Wes Santee wrote:
Greetings all,
I'm on 1.2.9.1, and I'm trying to get a dialplan working that uses text
labels, but it's not working. For instance, take the following macro
snippet:
[macro-dosomething]
exten => s,1,GotoIf($["${MACRO_EXTEN:1:1}" != "1"] ? scid)
exten => s,n,Set(MACRO_EXTEN=1${
Doug Lytle wrote:
Dan Elder wrote:
Hey All, probably missing something really obvious here, but when our
users
are trying to dial the phone, asterisk timesout really quickly if they
don't
press the digits fast enough. Is there a global timeout value for dialing
See:
http://www.voip-info
Hello Iam trying to communicate between two asterisk servers using IAX protocol. Details are SERVER 1 IP address-192.168.0.54 Clients 949 and 950 SERVER 2 IP address-192.168.0.11 Clients 449 and 450 When i tried to dial 949 from server
Interesting points on both messages
1) as far as multiple asterisk servers talking to the same database is
concerned, I will have to test this out. I know nothing about the
database side of things, and a newbie on asterisk and linux so I have
no idea what and where the development of either of th
>
> So I've just had the time to swap and disable usb in my bios and it
> changed nothing the quality is still the same (which means horrible).
> How could I check where the problem comes from?
>
> Ben
Hmmm... that's a shame. Apologies if you have already specified this,
but what are the version
Douglas Garstang wrote:
Yes, we tried to do the same thing. We wanted our Asterisk system to be multi-homed.
My head office Asterisk box is multi-homed: I have three networks across
two NICs. One dedicated to hardphones, another to the local LAN (and
PC-based softphones). The third network is
On Tue, 2006-07-11 at 16:31 +1200, kjcsb wrote:
> I have an Asterisk server with 2 network cards. One provides the LAN
> connection and the other provides the Internet connection. Currently this is
> set up in the following way:
>
> eth0 192.168.1.5. This provides LAN connectivity
>
> eth1 192.
Yes, we tried to do the same thing. We wanted our Asterisk system to be
multi-homed. Turned out to be a dissapointing limitation of Asterisk. It would
have been nice to have, because then you could have multiple NIC's, have
Asterisk listen on both, and if one failed, you had some degree of redun
DUNDi doesn't provide good redundancy for phone registrations. Each phone is
only registered on a single, primary Asterisk system. In the event that the
primary system for a given phone becomes available, the phone will not
re-register until it's registraiton expirey period, on it's secondary As
Asterisk realtime hardly provides redundancy.
1. There's no support for realtime SIP where multiple Asterisk systems can
reference the same MySQL database for SIP peers. Ask Kevin Fleming about this.
It's known not to work.
2. The IP address of the MySQL server is hard coded into the Asterisk c
The problem is in the space. You've got it as " ? scid)"... In order
for the label to work, you need to get rid of the space. Make it
"?scid)" and it should work fine.
The error's in the details:
pbx_extension_helper: No such label >>' scid'<< in extension 's' in
context 'macro-dosomething'
On
On Mon, Jul 10, 2006 at 08:54:37PM -0700, Wes Santee wrote:
> Greetings all,
>
> I'm on 1.2.9.1, and I'm trying to get a dialplan working that uses text
> labels, but it's not working. For instance, take the following macro
> snippet:
>
> [macro-dosomething]
> exten => s,1,GotoIf($["${MACRO_EXTE
Another note:
On Mon, Jul 10, 2006 at 10:05:22PM +0900, Ganbaa wrote:
> Hi,
>
> I have configured digium tdm04b card with asterisk on debian. Incoming
> call is ok. But outgoing call has problem. Would you give me advice ?
It would have helped to get a CLI trace of of such a problematic call,
kjcsb wrote:
> I have an Asterisk server with 2 network cards. One provides the LAN
> connection and the other provides the Internet connection. Currently
> this is set up in the following way:
>
> eth0 192.168.1.5. This provides LAN connectivity
>
> eth1 192.168.1.251, gw 192.168.1.252 (Note that
The last log line suggests I can't use labels, but according to
http://www.voip-info.org/wiki/index.php?page=Asterisk+priorities it
shouldn't be a problem.
Labels work fine (and have been for a while). The snippet you provided
looks correct to me too. Are there are warning/errors when loading
ex
On Mon, Jul 10, 2006 at 04:20:53PM +, Julian Varanini wrote:
> Hi,
>
> Has anyone used the cooker RPM for asterisk version 1.2.9?
> I would like to hear some feedback before I install it.
Why don't you just try to install it?
Alternatively grab its srpm. If it is well-done it should hav
I have an Asterisk server with 2 network cards. One provides the LAN
connection and the other provides the Internet connection. Currently this is
set up in the following way:
eth0 192.168.1.5. This provides LAN connectivity
eth1 192.168.1.251, gw 192.168.1.252 (Note that other nodes on the net
looking on the code, I think that the time on the table is the time that
needed the CPU in order to translate 1 second of media.
So, on the case of calls with ulaw <-> alaw translation (1 ms in each
translation), this CPU could sustain, theoretically, 500 calls without delay
(two "simultaneous" tr
Download patch for ncs support in mgcp from http://asterisk.urtho.net/tiki-index.php .
2006/7/7, Christian Schnelle <[EMAIL PROTECTED]>:
Hi,i try to use asterisk together with a webstar mgcp cable modem. if iconfigure the cable modem to act as a nuera
v5.2 call agent it connectsto asterisk. but i
unplug, thanks for pointing that out as well as opposed to or in
complement with ARA where you can either implement DUNDi between
clusters of Asterisk servers or have a redundant pair of DUNDi lookup
servers (just like DNS) somewhere remote to the local asterisk
servers. DUNDi is a p2p IAX based p
Greetings all,
I'm on 1.2.9.1, and I'm trying to get a dialplan working that uses text
labels, but it's not working. For instance, take the following macro
snippet:
[macro-dosomething]
exten => s,1,GotoIf($["${MACRO_EXTEN:1:1}" != "1"] ? scid)
exten => s,n,Set(MACRO_EXTEN=1${MACRO_EXTEN})
exten
I found a website talking about DUNDI witch asterisk clustering. What
is DUNDI actually? Does it can implement in asterisk
clustering/loadbalancing?
On 7/11/06, RR <[EMAIL PROTECTED]> wrote:
Alejandro,
doesn't sound like you've read up or done research on ARA (Asterisk
Realtime Architecture)
Alejandro,
doesn't sound like you've read up or done research on ARA (Asterisk
Realtime Architecture)? That's what allows you to build asterisk
server clusters which draw upon configs either for individual config
files OR entire family of processes froma common database (which can
then be made r
On Tue, 2006-07-11 at 12:29 +1000, MBIT Technologies wrote:
> Hi Guys
> I am just looking for a bit of help here. I am trying to integrate the
> 2 of these together via a E1 link. The link has no signalling and is
> basically a dumb 2 meg link.
I would have thought that you would have _some_ typ
At 06:40 PM 7/10/2006, you wrote:
exten => s,17,GotoIf($[${DIALSTATUS} = NOANSWER],106)
exten => s,17,GotoIf($[${DIALSTATUS} = NOANSWER]?106)
Shouldn't the comma be a question mark?
Ira
___
--Bandwidth and Colocation provided by Easynews.com --
a
I have done this once before (here in Melbourne) so I know it can be
done.
This was done when I was a contractor, so I am not privy to the
information. If you really need to know, I can put you in touch with the
person who hired me to do the work.
>From memory, the biggest hassle was the NEAX. A
Hi Guys
I am just looking for a bit of help here. I am trying to integrate
the 2 of these together via a E1 link. The link has no signalling and is
basically a dumb 2 meg link.
Does anyone have any zaptel configurations they could give me.
I am using a Digium TE110P.
__
You need a ? after the condition, not a ,.
eg:
exten => s,17,GotoIf($[${DIALSTATUS} = NOANSWER]?106)
later,
Paul Hales
--
Paul Hales
Technical Manager
AsteriskIT
www.asteriskit.com.au
ph: 03 8320 8100
On Mon, 2006-07-10 at 18:40 -0700, Eric Lyons wrote:
> Hmm.
>
> GotoIf seems to be doing
Hmm.
GotoIf seems to be doing something, uh, wrong in my dialplan: Despite getting a TRUE value in the conditional (as shown in the
debug below), it doesn't take the jump! The following is part of the macro being executed in the dialplan, starting at priority 16:
exten => s,16,Dial(IAX2/six
On 7/10/06, trixter aka Bret McDanel <[EMAIL PROTECTED]> wrote:
zphone is phil zimmermans (creator of pgp) encrypted rtp system. Unlike
SRTP this does not rely on the server itself to provide the encryption.
It also lets you be reasonably assured that if the numbers displayed
match then not only
Hi all,
I have read a lot about * server redundancy, however I still don't know
how to do it. Can any of you give me any advise?
For example, I've read about ranchnetworks appliances but don't know if it
will solve my problems.
As you may guess, I need to have two servers with the same inform
That is at the server iself - you could then argue that the transit RTP
could be tapped by a corrupt tech working for your ISP or provider, which
could happen also with physical lines, the difference being that the RTP
tap is so virtual it can be made to leave no trace. A physical tap can be
fo
Is NVFaxDetect for PSTN calls or works for SIP/IAX as well?
On 7/9/06, Thomas Kenyon <[EMAIL PROTECTED]> wrote:
Olivier wrote:> 2006/7/6, Maxim Vexler <[EMAIL PROTECTED]
[EMAIL PROTECTED]>>:>> NVFaxDetect does just that ;)>> Do you think NVFaxDetect is reliable ?> Could you use it along a voic
My service provider had issue with his Cisco hardware when it came to MoH. They were new with Asterisk at that time. I told them many times that they had problem in their system, but they never agreed, until one day when one of their engineers figured out that the Cisco hardware was compressing the
Somebody mentioned about FoIP (Fax over IP) and faxserver.com. Has somebody used it?
What about NVFaxDetect? Is that for VoIP or needs PSTN?
Zeeshan A Zakaria
On 7/10/06, Lee Howard <[EMAIL PROTECTED]> wrote:
Zeeshan Zakaria wrote:> I am trying to setup fax on my phone system. Which fax-to-ema
Mike Puchol wrote:
I would have to strongly disagree - if Asterisk was toted as a kid's
toy, and sold by Fisher Price, then maybe security has no importance.
But, if Asterisk or any other VoIP platform, for that matter, is to be
introduced into the enterprise, it *has* to provide security. Tap
On Tue, Jul 11, 2006 at 06:53:14AM +1000, Eric Bishop wrote:
>What us meant by "blended rate"?
It generally means a rate that the provider sets that is fixed even if
the provider is charged different rates
i.e. BT International provide blended fixed and mobile termination
rates, even though
Julian Varanini wrote:
Here comes a big newbie questions if compiled how would I uninstall it?
By deleting the directories that the install creates. I think that
there are 5 or so.
Doug
-- Ben Franklin quote: "Those who would give up Essential Liberty to
purchase a little Temporary Safety
I have been playing with ztdummy on both Linux and BSD on a variety
of hardware without good results from zttest (99.2-99.5% where one of
the FAQs says you're looking for 99.975 or better).
Would slapping a Digium card (like the TDM400) in the box help or
would the problems that keep ztdummy f
Awesome thanks that is exactlly what I was looking for. Thanks!
On 7/10/06, voiplist <[EMAIL PROTECTED]> wrote:
Yes, you want to use different context for each house.
In your sip.conf:
[house1]
username=house1
secret=house1pass
context=house1
--->Other sip options here<---
[house2]
username=h
Yes, you want to use different context for each house.
In your sip.conf:
[house1]
username=house1
secret=house1pass
context=house1
--->Other sip options here<---
[house2]
username=house2
secret=house2pass
context=house2
--->Other sip options here<---
In your extensions.conf:
[house1]
;House1
Here comes a big newbie questions if compiled how would I uninstall it?
Thanks
Julian
> Date: Mon, 10 Jul 2006 16:53:37 -0400> From: [EMAIL PROTECTED]> To: asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Mandriva 2006 Cooker RPM for Asterisk 1.2.9> > Julian Varanini wrote:>
Giedrius Augys wrote:
Hi,
My situation is : I need to send fax from sip device attached fax
over zap channel. Using G711, fax send ok, but is it posible to use
t.38 protocol. Maybe someone can suggest me what software to use?
You want to go to the spandsp site and read all the challenges.
Julian Varanini wrote:
I have not done a lot of compiling in Mandriva. Did you create a
directory for it, e.g. /data/asterisk?
I extract it to a temp directory, during the compile/install, the
directories are created automatically.
Doug
--
Ben Franklin quote:
"Those who would give up
What us meant by "blended rate"?
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Dan Elder wrote:
Hey All, probably missing something really obvious here, but when our users
are trying to dial the phone, asterisk timesout really quickly if they don't
press the digits fast enough. Is there a global timeout value for dialing
See:
http://www.voip-info.org/wiki/index.php?p
Hey All, probably missing something really obvious here, but when our users
are trying to dial the phone, asterisk timesout really quickly if they don't
press the digits fast enough. Is there a global timeout value for dialing
that I could increase to make dialing easier for our users? Using * 1.2.
I have not done a lot of compiling in Mandriva. Did you create a directory for it, e.g. /data/asterisk?
Thanks
Julian
> Date: Mon, 10 Jul 2006 15:23:22 -0400> From: [EMAIL PROTECTED]> To: asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Mandriva 2006 Cooker RPM for Asterisk
On Jul 10, 2006, at 10:48 AM, Andrew Niemantsverdriet wrote:
Is that the standard way of doing things? I found a bunch of asterisk
hosting providers in my search on the best way to do this. Is this
what they are doing?
Yes,l I think that's what contexts are for... I am also relatively new
at
Julian Varanini wrote:
Hi Doug,
I thought of that as well. I am not a total n00b with Mandriva, just
enough to be dangerous. Do you have any walk throughs that I could
use as a guide? Do you create an asterisk user for it to run under?
What other software should I install in order for it
I use stable 1.2.9.1 for my servers. How can I maintain my asterisk
1.2.9.1 updated with the patches produced for that release, in case a
patch fits a need?
what should I do in MANTIS to see patches applied to 1.2.9.1?
While looking at MANTIS I just (?) saw one entry for Product build
1.2.9.1 howe
But my question is, those that mean that it will take 1 second to
convert 50 channels? if so do I get a 1 second latency when coverting
50 channels?
On 7/10/06, Fabio <[EMAIL PROTECTED]> wrote:
I think it's the same,
10 calls in 200ms = 50 calls in 1s
because 1s = 5 x 200ms
IMHO, is bet
Yes that is correct.
Bill
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph
Sent: Monday, July 10, 2006 12:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Choppy MOH (Cisco gateway)
On Jul 10,
Is that the standard way of doing things? I found a bunch of asterisk
hosting providers in my search on the best way to do this. Is this
what they are doing?
On 7/10/06, Tom Lynn <[EMAIL PROTECTED]> wrote:
You can place the phones at each house in a different context. Trunks, too.
On 7/10/06
Ariel Batista wrote:
Justin Johnson wrote:
Hi All,
I have centOS 4.3 installed and have attempted to install asterisk
separately. I have installed all the modules as suggested on Asterisk
downloads, more (via SVN) However, on the zaptel install I am getting
the following errors.
centosbug is
You can place the phones at each house in a different context. Trunks, too.
On 7/10/06, Andrew Niemantsverdriet <[EMAIL PROTECTED]> wrote:
I have a asterisk box up and running great. I have another house in mybackyard that also wants to use my asterisk box. I am running trixbox
now and have tw
I have a asterisk box up and running great. I have another house in my
backyard that also wants to use my asterisk box. I am running trixbox
now and have two POTS lines connected to digium TDM400P as well as 1
voip line for long distance. I would like to keep these two houses as
seperate as possib
On Jul 10, 2006, at 4:49 AM, Bill Gibbs wrote:
And of course I just found this article
http://www.cisco.com/warp/public/788/voice-qos/hissing.html#topic3
Hope this helps some other people out as well!
So was the fix to reconfigure your gateway to not use VAD?
Just want to be clear...
Marty
Hi Doug,
I thought of that as well. I am not a total n00b with Mandriva, just enough to be dangerous. Do you have any walk throughs that I could use as a guide? Do you create an asterisk user for it to run under? What other software should I install in order for it to not only compile prope
shadowym wrote:
Yes, I am using the 1.2.7.1 patch on 1.2.9.1. It seemed to work fine.
Still curious if anyone has this working on an Aastra phone? I can't get it
to work but someone in the bug.digium.com list said they had it working on
an Aastra phone. Maybe I am missing something. I tried
Julian Varanini wrote:
Hi,
Has anyone used the cooker RPM for asterisk version 1.2.9? I would
like to hear some feedback before I install it.
I haven't, I find it just to easy to compile it under Mandriva.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to pur
Kai Fürstenberg napisał(a):
> If you want to connect a telephone to the HFC card you need a crossed
> cable to connect to an NTBA to which you connect the phones.
i'm connecting to ISDN-NTBA from T-Com.
> As far as I know you don't need a crossed cable when you connect the
> phone directly.
>
>
There is an old, very old document that I found somewhere that this
PoE switch was designed for NBX phones at that time.
Does anybody in this list is using this switch with non-3com NBX PoE phones?
--
Erick Perez
Panama Sistemas
Integr
Hi,
Has anyone used the cooker RPM for asterisk version 1.2.9? I would like to hear some feedback before I install it.
Thanks
Julian
> From: [EMAIL PROTECTED]> To: asterisk-users@lists.digium.com> Subject: [asterisk-users] Re: Metermaid phone compatibility> Date: Mon, 10 Jul 2006 09:0
Hi Everyone,
I was wondering if anyone had any ideas regarding this. I can see in the sip debug that music on hold is called when a person is put on hold, however I hear nothing. Any help would be appreciated.
Thanks
Julian
From: [EMAIL PROTECTED]To: [EMAIL PROTECTED]; asterisk-users@lis
Yes, I am using the 1.2.7.1 patch on 1.2.9.1. It seemed to work fine.
Still curious if anyone has this working on an Aastra phone? I can't get it
to work but someone in the bug.digium.com list said they had it working on
an Aastra phone. Maybe I am missing something. I tried just about
everyt
Zeeshan Zakaria wrote:
I am trying to setup fax on my phone system. Which fax-to-email and
email-to-fax solution really works on IAX and SIP?
Due to the way that your question has been asked I assume that you're
talking about IAX or SIP over a latent internet connection to some VoIP
provide
I'm trying to provide dial tone on E&M Wink type trunks. I found where
in source, 'chan_zap.c' where I believe the code needs to be added.
Basically I believe I can copy parts used for PRI in to E&M and E&M Wink
signal types.
However with my attempts, it fails to compile at chan_zap. And I'm n
Title: Message
I want to allow a
SIP caller to place multuiple consecutive calls -
So a caller connects
to Asterisk and gets routed to a destination with the Dial
command.
After the call
completes I would like to let them optionally enter a new destination.
Currenty the call always d
Hi Marcin,
Marcin J. Kowalczyk wrote:
hI,
I've got problem with zaphfc kernel module. After I load into kernel i
receive something like that into syslog:
0 16:56:42 viperprint-de kernel: zaphfc: empty HDLC frame or bad CRC
received (framelen = 5, stat = 0xff, card = 0).
Jul 10 16:56:42 viperp
I have an image in CIP format that i'm trying to load onto a 7960 phone using
sccp. I don't know where to reference the data in which config file. Any
help available?
--
Edward F. Klimowicz
Voicenet Systems Administration
[EMAIL PROTECTED]
215.259.2131
___
hI,
I've got problem with zaphfc kernel module. After I load into kernel i
receive something like that into syslog:
0 16:56:42 viperprint-de kernel: zaphfc: empty HDLC frame or bad CRC
received (framelen = 5, stat = 0xff, card = 0).
Jul 10 16:56:42 viperprint-de kernel: zaphfc: empty HDLC frame
Martin Joseph wrote:
On Jul 10, 2006, at 1:23 AM, yusuf wrote:
Hi all,
I am running Asterisk 1.2.7.1, with a Sangoma A101 card in it. I also
have 2 Digium FXO cards, and I have premicells connected to the FXO's
. Calls come in off the Sangoma E1 cards, from a Philips PABX. The
problem I
Sorry Misreading of the wiki,
I have to use instead
Action: QueuePause
Interface: Agent/ID
Pause: 1|0
I write it to the mailing list so that people will be aware of the way
to use this manager command ;)
Tristan a écrit :
Hi List,
Just a little question about QueuePauseMember()
I use it in
> > > > > > Hi everyone,
> > > > > >
> > > > > > I have Asterisk SVN-trunk-r7498 running for a few months and
I'm quite
> > > > > > happy with it. However, I am experiencing a quality issue
with my AVM
> > > > > > Fritz!card PCI which is used with chan_capi. When somebody
calls me on
> > > > > > t
Hi
I've setup asterisk as client, it was working very fine 2 months
before, but now there is no audio on both side, as i'm on same network
as provider.
My IP = 111.111.20.*
Provider Host IP = 111.111.10.10
sip.conf
[general]
context=sip-incoming
bindport=5060
bindaddr=0.0.0.0
insecure=very
srvlo
Hi List,
Just a little question about QueuePauseMember()
I use it in the manager with the following action:
Action: Command\r\n
Command: PauseQueueMember(|Agent/$id)\r\n\r\n;
where $id is the agent's ID
but the agent is still taking calls...
Do I have to use the phone number agents ar
Hi all,I'm stuck here! I am trying to get DUNDi to work and it seems DUNDi is working accept the IAX part I think. I'm trying to let an extension from Trixbox1 call an extension on Trixbox2 with the use of DUNDi.
Ext 1301 * TrixBox1 * ---IAX2 * TrixBox2 *---Ext 160
I am using asterisk 1.2.9.1.
I had been using the option D to send some dtmf tones after the call is
answered.
This doesnt seem to be working for me now.
I am using and IAX2 connection from one machine to another.
my extensions.conf has:
exten=> 57,1,Dial(IAX2/boxa_to_boxb/597,,tD(101))
Whe
I'm not a a guru, but
Check this line:
exten => _9.,2,Dial(Zap/g1/${EXTEN})
do you really want to dial digit 9 through your ZapLine? are you
connected to another pbx?
If you don't want do dial 9 to PSTN line , but you want your users to
dial 9 to place outgoing calls, try this:
exten => _9.,2,
Are you sure they are sending you all 10
digits and not just the last four? Our provider just sends the last four digits
on DID. If this is the case you would have this:
exten => 4567,1,Answer()
exten => 4567,1,DIAL(SIP/user,20)
Hope this helps.
From:
[EMAIL PROTECTED]
[m
Hi,
I have configured digium tdm04b card with asterisk
on debian. Incoming call is ok. But outgoing call has problem. Would you give me
advice ?
Here is my config files.
zaptel.conf
fxsks=1fxsks=2fxsks=3fxsks=4
loadzone=usdefaultzone=us
zapata.conf
[channels]language=en
context=inc
Haven't read this whole thread (got way behind in this list :) )
Polycom has a softphone with video support also. Not sure if it is good
or not, just downloaded the trial version to test it out.
-Jonathan
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL P
Hi
first:
exten => 3031234567,1,Answer()
exten => 3031234567,2,DIAL(SIP/user,20)
if this still don't work try
exten => _3031234567,1,Answer()
exten => _3031234567,2,DIAL(SIP/user,20)
second:
You have in sip.conf [teliax] configured, did You specify context=
?
if yes, t
On Friday, June 23, 2006 4:08 PM Steven wrote:
> Exchange changes
>
> http://www.microsoft.com/exchange/techinfo/tips/mailtip01.asp
Looks promising and helps a bit. Still no use of precedence bulk etc. though.
Very poor detection of "lit" mails.
___
-
Steve Davies wrote:
I assume that iaxmodem talks to the PRI, and then HylaFax talks IAX to
asterisk? Any downsides/gotchas to this that I should be aware of?
No,
iaxmodem gives HylaFAX software modes that can also communicate with
Asterisk.
From the iaxmodem home page:
IAXmodem is a softw
I can confirm that the 1.2.7.1 patch works with 1.2.9.1 as well.
On 7/10/06, Steven <[EMAIL PROTECTED]> wrote:
The metermaid changes in head are very different, but there is a working
1.2.7.1 patch in the bug tracker.
http://bugs.digium.com/view.php?id=5779
I believe that the 1.2.7.1 patch als
I think it's the same,
10 calls in 200ms = 50 calls in 1s
because 1s = 5 x 200ms
IMHO, is better to use seconds as period, because is more ease to compare
rate speeds of each codec that are in bits per second.
fabio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTE
Hi friends,At present, I am making outgoing calls using Teliax service with Asterisk. But, I am unable to receive calls. My DID number is: 3031234567. I am using SIP Server (Asterisk) setup, which is provided on Teliax website support. I have replaced my DID number i.e., 3031234567 in YOURNUMBER. B
On Mon, 2006-07-10 at 07:34 -0500, Mike Bates wrote:
> Are you talking about ZiPhone a USB device ?
>
> Mike
>
zphone is phil zimmermans (creator of pgp) encrypted rtp system. Unlike
SRTP this does not rely on the server itself to provide the encryption.
It also lets you be reasonably assured
Hi,
I set the sip.conf parameter call-limit=1 to limit outbound calls and
'disable' call waiting.
But internally, I want to enable transfers. If the call-limit=1, the
transfers fails.
Any help ?
Thanks all,
Alexandre
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Are you talking about ZiPhone a USB device
?
Mike
Simple Simon
http://www.simplesimon.com
- Original Message -
From:
Michael Graves
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, July 10, 2006 6:44 AM
Subject: Re: [asterisk-users]
Is this correct:
zaptel: make clean; make; make install
asterisk: make clean; make; make install
Will this recompile everything needed? I tried, but the meetme app still
does not get compiled (and no music)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of B
Joe Baptista wrote:
> On Sun, 9 Jul 2006, Andrew D Kirch wrote:
>
>> To some extent I see your point and have been on the receiving end of
>> one of Jeremy's tirades.
>> I've since decided that NuFone is an interesting study in whether your
>> business can survive
>> with only clueful customers.
Hello
Can any one may send me log when channel bank is work
Best regards
Viktor Tatianin
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You may need to recompile now that you've got zaptel/ztdummy
installed so that your install sees that the proper zaptel exists now.
On 7/10/06, Dean @ INKnBITs <[EMAIL PROTECTED]> wrote:
After using the trunk versions as below, it all compiled ok, and the polycom
acd is working great, but the mu
Hi Raymond,
Raymond McKay wrote:
Agreed. I have seen and heard of a lot of attempts to bring SRTP
support into Asterisk but the idea of SRTP just doesn't make sense to
me. Asterisk, and VoIP servers in general, are meant to be
communications services not security services. In my mind at lea
Maybe in Asterisk 1.4 SecureRTP application would do that.
Regards
Henry J. Cobb escribió:
Hi,
Is it possible to encrypt the conversation between two parties on SIP,IAX
or
ZAP channels?
Sure, setup a VPN.
You can get a Linksys VPN router for less than $100 and run whatever
protocol you like
On 7/10/06, Doug Lytle <[EMAIL PROTECTED]> wrote:
>
> Any pointers on how to diagnose or improve this would be appreciated.
>
Install HylaFAX and iaxmodem on your Asterisk box.
Thanks, I will do.
I assume that iaxmodem talks to the PRI, and then HylaFax talks IAX to
asterisk? Any downsides/go
Has anyone here tried to use zphone with SIP soft phones and Asterisk?
Michael
On Mon, 10 Jul 2006 07:35:34 -0400, Raymond McKay wrote:
>
>>> Hi,
>>> Is it possible to encrypt the conversation between two parties on SIP,IAX
>>> or
>>> ZAP channels?
>>
>> Sure, setup a VPN.
>>
>>
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