chan_capi) can do remote-capi without any
> change...
>
The more I look, the more I think that the bintec protoc
Doug,
I think that most companies dont know much about
asterisk. Thier current PBX works for them and they believe in "if aint broken
dont fix it". The only ones that seem to know about it are people that are
currently working as IT or PBX people and they came across asterisk one way or
ano
hey all,
I'm trying to double forward a DID , from PSTN to SIP DID then to my *.
Double forwarding my DID , but the Digits are not getting relayed correctly
... sending double digits and sometimes missing digits?
Is here anywhere I can put a delay on DTMF? Or a different solution for
this?
I use
The more I look, the more I think that the bintec protocol might be the
one required to talk to the Cisco anyway. Do you have those patches
somewhere?
According to the lists
http://www.c10.com.au/web/Products/ISDN/RVSCOM_Compatible_Routers.pdf
the CISCOs support the DCP of RVS-COM ... But if
I'm not so sure about that. I tried putting a Monitor() command right before
the Dial in extensions.conf that AgentCallbacklogin() uses to call an agent
back. I get a very small file of recorded audio before recording stops. I
assume the Queue application doesn't like Monitor() being called on i
Hello All,
I am just curious if anyone has interfaced the FWT
(LSP-350T) or similar FWT phones to Asterisk?
Regards,
Dinesh Birlasekaran
Network Engineer,
ComIT, Institute of Molecular and Cell Biology
61 Biopolis Drive, Singapore 138673
HP : 92962676 DID : 65869804 Fax : 677911
Tzafrir Cohen wrote:
"Experimental": Asterisk 1.2:
At the moment they are not that experimental anymore and should be ready
for use, but are not well-tested yet.
To use it, define both sources:
deb http://rapid.dotsrc.org/ experimental/
How does this compare with Asterisk 1.2.1.dfsg-1
Hello,
I'm interesting with asterisk, my plan is replacing our PBX office with
asterisk, now i've AAH and it's worked. Now i have a question, how can
limit the user to call international calling, linterlocal calling, and
mobile phone calling.
international calling started with = 00 in my country
Any one out there done the JiveMesenger jabber server?
www.jivesoftware.com/messenger/
I want to get this running to then do the next step of tie-ing it in
to the * server for presence & callerID screen pops.
Pursued their site a bit but never found a HOWTO or anything that
looked relevant.
Ap
If you try to compare Asterisk to other PBX's TODAY, Asterisk is running
somewhere close to 0%. Its simply too new still as most companies didn't
even begin taking a look until version 1.0 and even more with 1.2. Of course
this will change over time. We are selling several systems a month right
now
We seem to be accumulating Cisco 8XX series ISDN routers as DSL becomes
more and more available in Australia and clients upgrade.
Does anyone know if those routers can make the ISDN channels available
in a way that can be used by Asterisk? Preferably in a fairly raw form,
eg not SIP.
Thanks
Jame
I think it would be biggest is in consulting. The people that refuse or
cant to pay for call manager or Avaya's one. Example asterisk &
sugarcrm.com they work together. Thats really good to sell. They arent
in monster.ca they are banging on doors making $.
Make a buch of pre setup asterisk co
Eric "ManxPower" Wieling wrote:
JCC wrote:
I don't get it. What is the advantage of using a GSM gateway? VOIP
calls are
pretty inexpensive as they are now. Is the use of a gateway intended
as a
backup incase a wired network connection goes down? I have being looking
around the net for inform
Well, I sure hope it becomes enormous!
I moved from Los Angeles to a certain city that shall remain nameless a few
months ago to work for a CLEC. I've realised that while I love working with
Asterisk, I simply can't remain in this city (wy to small) and would like
to return to LA.
I'm tr
Mike Hammett wrote:
I've been Googling around for some time now (a few hours on dial-up).
I find all kinds of questions similar to mine, but either there is no
answer or the answer has nothing to do with the question. Hopefully
this post isn't another one of those.
Does Asterisk favor FPU
Sorry in advance if this is a FAQ...
I've got a working Asterisk setup based on [EMAIL PROTECTED] 2.2. I have a
TDM400
card with 2 FXS and 2 FXO ports; PSTN connections come in via the TDM
card.
I haven't been able to get inbound fax with spandsp and rxfax to work.
Occasionally an all-text fax w
Is there anything like FastCGI for Asterisk so that AGIs can have
persistent processes?
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> I think it would be biggest is in consulting. The people that
> refuse or cant to pay for call manager or Avaya's one.
That's certainly been our experience over the last 9 months or so we've been
involved with Asterisk. The bigger companies don't seem particularly
interested (or if they are, t
Either include the context containing the definition:
[second-context]
include => other-context
or specify the context in the dial command:
Dial(local/[EMAIL PROTECTED])
Matt wrote:
Hi,
What do I have to do to get local\number to work in a context?
It works from my [from-internal]... howeve
I have it working both here and at a client's place. I'm using Trillian Pro
as the IM client here, and Spark on the client's computers.
Seems to work pretty well. I seem to remember the steps I followed were:
1) Set up a manager account in asterisk (manager.conf)
2) Install Jive Messenger Server
http://www.voip-info.org/wiki-Asterisk+FastAGI
Mike Fedyk wrote:
Is there anything like FastCGI for Asterisk so that AGIs can have
persistent processes?
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David Sampson wrote:
Hello –
I have a non-PRI T1
[...]
How do I take incoming calls on these same channels?
You should get a PRI T1.
The minute you get close to capacity on this line you will run into
timing issues with incoming and outgoing lines competing with each
other. This proble
Consulting is fine, as long as I'm working for someone else. Setting up my own
company etc isn't really what I'm looking for. I don't want the risk. If there
aren't actual companies offering good paying positions, then there's really no
opportunities for me.
-Original Message-
From: S
Consulting is fine, as long as I'm working for someone else. Setting up my own
company etc isn't really what I'm looking for. I don't want the risk. If there
aren't actual companies offering good paying positions, then there's really no
opportunities for me.
-Original Message-
Just implemented a similar feature here - apparently the chanprefix
won't accept a full channel identifier, so I ended up dropping the last
character (this works for me since all the sip delivery we want to
monitor is to individual handsets - I won't be monitoring any channels
that are delivere
I am not sure why you are looking for jobs doing Asterisk work when less
than two weeks ago you were publicly bashing on the list.
Steve
>
> Consulting is fine, as long as I'm working for someone else. Setting
up my
> own company etc isn't really what I'm looking for. I don't want the
risk.
> If
Douglas Garstang wrote:
I have the following in sip.conf. It was my understanding that this configuration (ie with deny/permit) would only allow connections from hosts 192.168.10.4 and 192.168.10.5. That doesn't seem to be the case. Asterisk is accepting INVITE's from other addresses.
[a000901
Hi all
I am able to Port and Run asterisk On ARM based
Platform.
It was a really tricky process but manage to port
the asterisk in 5 hours. the was almost about 15 -20 MB.
And possibly I would be able to manage it 2.5 MB or
so.
It was really greate experience
Cheers
Mohan W
Embedded Ar
On Mon, 2006-01-09 at 09:50 +0530, Mohan Rao wrote:
> Hi all
> I am able to Port and Run asterisk On ARM based Platform.
>
> It was a really tricky process but manage to port the asterisk in 5
> hours. the was almost about 15 -20 MB.
>
> And possibly I would be able to manage it 2.5 MB or so.
>
Who? me? :)
-Original Message-
From: Steve Totaro [mailto:[EMAIL PROTECTED]
Sent: Sun 1/8/2006 8:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: RE: [Asterisk-Users] Asterisk Jobs
Hi All
Thanks for ur reply.My phone having 2 line with same
extension and also I configure voicemail if the user
not pickup the phone within 25 seconds for tht
extension but i want if my line 1 is busy then forward
the call to some other extension .my config is like
following.my phone having the a
okay, my needs have been met. Thanks much. Now, for the sake of Google,
could someone respond with how Asterisk likes single vs. HT vs. dual core
vs. dual cpu, etc.?
--Mike
--
Message: 15
Date: Sun, 08 Jan 2006 18:57:12 -0800
From: Mike Fedyk <[EMAIL PROTECTE
Thanx for the reply and the help, but i want to tell u , i'm a newbe for asterick . as to use any opensouce , i gone through the docs and the installation guide , read the few faqs as well. So finally i got installed successfully. To talk with GrandStream Budge Tone- 100 , do i have to install a
I have Asterisk 1.2.1 installed on FC4 box, a 2451E and 2440 TDM
Digium cards on PCI slots 2 and 1 respectively. When the system boots
up, it freezes when it reaches Asterisk, and if I go into interactive
startup and reject Asterisk, it boots up. When I enter the following
command "service a
This problem has been already corrected in Asterisk 1.2. See this bug:
http://bugs.digium.com/view.php?id=6009
On Monday 09 January 2006 00:51, Peter Fern wrote:
> Just implemented a similar feature here - apparently the chanprefix
> won't accept a full channel identifier, so I ended up dropping
Hi all,
I am trying to install the ODBCexec and ODBCquery functions into my
dialplan but I'm having difficulties in the installation of
app_odbcexec.c itself. I'm using Asterisk 1.2.1 and I've downloaded and
installed unixODBC. But when I try to run "make" after editing Makefile
and copying the ap
Ahh, I'm running r7233, I'll update to the latest rev to pull in the
changes, thanks Juan.
Juan Jose Comellas wrote:
This problem has been already corrected in Asterisk 1.2. See this bug:
http://bugs.digium.com/view.php?id=6009
On Monday 09 January 2006 00:51, Peter Fern wrote:
Just imp
Steve,
I think that doug was new and didnt know how to act nicely. We taught him
and he learned. We are all assests to this list. We are all human and we all
make mistakes. What happend happend. We have to look forward from now on.
David
I am not sure why you are looking for jobs doing Aster
Could anyone help me set up Asterisk in such a way that it makes SIP-SIP
transfers using the REFER / NOTIFY method according to RFC-3515 ?
SCANARIO:
- Asterisk registers with PSTN<->SIP VoIP provider "V" (Vonage) as a friend
- Asterisk is located in Europe, Vonage in located US.
- Asterisk acts a
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