El Lunes, 7 de Marzo de 2005 21:45, Ramon Roca escribió:
> Thanks Max & Julian,
> Well, I'm still not getting out with my card, but at least I've
> clarified those basic concepts (NT & TE, ptp & ptmp,,, etc)
> Appreciated ;)
>
> Currently I use group = 0, TE mode, bri_cpe_ptmp
>
> In /var/log/m
I solved the problem by adding the bindaddress option to iax.conf file.
bindaddr=192.168.1.251 and everything works.
Bart,
> Hi everyone,
>
> THis is my second thread regarding the issue.(before I was having problems
> with accessing my email, which slow down my responses, sorry for that)
> My s
This worked for me on Fedora Core 3
pwlib - 1_6_6
openh323 - 1_13_5
asterisk-oh323 - 0.7.1
cp pwlib-v1_6_6-src.tar.gz openh323-v1_13_5-src.tar.gz
asterisk-oh323-0.7.1.tar.gz /usr/src/
cd /usr/src
tar zxf pwlib-v1_6_6-src.tar.gz
tar zxf openh323-v1_13_5-src.tar.gz
tar zxf asterisk-oh
On Mon, 7 Mar 2005, Fabio Margarido wrote:
> I'm an asterisk newbie and have just joined this mailing list. I have to
> use asterisk as a call agent that supports MGCP requests. I'm reading
> the documentation from asteriskdocs and voip-info.org but those cover
> more specifically only IAX and
On Tue, 8 Mar 2005, Adnan Ahmed wrote:
> I have a question regarding asterisk in asterisk is video confrencing
> is possible like meetme i am out of touch quite a long time so don't
> bother with my question if video confrencing is possible what kind of
> hardware required i already working on so
Hi all,
i have a problem with some cisco 7960. Yesterday i did a
firmware-upgrade from 3.1 (1.2) with P0S30203.bin as described in the
most documents. Now i get the message Phone unprovisioned and in
TFTP-Log i find the following line:
07.03.2005 19:58 :Timeout error sending P0S3-07-3-00.bin to
I've been running a pair of Sipura 841's with my business partner. So far
I've been very impressed for the money. Having multiple lines (with
multiple registrations) has also been really nice as we both have our "work"
line and "home" line off the phones. (And you can update them for 2 more
line
my means , how could use asterisk and ser in same box.
my ser support mysql database , so whether asterisk don't config user in
sip.conf ? and how to do I should ?
thanks a lot.
and I want to agent asterisk product in China Mainland, who can contect me.
Best Regards
Zhao Zigang 赵子刚
Alcatel Sh
On Tue, 2005-03-08 at 16:56, Brian Nehring wrote:
> Xlite for OS X actually.
bummer, I've been wanting to get it running under Linux.
>
>
> On Tue, 08 Mar 2005 15:00:24 +1100, Howard Lowndes <[EMAIL PROTECTED]> wrote:
> > On Tue, 2005-03-08 at 14:57, Brian Nehring wrote:
> > > I actually got X-
Xlite for OS X actually.
On Tue, 08 Mar 2005 15:00:24 +1100, Howard Lowndes <[EMAIL PROTECTED]> wrote:
> On Tue, 2005-03-08 at 14:57, Brian Nehring wrote:
> > I actually got X-Lite talking to the server, finally. I didn't have to
> > change any of my Asterisk servers... I just kept fooling around
Thanks Pete
I never looked in the channels directory of the source tree. Doh!
I have been using a separate source tree for asterisk-oh323.
On the strength of that, I modified the Makefile in the asterisk
channels/h323 directory and it compiled immediately.
The only problem now is with the linke
Hi Mark
Funny you should ask this question, I just spent yesterday integrating
building asterisk with h323 support to connect to a Cisco call agent.I
cant say if it will work for you but it compiles and loads nicely ! I will be
testing this evening
# cd /root
# wget http://www.voxgratia.
I like the Cisco phones because of their high quality
and XML directory interface. Great for home use. But
they are a bit hard to convert to SIP.
Polycom is also good. Take a look at [EMAIL PROTECTED] it
has a web interface for configuring Cisco phones and a
lot of useful home office features.
Thx for all yourideas Guys... It worked like a charm!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kristian
Kielhofner
Sent: Lunes, 07 de Marzo de 2005 04:57 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Hello *'s,
I have a question regarding asterisk in asterisk is video confrencing
is possible like meetme i am out of touch quite a long time so don't
bother with my question if video confrencing is possible what kind of
hardware required i already working on softphones setup with asterisk
includin
I'm not planning to use it but still this is really cool !!!
Thanks
regards
m.
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Hello,
Cedric Hans has released an UNISTIM channel driver for asterisk (stable).
You can download it at :
http://mlkj.net/asterisk/chan_unistim-0.9.2.tar.bz2
Copy of README :
This is a channel driver for Unistim protocol. You can use at least Nortel
i2004 phones with it.
Only few features are sup
Dear All,
I understand the excitement surrounding a service like Asterisk, and how
easy it is to jump in and ask a heap of questions. I also know how
frustrating it can be dealing with a 200+ post per day mailing list as one
of the question "answerers".
When I discovered Asterisk, I had a lot of
On Tue, 2005-03-08 at 14:57, Brian Nehring wrote:
> I actually got X-Lite talking to the server, finally. I didn't have to
> change any of my Asterisk servers... I just kept fooling around with
> X-Lite and watching the diagnostics log and it finally worked. I can't
> really say what fixed it, I do
Great! That is what I was looking for and within my price range. Thanks.
Ryan
- Original Message -
From: "Kristian Kielhofner" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, March 07, 2005 9:32 PM
Subject: Re: [Asterisk-Users] Recommended
I actually got X-Lite talking to the server, finally. I didn't have to
change any of my Asterisk servers... I just kept fooling around with
X-Lite and watching the diagnostics log and it finally worked. I can't
really say what fixed it, I don't even feel like I changed anything.
Oh well, thanks for
Hi
there
I have Asterisk
running beautifully on our test server. Over the past few days I have been
tearing my hair out trying to compile various versions of asterisk-oh323 on
various versions of pwlib and openh323.
pwlib is now up to
1.8.3 and openh323 is now 1.15.2 stable.
asterisk-oh
Ryan Burke wrote:
Wow.. what an awesome mailing-list!
I appreciate the input, I'm looking at the Polycom 300 right now and
debating on spending the extra $60 / phone to upgrade from the
BudgeTone. Chris, I like hte Sipura 2100 idea, but I need at least 3
phones and would like to stick with nativ
Now that I've finally got faxing working, I'm wondering if there's any
way to retrieve specific fax status messages from SpanDSP when sending
faxes through *. E.g. completed/error status, etc. If so, is there any
simple way to have this status dumped into * CDR?
Wow.. what an awesome mailing-list!
I appreciate the input, I'm looking at the Polycom 300 right now and
debating on spending the extra $60 / phone to upgrade from the BudgeTone.
Chris, I like hte Sipura 2100 idea, but I need at least 3 phones and would
like to stick with native IP phones vs run
Hello,
I would like to know if FreeBSD has drivers for TE405P cards submitted
to the ports. I have acquired a card and would really like to use it
with FreeBSD. The wiki said the drivers havent been tested yet. Have
there been any updates since then?
regards
Kavit
__
To all those still struggling with faxing
I don't know exactly _what_ fixed the problem, but I tried faxing again on a
later version of libpri, zaptel and asterisk (all CVS HEAD as of 1st March)
with spandsp-0.0.2pre10, and now everything works perfectly. Faxing in both
directions, different
Hello,
I would like to know if FreeBSD has drivers for TE405P cards submitted
to the ports. I have acquired a card and would really like to use it
with FreeBSD. The wiki said the drivers havent been tested yet. Have
there been any updates since then?
regards
Kavit
__
Try the FOP mailing list (you can subscribe thru their web site).
On Mon, 7 Mar 2005 16:45:31 -0600, Anton Krall
<[EMAIL PROTECTED]> wrote:
> Ive been playing around with FOP.
>
> What is your queues, agents and extensions config... also your FOP buttons
> for this..
>
> Ive got mine working wi
Hello,
I would like to know if FreeBSD has drivers for TE405P cards submitted
to the ports. I have acquired a card and would really like to use it
with FreeBSD. The wiki said the drivers havent been tested yet. Have
there been any updates since then?
regards
Kavit
__
J P Edmund wrote:
I cannot figure out how to configure the box to set the "from" address
to a correct domain, as my outgoing isp will not pass mail from
[EMAIL PROTECTED], as I expect it wouldn't.
You can pass an argument to sendmail to accomplish this.
mailcmd=/usr/sbin/sendmail -t [EMAIL PROTEC
Hello,
I use the Sipura 2100 it only has 2 ports but it is works just fine.
I can even recv Faxes on it. with out any loss.
That is the one I recomend.
My Wife also has one of those ports and it works well for er also.
Then of course I put a punch down block and ran telephone wire all over the
hou
Hi Ryan,
I've used the BudgetTone 101 on several accounts and they certainly
aren't the best phone on the market, but they have so far been reliable
(touch wood) and are pretty straightforward to set up.
Call quality would probably rate at a 8 out of 10 on these phones, but
that's not much wo
I use Polycom 300's at my office and they are about $130 and work fine. I also hear that the sipura's are nice. They have an $85 model.
On Mon, 2005-03-07 at 20:32, Ryan Burke wrote:
Hello everyone, I've been watching this list for a while, but it is the
first time I've posted. I'ved decid
I recall you said thatI forgot.
When I rebooted, asterisk kept trying to restart. It bombed repeatedly
until I could start the wanpipes and ztcfg, then it stayed up.
I don't recall how to ask wanpipe to start at boot.
Jon
- Original Message -
From: "Kristian Kielhofner" <[EMAIL PROT
Hello everyone, I've been watching this list for a while, but it is the
first time I've posted. I'ved decided to setup a * server for my house and
will need 3 phones (one main, one for my wife, and one for my office). I was
wondering if there was a particular brand that people reommended? I'd li
1. how about something like a gotoif statement that compares CALLERIDNUM = EXTEN and if they match, goto a VoicemailMain(${CALLERIDNUM}) priority.
2. In phoneXXX.cfg, set your MWI settings. You can set the msg.mwi.1.callback to be your check vm extension, in my case 299. Also, you might want
I had this problem and solved by editing /etc/hosts and moved
localhost.localdomain to appear after my fully qualified domain name. I
am running sendmail on RH9.
127.0.0.1 domain.net asterisk asterisk.domain.net localhost
localhost.localdomain
Happy Hunting
--- J P Edmund <[EMAIL PROTEC
A quicker way to get to the missed calls list is to hit the down arrow button. Just exiting out just clears the display of missed calls and resets counter, the records are still in there. Look for the polycom remote reboot script and reboot the phones daily to clear the list for good so you d
If you're using broadvoice for your outgoing calls, you probably need to change
the entry in your sip.conf.
They made a change over the weekend. They announced it (as far as I can tell)
via a little note buried in the comments on the voip-info.org wiki. Thank god
I've been "trained" via this mai
set serveremail= to the address in voicemail.conf
On Mon, 2005-03-07 at 18:42, J P Edmund wrote:
I have been searching all over for the answer on all sources online and
have come to the conclusion that it must be rudimentary or I am asking
the wrong question.
I cannot figure out how to con
On Mon, 2005-03-07 at 19:19 -0500, Karl H. Putz wrote:
> >-Original Message-
> >From: [EMAIL PROTECTED]
> >[mailto:[EMAIL PROTECTED] Behalf Of Steven
> >Critchfield
> >Sent: Monday, March 07, 2005 6:08 PM
> >To: Asterisk Users Mailing List - Non-Commercial Discussion
> >Subject: Re: [Asteri
I have tried running 2 open source VXML browsers with * but without success:
1. sipXvxml - when started, it acts as a SIP endpoint. However I was unable
to make * to pass it a URI (which is posible i guess from a post i read in
this list). Also it seems to use the dsp as *, therefore if sipXvxml is
Thanks for both of the responses. I have 2 questions now, is there a better codec to use? I know if I can limit the voip delay that exists, it might help the problem some. 20ms is what is current but maybe 10 would help. What would be best, bandwidth is not an issue currently. This doesn't
Ah! I see that on the goto you are refering to extentions/priorities by name
and not number.. That one threw me off guard :)
Thx
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes
Sent: Lunes, 07 de Marzo de 2005 07:16 p.m.
To: Asterisk User
Check:
http://lists.digium.com/pipermail/asterisk-users/2003-October/025094.html
What you need is convert the document you need to send into a tiff file
and put
it in a directory. Which I believe is functionality you want to achieve.
Selon Justin Newman <[EMAIL PROTECTED]>:
We have something on
I tried many things and voice still doesnt work.
The setup is:
Phone -> * -> NAT -> Internet -> NAT -> Other phone
I would think this should work somehow. If not what will?
[EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
03/07/2005 03:33 PM
Please respond to
Asterisk Users Mailing List - Non-Co
--- J P Edmund <[EMAIL PROTECTED]> wrote:
> I have been searching all over for the answer on all
> sources online and
> have come to the conclusion that it must be
> rudimentary or I am asking
> the wrong question.
>
> I cannot figure out how to configure the box to set
> the "from" address
>
Kristian Kielhofner wrote:
Eric Wieling wrote:
Kristian Kielhofner wrote:
Goto(my-internal-sipphones,1234)
Goto(my-internal-sipphones,8005551212)
The first argument is the context, the next argument is the
number. Your internal sip phones should be able to reach each other
via "extension" dia
--- Steven Critchfield <[EMAIL PROTECTED]> wrote:
> On Mon, 2005-03-07 at 14:21 -0800, beonice wrote:
--- snip ---
> > Would it make more sense to write a custom
> application
> > in C instead, designing it to work sort-of like
> the
> > built-in app_voicemail.c and others? I do know C,
> but
On Tue, 2005-03-08 at 11:43, Anton Krall wrote:
> Wow, too professional for me hahaha can you explain to me the last part of
> the goto?
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes
> Sent: Lunes, 07 de Marzo de 2005 06:22 p.m.
>
Eric Wieling wrote:
Kristian Kielhofner wrote:
Goto(my-internal-sipphones,1234)
Goto(my-internal-sipphones,8005551212)
The first argument is the context, the next argument is the
number. Your internal sip phones should be able to reach each other
via "extension" dialing and (hopefully) be abl
Interesating approach Sounds very logical.. I guess it's a mix of all
the ideas given... You can validate the desired forward number using
variables and len just to check it's a valid one... And then use the goto
based on the phones context... As you put it, its flexible... Also, some
checks to
Kristian Kielhofner wrote:
Goto(my-internal-sipphones,1234)
Goto(my-internal-sipphones,8005551212)
The first argument is the context, the next argument is the number.
Your internal sip phones should be able to reach each other via
"extension" dialing and (hopefully) be able to reach outside n
Digium shipped me a replacement card, but they sent the wrong one, so they
fedex'd another and its just arrived.
Should be testing in the next two days (the box is in another state...)
The last I heard from Eric Bishop (on the 1st march) was that he had
received an updated card from digium, but
Mark F. Vickers wrote:
Tony Mountifield wrote:
In article <[EMAIL PROTECTED]>,
Eric Bishop <[EMAIL PROTECTED]> wrote:
Hi all,
Just got a brand new server and a Digium TE410P. I get the sequential
(knight rider) lights before loading the zaptel driver. As soon as I
load the driver all loghts go off
Howard Lowndes wrote:
On Tue, 2005-03-08 at 10:48, Anton Krall wrote:
Nice idea.. Now, also We would need to check the number of digitsentered, if
more than X, then call is an outside number, is less than X, then its an
internal extension..
Simple.
SetGlobalVar(DIGITS=4)
GotoIf($[${LEN(${EXTEN}) >
Was there any resolution on this I also have a TE410P in an box with an
Intel E7501 chipset?
-Vickers
Original Message
Subject: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4
server
Date: Tue, 8 Feb 2005 11:13:24 +1030
From: Peter Childs <[EMAIL PROTECTED]>
Reply
Wow, too professional for me hahaha can you explain to me the last part of
the goto?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes
Sent: Lunes, 07 de Marzo de 2005 06:22 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sub
I have been searching all over for the answer on all sources online and
have come to the conclusion that it must be rudimentary or I am asking
the wrong question.
I cannot figure out how to configure the box to set the "from" address
to a correct domain, as my outgoing isp will not pass mail fr
Tony Mountifield wrote:
In article <[EMAIL PROTECTED]>,
Eric Bishop <[EMAIL PROTECTED]> wrote:
Hi all,
Just got a brand new server and a Digium TE410P. I get the sequential
(knight rider) lights before loading the zaptel driver. As soon as I
load the driver all loghts go off. It appears the card i
On Tue, 2005-03-08 at 10:48, Anton Krall wrote:
> Nice idea.. Now, also We would need to check the number of digitsentered, if
> more than X, then call is an outside number, is less than X, then its an
> internal extension..
Simple.
SetGlobalVar(DIGITS=4)
GotoIf($[${LEN(${EXTEN}) > ${DIGITS}]?s-e
Steve Kann wrote:
What he describes is echo suppression. Because an echo canceller can,
generally, only remove some part of an echo, not the entire echo,
systems are generally designed to suppress the residual echo in some
circumstances. Old speakerphones had poor on no echo cancellation, so
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of Steven
>Critchfield
>Sent: Monday, March 07, 2005 6:08 PM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] Question about AGI vs. FastAGI vs.
>straight C/DB developme
On Mar 7, 2005, at 5:52 PM, [EMAIL PROTECTED] wrote:
Hi, all
I have two questions regarding usage of Polycom SP300 with Asterisk.
No sure if it
is Astersisk or phone related, though.
1. When dialing an extension, one has to perss Dial or Send on the
phone after
number is entered. Is it possible
Dennis Webb wrote:
This seems to be how AGGRESSIVE_SUPPRESSOR works. To make sure you
don't get echo, it does what a speakerphone does, mute the other party
if it hears audio from your end. There is a setting in mec2_const.h
for AGGRESSIVE_HCNTR=160 that says in the comments 20ms, I'm assuming
We have something on the way.
Regards,
Justin Newman
[EMAIL PROTECTED]
> - Original Message -
>
> > Does anyone know of a software SIP fax client? Something I can install =
> > on a PC which connects to the asterisk server and sends/receives faxes?
=
> > Something like XLite - but to fa
Hi, all
I have two questions regarding usage of Polycom SP300 with Asterisk. No sure if
it
is Astersisk or phone related, though.
1. When dialing an extension, one has to perss Dial or Send on the phone after
number is entered. Is it possible to avoid this and only enter the number?
2. This is
Got ya! With yourideas and Howard's... I think I can code some interesting
dialplans..
Thx Guys
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kristian
Kielhofner
Sent: Lunes, 07 de Marzo de 2005 04:57 p.m.
To: Asterisk Users Mailing List - Non-Commerci
Hi all,
Can anyone help me with a CAPI problem that I am having. I've got one
BRI trunk (will have 4 when it goes into production) and when one of
the B channels is in use (i.e. there is an incoming/outgoing call in
progress) I can't get Asterisk to answer the other ringing B channel
(Asteris
Nice idea.. Now, also We would need to check the number of digitsentered, if
more than X, then call is an outside number, is less than X, then its an
internal extension..
Sounds good?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes
Sent: L
Hi Mike
FOr a home solln, $1000 isnt overly cost effective and the payback period for
myself would be too long. The technology is interesting and as everyone says,
its **supposed** to work but I hate being the guinea pig.
The dock-n-talk is interesting and supposedly it does work ok, the signall
Please see the other thread -- it's a long discussion in this issue:
http://lists.digium.com/pipermail/asterisk-users/2005-March/092953.html
--Luki
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Steven Critchfield wrote:
On Mon, 2005-03-07 at 16:06 -0500, Eric wrote:
Hi Vinko,
MySQL blobs will store binary data, so you should be OK there. I'd
focus on whether or not storing the data in a variable is a good idea.
Typically, with any programming language, it's good practice to
keep varia
Afaik, caller-id name is not passed on between lecs and clecs (via SS7)
-- that's what I remember from a thread I read here in the past. That's
why most clecs maintain their own DBs, and that's also why it takes
weeks for callerid information to propagate, and why your name may show
up in differen
Hey there,
I'm an asterisk newbie and have just joined this mailing list. I have
to use asterisk as a call agent that supports MGCP requests. I'm
reading the documentation from asteriskdocs and voip-info.org but
those cover more specifically only IAX and SIP configuration. I'd
really appreciate it
On Mon, 2005-03-07 at 14:21 -0800, beonice wrote:
> Folks,
>
> I want to build a custom IVR for my setup. I've got it
> working (well, the bells and whistles are not there
> yet, but the basic stuff works) using AGI, but I'm
> worried about how well this will scale.
>
> I've seen references to Fa
Hello all,
I am looking at the possibility integrating Asterisk with our current Mitel
200sx. If this is possible what physical connection is made between the
Mitel box and * box? Then can a user choose if a call is go out VoIP or not?
Has anyone had any luck in doing this?
Thanks,
Scott
Anton Krall wrote:
Im looking for something like the article on the wiki where the user can
dial *21 plus the number or extension to forward to...
I don't quite get the internal on that yet but it's a mix of that and what
you just posted... Although the wiki just send the call to another
extension
In zapata.conf I have (not zaptel.conf, my bad):
busydetect=yes
busycount=4
And I meant BUSYDETECT_MARTIN in dsp.c. But what I have as compile option is
just BUSYDETECT.
You may need to change BUSY_THRESHOLD, BUSY_MIN, and BUSY_MAX there to suit
your needs. I did. It depends on the tones in New
Ive been playing around with FOP.
What is your queues, agents and extensions config... also your FOP buttons
for this..
Ive got mine working with agentcallbacklogin with no problems.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken Sandell
S
On Tue, 2005-03-08 at 09:14, Anton Krall wrote:
> Guys.
>
> Im trying to implement some kind of call forward or DND, I checked the wiki
> and there are some examples of call forwards but I was wondering if anybody
> has implemented one that will let you forward calls to SIP, IX or ZAP
> channels a
Im looking for something like the article on the wiki where the user can
dial *21 plus the number or extension to forward to...
I don't quite get the internal on that yet but it's a mix of that and what
you just posted... Although the wiki just send the call to another
extension so I would need t
Is there any chance you would be willing
to share your code with me?
Thanks,
Ken T.
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Chris HARIGA
Sent: Monday, March 07, 2005 1:06
PM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [A
Hey guys – I’m having a couple of problems
getting the Flash Operator Panel to work.
It seems like most of the features are working, though when
I have an agent login (via AgentCallBack method), the panel doesn’t update
that an extension has changed to agent.
I edited op_server.cfg
Just realized, if you don’t hangup
the handset and then press speakerphone a second time it disconnects the call.
Thanks anyway, just need to make sure you
place the handset in the cradle when you use speakerphone.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On B
On Mon, Mar 07, 2005 at 11:16:18PM +0100, Alfredo Sola wrote:
> Ok, I'll correct. I would like to be able to set in the Makefile
> which user I'll be using, so that permissions can be set accordingly.
> No
> problem AFAIAC if the default user is root. Perhaps time for an autoconf?
H
> -Original Message-
> Sure, but wouldn't LiveVoip be using PRI as opposed to PSTN?
> I dunno about in the US, but here (Canada) we've got
> switch-based CallerID and user-based CallerID. As long as
> you're using a PRI based line the user can fire both caller
> number and caller nam
Do you have the handset still off the hook when you do
this?
If the handset is on hook and hit the speaker button it
should hagup the call.
If the handset is off hook, it should revert back to the
handset.
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dean
colli
Just pick up the handset and the speakerphone will
turn off automatically.
If the handset isn't hung up already, just hang it
up and pick it up again. Hanging up the handset won't hang up a
speakerphone call...
Paul
- Original Message -
From:
dean
collins
To: Ast
Anton Krall wrote:
Guys.
Im trying to implement some kind of call forward or DND, I checked the wiki
and there are some examples of call forwards but I was wondering if anybody
has implemented one that will let you forward calls to SIP, IX or ZAP
channels alike? For example, forwardto another exten
Don,
All you need to do is configure your VM button on the phone via the cfg
files. You setup the phone to look to an FTP server for config scripts
and then setup the scripts to work as needed. Here is an example of
where you set this VM button from my file called -phone.cfg...
- Original Message -
From: "Jay Milk" <[EMAIL PROTECTED]>
You won't be able to send caller-id NAME with any PSTN termination.
That's just not how that works. Each CLEC looks up the name in some
mystical database based on the phone number. How to get that DB, I
don't know, but it sure wo
Folks,
I want to build a custom IVR for my setup. I've got it
working (well, the bells and whistles are not there
yet, but the basic stuff works) using AGI, but I'm
worried about how well this will scale.
I've seen references to FastAGI, and presumably this
will be more efficient.
Question, tho
--- Steven Critchfield <[EMAIL PROTECTED]> wrote:
> On Mon, 2005-03-07 at 13:30 -0800, beonice wrote:
> >
... snip ...
> > Would it help to split the db off to a separate
> server
> > (that should reduce the CPU load on the asterisk
> > server)?
> >
> > Any other alternatives? Anyone verifi
Maybe I’m loosing my mind but I’ve just noticed
that if I put a call on speakerphone and I press speakerphone again it hangs up
the call, you would expect it to take the call off speaker back on to the hand piece.
I’m using V 1.0.5.22 firmware.
Is there any other way to turn off spe
Guys.
Im trying to implement some kind of call forward or DND, I checked the wiki
and there are some examples of call forwards but I was wondering if anybody
has implemented one that will let you forward calls to SIP, IX or ZAP
channels alike? For example, forwardto another extension, to an outsid
Hello Ty,
> VGhhdCBpcyB0cnVlIC0gSSd2ZSBydW4gaW50byBpdCBvbiBzb21lIG9mIG15IHBvbHljb21zLiAg
> QWZ0ZXIgdHdlYWtpbmcgdGhlIHBob25lJ3MgYnVpbHQtaW4gZWNobyBjYW5jZWxsYXRpb24gSSB3
> YXMgDQphYmxlIHRvIGVsaW1pbmF0ZSBpdCB0aG91Z2guDQoNCg0KVHkNCi0tLS0tT3JpZ2luYWwg
> TWVzc2FnZS0tLS0tDQpGcm9tOiBEZW5uaXMgV2ViYiBbbWFpb
I am sure that asterisk is listening for SIP clients.
Did you configure your sip.conf correctly?
More info to look at...
site:lists.digium.com sip x-lite
If you are building this form scratch and cannot get the basics
compelted, I would just dump it and go to a build of [EMAIL PROTECTED] The
bu
>I've read through a good amount of documentation on voip-info.org, but
>hadn't found a solution, so I thought this list might help. I'm not
>great with linux, and I suspect there might be a port problem... maybe
>Asterisk isn't listening for SIP clients. How would I go about
>checking this? X-Lite
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