System execute "asterisk -rx reload"
?
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
> Sent: Saturday, 1 November 2003 5:18 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Asterisk: Reloaded
>
>
> Hello,
>
> Pretend I had a Perl
Hello,
Pretend I had a Perl script that did something to an Asterisk conf
file...
How can I [from Perl] ask Asterisk to reload?
;)
Ben
__
Benjamin Wakefield
[EMAIL PROTECTED]
http://www.dcsi.net.au/
DCSI - We do Internet.
64 Queen Street
Warragul, VIC 3820 AU
Ph: (+6
Does anyone have the Admin password for the phone
in order to change configuration
Roman
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi Nick,
You could put the 9 in the Dial.
exten => _XX,1,Dial(Zap/g1/9${EXTEN})
That would insert the 9 before any number you dialled.
;)
Ben
__
Benjamin Wakefield
[EMAIL PROTECTED]
http://www.dcsi.net.au/
DCSI - We do Internet.
64 Queen Street
Warragul, VIC
- Original Message -
From: "Phillip Jackson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, November 01, 2003 1:03 PM
Subject: [Asterisk-Users] VoicePulse Down?
> Anyone else having issues with voicepulse? Seems as if it may be down.
>
> Regards,
> Phillip
It works fine
Title: Huge silence breaks between Cisco 7960 phone & X-Lite
Does any one else have problems with huge, random silence breaks between an X-Lite and Cisco 7960 SIP phone? Both are running g.711. Softphone to/from softphone works, softphone to/from iax2 works, iax2 to/.from cisco phone works.
Anyone else having issues with voicepulse? Seems as if it may be down.
Regards,
Phillip
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Thanks Mark, we'll clear it.
> It changed IP addresses, might still have cached values. Should be:
>
>
> 69.73.19.178
>
>
> Mark
>
> On Fri, 31 Oct 2003, Rich Adamson wrote:
>
> >
> > I've not been able to register with iaxtel.com for the last couple
> > of days. Is
It changed IP addresses, might still have cached values. Should be:
69.73.19.178
Mark
On Fri, 31 Oct 2003, Rich Adamson wrote:
>
> I've not been able to register with iaxtel.com for the last couple
> of days. Is anyone else seeing this, or did I miss something?
>
>
>
> __
Hello,
The starting asterisk like that helped!
But when I start ./safe_asterisk I sill get this in log file.
So when I start manually from from terminal asterisk -vvvcng
everything works ok.
But when I start ./safe_asterisk i still get this in logfile
and G729 is not working.
Try starting a
480's have a headset port, 2.5 mm
Chad
> My problem with the Aastra phones is that NONE of them seem to have a
> headset jack.
>
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi!
> MGCP works on IP basis, it has no userid's or passwords.
Ouch - that means MGCP and NAT w/ dynamic IP (of the router) is a No-No?
Philipp
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Is
msn messenger capable of using asterisk as it's gateway?
Shoval
I've not been able to register with iaxtel.com for the last couple
of days. Is anyone else seeing this, or did I miss something?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
My problem with the Aastra phones is that NONE of them seem to have a
headset jack.
On Fri, 2003-10-31 at 14:23, Wade J. Weppler wrote:
> Digium will sell you PowerTouch 480's along with the security code for
> them, so you'll be able to use them with Asterisk.
>
> -wade
>
> > -Original Mess
Has
anyone tried using pcphoneline hardware from www.Pcphoneline.com ?
It
supposedly allowes you to connect asterisk to the pstn for a very low cost.
Shoval
If you already get dialtone, you just need
to setup the extensions
exten =>
321,1,Dial(MGCP/aaln/[EMAIL PROTECTED],90,T)
exten =>
322,1,Dial(MGCP/aaln/[EMAIL PROTECTED],90,T)
exten =>
323,1,Dial(MGCP/aaln/[EMAIL PROTECTED],90,T)
exten =>
324,1,Dial(MGCP/aaln/[EMAIL PROTEC
You're welcome.
Hope we can continue to
help each other.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos Arnt
Sent: Saturday, November 01, 2003
12:22 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] DTMF
x-lite
Yep,
that's right in RedHat 7.
Thanks a lot Ryan.
I actually lost almost two nights of sleep and
got these two answer myself.
I better have patience next time, eh?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan R. Fligg
Sent: Thursday, October 30, 2003
11:35 PM
To: [EMAIL PROTECTED]
Sub
Yep, that's right in RedHat 7.3 too !
Now works great!
Thanks alot.
On Fri, 31 Oct 2003 19:12:11 +0300, Shoval Tom wrote:> I fixed it. finally.>>> I was missing suidperl.>>> Apparently it isn't installed on redhat 9.0>>> You can download it from www.redhat.com and install it. And voile> everything
Hey all,
Has anyone ever had problems with flaky behaviour when registering
softphones dynamically with asterisk? I'm working with Kphone and
having problems getting consistent results with registration.
When Kphone is able to register, asterisk reports
-- registered SIP '2001' at XXX.XXX.XXX
Try setting the dtmfmode=rcf2833
and also make sure you have suidperl installed. These things really helped in getting my
X-Lite to work.
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer
Sent: Thursday, October 30, 2003
10:46 AM
To:
Digium will sell you PowerTouch 480's along with the security code for
them, so you'll be able to use them with Asterisk.
-wade
> -Original Message-
> From: Anton Tinchev [mailto:[EMAIL PROTECTED]
> Sent: Friday, October 31, 2003 1:12 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Us
Hi Lars,
I believe what you want to do is possible by applying a patch to the
Queue app, that being put together by some wizards.
You can find out about it here:
http://bugs.digium.com/bug_view_page.php?bug_id=214
I consider myself fairly new to Asterisk, but I'm going to assume that
patch w
Hi!
> I've set DTMFMODE to inband on both the sip.conf file and the x-lite
> configuration, and still it doesn't work.
>
> Anyone had this problem before>?
1. Why do you have to use inband DTMF? Stick with the default if you can.
2. From what I read on this list that'll only work with ulaw and
At 10/30/03 11:36 PM, "Dan" <[EMAIL PROTECTED]> wrote:
>Have you tried to use values like 0.5 or 0.8?
Hmmm, good suggestion, but it didn't help, unfortunately.
However -- I did some more testing, and found that using extremely large
negative values such as -20.0 does make it noticeably quieter
Try starting asterisk from /usr/src/asterisk
with the console
asterisk -vvvcng
regards
Martin
On Fri, 31 Oct 2003, Bartosz Jozwiak wrote:
>
> I just download a new one!
> And now I have that, it is even worser
>
> WARNING[16384]: File codec_g729b.c, Line 202 (lintog729_new): No available
>
Hi all,
We have a very basic * installation for testing purposes.
The * is connected to PSTN with BRI and setup with X-Lite
over plain lan. (local IP's)
OS: Linux/Debian unstable.
Asterisk CVS-10/29/03-23:46:26
chan_capi
On the IP side:
X-lite (build: 1084)
Calling and get calls on PSTN from X-
hello
you
can help me with a problem
I
have dlink DG-104S already and this registered in asterisk
but
not to call... between in ports
you
can help with an example the configuration me of
mgcp.conf
extensions.conf
Anton Tinchev wrote:
Is there any verified source for unlocked aastra phones?
Wade J. Weppler wrote:
All of the Aaastra PowerTouch phones are ADSI capable (350, 390, 392,
480). You just have to make sure they are UNLOCKED or you have the
security codes to be able to use the ADSI functions throug
I just download a new one!
And now I have that, it is even worser
WARNING[16384]: File codec_g729b.c, Line 202 (lintog729_new): No available
g729 resources for channel 0
WARNING[16384]: File codec_g729b.c, Line 342 (lintog729_framein): G729
resources are not allocated, exiting
Error Opening c
You are not using the new codec binary,
On Fri, 2003-10-31 at 12:59, Bartosz Jozwiak wrote:
> Hello,
>
>
>
> I have that problem with codec G729.
>
> Please can somebody help me!
>
>
>
> WARNING[16384]: File codec_g729b.c, Line 413 (load_module): Unable to
> initialize va stuff: -1
> =
Hello,
I have that problem with codec G729.
Please can somebody help me!
WARNING[16384]: File codec_g729b.c, Line 413
(load_module): Unable to initialize va stuff: -1 == Detected 4
licensed G.729 transcodersWARNING[16384]: File translate.c, Line 219
(calc_cost): Translator 'g729tolinb' d
I have the same problem and it was solved setting:
# Uncomment for aggressive residual echo supression under
# MARK2 echo canceller
#
KFLAGS+=-DAGGRESSIVE_SUPPRESSOR
in the makefile of zaptel and recompiling.
miklos
- Original Message -
From: "Ernest W. Lessenger" <[EMAIL PROTECTED]>
I'm testing out my SNOM 200 phone by trying to call out through NuFone.
When I do so, I don't hear an echo at all (in fact I can't hear myself
through the phone) but the callee can hear an echo when she speaks. NuFone
tells me their network is totally digital and so can't be involved in an
echo
Hi John,
- Original Message -
From: "John Todd" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, October 31, 2003 7:44 PM
Subject: Re: [Asterisk-Users] Some problems after an Asterisk update
> ...
> No, I don't see that problem.
>
> Try "make clean" and then "make install" in
> I want to have a list of companies providing services via IAX on my
> Asterisk web page. If you know of a company that does this or run a
> company that does this please e-mail me at [EMAIL PROTECTED] with 1) web
> site, 2) contact info and 3) services provided. I don't want to put
> pricing in
--- William Waites <[EMAIL PROTECTED]> wrote:
> Both of these have an "if" statment that checks to see if
> > the public address needs to be stuffed into the outbound
> > SIP packet. I would replace this "if" with one that checks
> > the result of a STUN query. STUN simply makes Asterisk
> > mo
rnc Info Lists wrote:
Hi,
-Original Message-
The portion of extensions.conf is:
exten => 3001,1,Dial(MGCP/aaln1,20)
exten => 3001,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20)
Or aaln/1@ should do just fine. However this doesn't explain why there
is no dialtone on the
change it to
[192.168.0.5]
host=192.168.0.5
context=blabla
Martin
> notice that when I booted up everythign tonight that the MGCP SHOW
> ENDPOINTS now shows:
> Gateway 'ip10' at 0.0.0.0 (Dynamic)
>-- 'aaln/[EMAIL PROTECTED] in 'from-sip' is idle
>
> In the messages at start up there is:
> ==
Hi,
Yesterday evening I have done a full update of Asterisk on a test system.
The version is CVS-08/25/03-15:55:51
After this operation I get some big problems:
- the Voicemail2 application does not work anymore. I must disable it in
modules.conf file in order to be able to start * without crashin
Hi
I work at the Rand Afrikaans University in South Africa and we have +- 2200
staff. We are currently able to manage our Philips based PBX using Telephony
Management Software called TABS. We are looking at integrating * into our
current operations to bridge campus' and run a call centre. Since th
Hello all,
Is there a way to enter a dial string that contains the
';' character? Currently, Asterisk thinks that the part
following the ';' character is a comment and, correctly,
ignores it.
Michael.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Fri, 31 Oct 2003 09:09:22 -0800 (PST), Chris Albertson wrote
>
> Stephens, I think preferably, introduces one new sip.conf
> line for the internal _network_ which acceprts a "network
> address in the form inside=192.168.111.0/14 Where the "14"
> would be the number of zero bits in a 32-bit mask
I think the goal here is not to save the cost of
1GB of disk space but to par down the system to the point
where it can run with ZERO disk drive and save the space,
power and heat and the risk of it failing.
You can set up a firewall using a cheap PC and Linux for
a few hundred bucks OR.. you cou
On Fri, 2003-10-31 at 10:24, David Gomillion wrote:
> I can understand the size concerns for putting it in an appliance or
> what-not. However, my opinion is that, due to the low cost of hard disk
> space, it is cheaper for the company to go out and buy another hard disk
> to replace the extra 500
Hi,
Yesterday evening I have done a full update of Asterisk on a test system.
The version is CVS-08/25/03-15:55:51
After this operation I get some big problems:
- the Voicemail2 application does not work anymore. I must disable it in
modules.conf file in order to be able to start * without crashin
Is there any verified source for unlocked aastra phones?
Wade J. Weppler wrote:
> All of the Aaastra PowerTouch phones are ADSI capable (350, 390, 392,
> 480). You just have to make sure they are UNLOCKED or you have the
> security codes to be able to use the ADSI functions through Asterisk.
>
>
I fixed it. finally.
I was missing suidperl.
Apparently it isn't
installed on redhat 9.0
You can download it from www.redhat.com and install it. And voile –
everything works.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos Arnt
Sent: Friday, October 31,
All of the Aaastra PowerTouch phones are ADSI capable (350, 390, 392,
480). You just have to make sure they are UNLOCKED or you have the
security codes to be able to use the ADSI functions through Asterisk.
There were some long discussions on the list a while back on this very
issue. Best to sea
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> David Gomillion
> Sent: Friday, October 31, 2003 11:25 AM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Absolute Minimum Installation Packages
>
>
[...]
>
> What are the benefits to a re
> Hi,
>
>> -Original Message-
>> >The portion of extensions.conf is:
>> >exten => 3001,1,Dial(MGCP/aaln1,20)
>>
>> exten => 3001,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20)
>
> Or aaln/1@ should do just fine. However this doesn't explain why there
> is no dialtone on the phone..
>
> Oh, one thou
At 10:24 AM 10/31/2003 -0600, you wrote:
I can understand the size concerns for putting it in an appliance or
what-not. However, my opinion is that, due to the low cost of hard disk
space, it is cheaper for the company to go out and buy another hard disk
to replace the extra 500 MB they wasted on
I just need to buy 5-6 ADSI Phones.
I wondering which of these models to choose
- aastra 390 - I don't know is this an ADSI phone at all. Is there versions with and
without ADSI
- aastra 350 - I'm sure that it have ADSI.
If there is some other good working model, it will be great, if someone p
It has been extremely slow for me too.
Regards,
Mike
On Fri, 31 Oct 2003 15:46:49 +0100, Roy Sigurd Karlsbakk <[EMAIL PROTECTED]>
wrote :
> I just went there.
> Do they share a single isdn B channel with 50 other servers?
> it was sloow.
> I'll put it there, eventually
>
> On Fri, 2003-10-3
http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ
Welcome to add short questions and answers!
/O
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
I can understand the size concerns for putting it in an appliance or
what-not. However, my opinion is that, due to the low cost of hard disk
space, it is cheaper for the company to go out and buy another hard disk
to replace the extra 500 MB they wasted on a sub-optimal installation
than to pay me
On Thu, 30 Oct 2003 16:18:23 -0800 (PST), Chris Albertson wrote
>
> This would be VERY much like the two current patches do except
> that we would no longer need the new lines in sip.conf as STUN
> would figure this out for us.
>
you would still need the lines to specify the internal network/mask
what exactly do you want to make with phones ? change image ?
Unofficial Asterisk Forums
URL : http://asterisk.xvoip.com
Registration is : http://asterisk.xvoip.com/profile.php?mode=register
***
Hi David,
> Sorry this is a little of topic but, I have just received our 10 cisco
> 7940 phone. I want to change them to sip but it seems that I need a
> support contract with cisco. I'm in the UK does anybody know were to go
> to get one or how much it will cost :-(
We had exactly this probl
To try voicemail main you need to define an extension for it.
You can make samples for asterisk and look in the extensions.conf file for a
sample of defining a voicemail extension.
You can also use a web interface to check your messages - there you also use
the password to check your voicemail - d
I'm new to Asterisk, but, Managed to get it working for outound calls from
my ATA --> Asterisk --> Cisco 2620 using SIP. However, I'm having problems
with Inbound calls from the Cisco.. Cisco 2620 --> Asterisk --> ATA .. In
fact, voice mail won't even work..
This is a snippet of what I'm getting
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> David Stubbs
> Sent: Friday, October 31, 2003 10:48 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Cisco Support Contacts
>
> Sorry this is a little of topic but, I have just received our
>
Hi all,
Sorry this is a little of topic but, I have just received our 10 cisco
7940 phone. I want to change them to sip but it seems that I need a
support contract with cisco. I'm in the UK does anybody know were to go
to get one or how much it will cost :-(
Thanks,
David Stubbs
Idessa UK Ltd
> Agreed. There is no need to create YAAWS (Yet Another Asterisk Website.
> Before you post answers at least please do a check to verify their
> accuracy. There has been alot of questions about doing SIP from behind
> NAT. Even this week it was discssed on the list rather extensively that
> it CA
I found it nice and fast...
On 31 Oct 2003, at 14:46, Roy Sigurd Karlsbakk wrote:
I just went there.
Do they share a single isdn B channel with 50 other servers?
it was sloow.
I'll put it there, eventually
On Fri, 2003-10-31 at 15:21, Rich Adamson wrote:
Roy,
I've started to write an FAQ fo
I want to have a list of companies providing services via IAX on my
Asterisk web page. If you know of a company that does this or run a
company that does this please e-mail me at [EMAIL PROTECTED] with 1) web
site, 2) contact info and 3) services provided. I don't want to put
pricing info on the
Hi.
Il ven, 2003-10-31 alle 02:31, JR Richardson ha scritto:
> I’m trying to get the total Linux/* installation size as small as
> possible. I’m wondering if anyone has looked at the installed
> packages list from the Redhat installation [rpm –qa] and has parsed
> out all packages not needed for
I just went there.
Do they share a single isdn B channel with 50 other servers?
it was sloow.
I'll put it there, eventually
On Fri, 2003-10-31 at 15:21, Rich Adamson wrote:
> Roy,
>
> > I've started to write an FAQ for asterisk, available here:
> > http://asterisk.pronto.tv/faq.php
> >
> >
Apologies for breaking the thread - the link to my mail server is down,
so am replying from the Pipermail web archives.
I'm using a cut-down Debian-based distro called 'pebble' for my firewall
machine using a 128MB CompactFlash card (it would just about fit on
64...)
The various system files have
>>> I've started to write an FAQ for asterisk, available here:
>> http://asterisk.pronto.tv/faq.php
>>
>> Please help me fill it up with the good stuff :)
>
> Why don't you put it here:
> http://www.voip-info.org/tiki-index.php
> and folks can updated/edit online?
>
>
>
Agreed. There is no need
> Hello All,
>
>
>
> I have pingtel and asterisk working really well. I have a really
> annoying little problem - mainly with pingtel. When a call comes in
> pingtel displays the caller ID on the phone. If I miss it then I click
> on the number for redial - this doesn't include a 9 to dial an outs
Hello ,
We are in second stage for interoperability testing
between Asterisk and our xvoip network.
Now we are going to conduct performance, quality
and qos tests.
We need 25 testers with different hardware to
participate in this test.
Testing schedule is next :
1: Registration
2: Inte
Hi All,
I connect SIP phones and H323 phones,
with *.
But, if I call from H323 to SIP phone, i can
hear nothing. I'm using h323 library.
I found some e-mails at the Digium's
Mail-list with the same problem. But, I coudn´t find the solution
yet.
Could you please help me?, any
> > Oh, one thought: Did you set your toneconfiguration to Europe or US ? If
> you
> choose custom you need to configure it another way...
>
Florian
The tone config on the phone is set to Europe. Asterisk is USA.. Hmm..
Will change the phone to USA when I get home and see if that makes a
differen
Roy,
> I've started to write an FAQ for asterisk, available here:
> http://asterisk.pronto.tv/faq.php
>
> Please help me fill it up with the good stuff :)
Why don't you put it here:
http://www.voip-info.org/tiki-index.php
and folks can updated/edit online?
__
Title: RE: [Asterisk-Users] Absolute Minimum Installation Packages
There are good instructions for building a minimal RedHat 7.2 (http://www.muine.org/~hoang/minired.html) or 7.3 (http://www.muine.org/~hoang/minired73.html) system.
I have not followed these instructions, but I have followed
Hello All,
I have pingtel and asterisk working really well. I have a really
annoying little problem - mainly with pingtel. When a call comes in
pingtel displays the caller ID on the phone. If I miss it then I click
on the number for redial - this doesn't include a 9 to dial an outside
line. The
Hi
I work at the Rand Afrikaans University in South Africa and we have +- 2200
staff. We are currently able to manage our Philips based PBX using Telephony
Management Software called TABS. We are looking at integrating * into our
current operations to bridge campus' and run a call centre. Since t
> I launched Asterisk with ./asterisk -vvvc and I got *CLI> promt.
> I run the "dial" and got some messages.
What messages???
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
hi all
I've started to write an FAQ for asterisk, available here:
http://asterisk.pronto.tv/faq.php
Please help me fill it up with the good stuff :)
RoyK
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/aster
Hi,
> -Original Message-
> >The portion of extensions.conf is:
> >exten => 3001,1,Dial(MGCP/aaln1,20)
>
> exten => 3001,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20)
Or aaln/1@ should do just fine. However this doesn't explain why there
is no dialtone on the phone..
Oh, one thought: Did you s
Hi,
> -Original Message-
> >The following comes on the Asterisk console at powerup. The
> items between
> >the repeat.
> >MGCP Show endpoints doesn't show anything. Evidently the phone isn't
> >registered but not sure why since there doesn't seem to be a place to
> >associate a us
Hi All,
I am quite new to Asterisk.I have one PIII,512 MB RAM,One full duplex sound
card,One Mediatrix 1204 switch (It converts sip packet to PSTN signal
vis versa),One CISCO ATA-186.I already installed the asterisks software
in my LINUX 7.3 box using thses command in konsole
make,make install,make
rnc Info Lists wrote:
I have a SwissVoice IP10S but can not seem to get it to have dialtone or
dial on *. Calls to or from 3001 don't work.
Any ideas are appreciated.
Robert
mgcp.conf is:
[general]
port = 2427
bindaddr = 192.168.0.110
[ip10]
host = 192.168.0.5
context = from-sip
line => aaln/1
Th
rnc Info Lists wrote:
Citeren rnc Info Lists <[EMAIL PROTECTED]>:
I have a SwissVoice IP10S but can not seem to get it to have dialtone or
dial on *. Calls to or from 3001 don't work.
Were you able to configure the phones through their webinterface ?
You could try entering 'mgcp deb
> Using this technique I've got the bare essentials from a Mandrake Cooker
> installation onto an old 340Mb disk. It's now integrated into the
> network and scp, ssh are working. Today's exercise is to set up a web
> configuration server. Total space used so far is 31466 1K blocks. This
> could pos
Is there an option to configure Voicemail2 to NOT store the voicemail
messages on the disk once they have been emailed to one's mail server.
It can be a pain for some to receive voicemail via email and then go to
Asterisk just to clean out the voicemail you just heard.
_
Hi all!
Every time i receive a sip call MOH begin to
play and i can´t talk to the caller.
My setup is the default.
Someone knows what is the problem?
thanks
Miklos
iPFONE Telefonia IPRua
Caio Graco 735 São Paulo SP iPBX +55 11 3801-3702FWD 64662ICH
31451543www.ipfone.com.br[EMAIL
multiple servers, the first problem would be sharing the SIP
>registration information between the two or more servers..
>This is so that if UA1 is registered on Server1 and UA2 is registered
on
>Server2.. Then when a call is made from UA1 to UA2, Server1 would know
>the registration details of
On Fri, 2003-10-31 at 02:39, Adam Hart wrote:
> why start this with redhat? I'd say it's the worse linux dist to
> attempt to make a small footprint. Try gentoo.
If you really want to get to a minimum why not Linux From Scratch?
Or try this, be prepared for a bit of a long typing session.
Creat
Hi
I know this is a bit off topic, but still pretty interesting.
I'm running Asterisk on my Linux router/NAT/FW connected via
cable (1mbit/200kbit) to the internet.
Now, I wanna do local QoS implementation. Just very simple to
give RTP (UDP) highest priority on my outbound interface. So,
wheneve
I am currently using Asterisk with G.711 codecs and in-band DTMF for
several Cisco 7960's
and an Audiocodes GW. When allowing out-of-band DTMF, I could use
voicemail menus and
anything else on Asterisk that required DTMF but I could not get the
DTMF relayed out of the
GW. Has anyone verified that
if you will use voicemailmain2 or voicemailmain() application then it will
ask you for mailbox and corresponding password.Voicemail application only
ask for mailbox.
No you don't need to do anything with sendmail for voicemail.
Just confirm if the sendmail is working fine.
-Manoj k Gupta
- Or
Béasse Christophe wrote:
I have some troubles with voicemail sending message to my email address.
Do I have to configure sendmail to use Voicemail ?
I found that the use of sendmail was hardcoded into voicemail. Therefore,
I've submitted a patch to voicemail2 that let's you configure mail comma
I configure sendmail, and now al is running OK - Fine
Thanks a lot for your help.
Best regards
Christophe
- Original Message -
From: "Olle E. Johansson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, October 31, 2003 11:06 AM
Subject: Re: [Asterisk-Users] Password in VoiceMail
> Few more
things, > the SIP users are
connected to the Asterisk through Local LAN, on G711. > We also
discovered that, once an outside caller put onhold, the 'music on
hold' > they hear, is, also
intermittent. > > pls help
us. > > Herc
I experience a somewhat similar
Chee Foong wrote:
Hello,
I have te following setup:
IAX client -(iax)-> Asterisk -(h323)> Cisco AS5300
At the present moment GSM codec is used betwee IAX client and Asterisk. G729
is used between Asterisk and Cisco AS5300.
I am thinking that switching from GSM to G729 between IAX
Peter Zeltins wrote:
Checking e-mail this morning it looks like we have two independent
"fixes" that both do what has been suggested in this thread.
No need for a third except posibly a merge of the two.
Would you care to elaborate? I don't see anything in asterisk-users, and no
mention of SIP-b
Béasse Christophe wrote:
Hi,
In voicemail. i declare :
1000 => 1234, ,[EMAIL PROTECTED]
I don't see what's password 1234 is for ?
password for what ?
Where this password is used ?
Where this password is defined ?
This is a pin code used by the VoiceMailMain application to let user
1 - 100 of 115 matches
Mail list logo
