Nick Crocker wrote:
We have a test asterisk box setup and can call each other on our sip
phones and receive calls in on the PRI to our phones no problem. Our
problem is getting asterisk to allow us to dial out using our PRI.
Digium has instructed us that we need to strip the leading 9 from the
chawki hammoud wrote:
Hi:
Internet Bandwidth in my country is expensive so I am
trying to figure out a way to use the most of what I
have.
All the calls are between two servers only.
How can I reduce the ip header bandwidth to the
minimuim whether I am making one call or multiple calls?
Send
> I know it can be a real pain in the butt getting hold of the firmware,
> so any help in obtaining it relatively fast and painlessly would be much
> appreciated.
Can't help with the Cisco 7910, but I noticed that these two files are
floating around on the Gnutella network (Cisco firmware is sign
I can't tell you how to resolve your issue, but I can tell you about
mine. I was fighting for setting my outgoing number (MSN /
bri_cpe_ptmp), and showing or hiding the number, with Swisscom operator.
Showing or hiding the number is resolved by the CallingPres command. For
me, values 0 and 32 work
Hi:
Internet Bandwidth in my country is expensive so I am
trying to figure out a way to use the most of what I
have.
All the calls are between two servers only.
How can I reduce the ip header bandwidth to the
minimuim whether I am making one call or multiple calls?
Here is a quick script that will parse extensions.conf, any files
included via #include, and print out the sql commands to put them into
mysql.
I'll add on routines to do the same for sip, iax, and voicemail when I
get the chance.
Chris
--
Joel Duffield wrote:
Okay sounds like a stupid question but just to be clear do you have some
sort of timer on both machines?
Joel
And of course you would want more than one channel to see the benefits
of trunking.
--
Cheers,
Matt Riddell
___
ht
Robert Goodyear wrote:
On May 20, 2005, at 8:11 AM, chawki hammoud wrote:
--- Andrew Kohlsmith <[EMAIL PROTECTED]>
wrote:
http://www.mixdown.ca/~andrew/dump/rc.tc. It's what
Could you please tell me where and how to install it
Thanks.
GOOGLE. LEARN. DEPLOY.
You need a primer in IP network
Anton Krall wrote:
What re you guys doing for windows callerid from Asterisk besides using yac?
Any other working software?
I use:
MSN Messenger (this is a bit slow - uses centericq)
===
exten => s,2,System(/bin/echo -e 'Incoming Call From: ${
On May 21, 2005, at 3:46 PM, Johnathan Corgan wrote:
Robert Goodyear wrote:
Noted. To clarify, will dropping back to runlevel 3 still ensure a
smaller set of processes that would be as non-intrusive as if I had
installed Linux with console/command line support only or would there
still be s
On 5/21/05, snacktime <[EMAIL PROTECTED]> wrote:
> Got another question now after digging into this. How are regular
> include statements and ignorepat implemented in realtime? Do I just
> add them as additional fields to the extensions table I am using?
>
> Chris
>
Well that doesn't make any s
What re you guys doing for windows callerid from Asterisk besides using yac?
Any other working software?
I have the tapi driver installed but all software I have tried doesn't seem
to work or doesn't support the asterisk tapi driver.
Any suggestions?
___
Got another question now after digging into this. How are regular
include statements and ignorepat implemented in realtime? Do I just
add them as additional fields to the extensions table I am using?
Chris
___
Asterisk-Users mailing list
Asterisk-Users
Rodrigo Otavio de Fraga wrote:
Hi,
When I finished a call, the asterisk give a message : FAILED TO INSERT
INTO DATABASE.
Make sure that the details inside cdr_mysql.conf are correct.
I.E. if it has username bob, password fred, host 127.0.0.1, run mysql -u
bob -p
Then it will ask you for a
Hi.
I'd really like to start using Broadvox or one of the many companies
that resell connectivity to their network, since they are the only
VoIP provider out there that solidly advertises full support for T.38
(I'd be using the openh323 stuff for faxing, since Asterisk doesn't do
T.38).
However,
Joel Duffield wrote:
The firewall I'm using is a Linksys BEFSR41 V3 it says that it uses "The
router uses NAT and TCP/IP port inspections" not stateful inspections.
Make sure that your are using qualify=xxx for your IAX2 peers.
For example, if you set it to 400 (this is in iax.conf in the defi
You are in luck:
http://bugs.digium.com/view.php?id=4037
-Matthew
>> From: Richard Z <[EMAIL PROTECTED]>
>> Reply-To: Richard Z <[EMAIL PROTECTED]>, Asterisk Users Mailing List -
>> Non-Commercial Discussion
>> Date: Fri, 20 May 2005 21:17:11 -1000
>> To:
>> Subject: [Asterisk-Users] acd wi
Somebody have a Ericsson WebSwith 100? I had 2 of them, but,
unfortunatelly,I have only one CD that broked inside the drive.
If somebody have the mencioned CD that can send me, I'll be grate.
Bernardino Campos
[EMAIL PROTECTED]
___
Asterisk-Users
I have a setup for a 30 incoming channels with telcel. The incoming is
R2, they told me the outgoing is MF not R2. If the other channels are
fxo, you should change your zaptel.conf so you can use zapata.conf
and comment out those channels on unicall.conf.
ia
On 5/20/05, Andres Maduro <[EMAIL PRO
Snacktime,
I've found that the pipe thingy ( | ) is needed anywhere in your
extensions table that a comma ( , ) would normally be.
In your SIP peers/users/friends table/s, you need a semicolon ( ; ).
This is as much as I know for sure at present :-)
snacktime wrote:
On the wiki it
Your explanation is really messy but from what I understand it seem
like you just want to be able to take incoming call from one asterisk server
and forward it to another asterisk sever over IP that would terminate the call
but dont connect calls untill B party answers the phone on second
asterisk
On the wiki it say's that if you use the Goto commands you need to
replace ',' with '|' in the app data field. But in the examples it
uses '|' in place of ',' in the Dial command also in a couple of
places.
Is it safe to replace ',' with '|' everywhere in the app data field
when using realtime?
What does it mean? How to solve it?
-- Executing Dial("Local/[EMAIL PROTECTED],2",
"IAX2/[EMAIL PROTECTED]/011886229xx") in new stack
May 20 18:02:17 NOTICE[31410]: app_dial.c:973 dial_exec_full: Unable to
create channel of type 'IAX2' (cause 3)
== Everyone is busy/congested at this time
Is iaxtel down?
Ive been getting this:
May 21 19:23:42 NOTICE[29984]: chan_iax2.c:2782 auto_congest:
Auto-congesting call due to slow response
-- IAX2/Iaxtel-12 is circuit-busy
-- Hungup 'IAX2/Iaxtel-12'
is it down or am I doing something wrong?
_
[EMAIL PROTECTED] is believed to have said:
>
>In the advanced options there are a few options for hang-up detection
>including tone detection, and silence detection. They also have parameters
>to adjust timing and sensitivy. IIRC, they are not enabled by default.
>
Nathan,
thanks: this is so
[EMAIL PROTECTED] is believed to have said:
>
>I don't know if it is a phone like issue or not, but try the SPA-3000 setup
>at http://geekgazette.com.
>-Kerry
>
Kerry,
thanks for the hint. A first try did not get better results, but I was
doing it very quickly..
Aldo
Robert Goodyear wrote:
Noted. To clarify, will dropping back to runlevel 3 still ensure a
smaller set of processes that would be as non-intrusive as if I had
installed Linux with console/command line support only or would there
still be stuff hanging around that's inextricably there because I
Aaron O'Hara wrote:
Tim,
Aside from the firewall logs in /var/log/messages, what tools did u find
most helpful for seeing SIP/RTP traffic?
What are some of the key things to look for to see if there's a problem?
Oh, I generally use tcpdump to grab the packets and save them to a file,
then
I have problems to install zaphfc on system base on
knoppix 3.8.
with kernel update to 2.6.11.8 with
bristuff-0.2.0-RC8d-CVS
please help !!
ztcfg
ZT_SPANCONFIG failed on span 1: No such device or
address (6)
lspci
:00:08.0 Network controller: Cologne Chip Designs
GmbH ISDN network contro
Yours could look totally different than
mine depending on how you route calls.
It will start with “exten” and
have the word “Dial” in it. You may have several lines that you
need to change...
In the below example change the r at the
end to an m.
exten => _NXXNXX,2,Dial,IAX2
yep
I have hold music other wise
looks like I am going to have to go in to the [EMAIL PROTECTED] and configure it via that
method
can you give me pointers on what the dial line
looks like so I dont screw this thing up??
they dont recommend editing this stuff bye hand
unless you know what you
what config is this found in?
email:
[EMAIL PROTECTED]
gmail:
[EMAIL PROTECTED]
msn messenger:
[EMAIL PROTECTED]
aim:
hanksmith5
skype:
hanksmith5
- Original Message -
From: "Jon Gabrielson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturda
Edit the extensions.conf and put an m at
the end of the dial line.
Do you have hold music otherwise?
Sincerely;
Gary Lawrence
ITcom.Net
866.4ITcom1
866.448.2661
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hank smith
Sent: Saturday,
On May 21, 2005, at 10:29 AM, Johnathan Corgan wrote:
Robert Goodyear wrote:
So: knowing that the X11 window GUI is a resource hog, is it
appropriate to use the GUI to install and configure various
components, then set RUNLEVEL to 3 once all is nicely set up and
running cleanly? Would this
use option m in the cmd dial.
Cheers,
Jon.
On Saturday 21 May 2005 03:26 pm, hank smith wrote:
> hello how do I set up asterisk to play music on hold to callers while it
> rings my phones? I am using the amp portal to configure the asterisk pbx
> just to let you all know. thanks
> hank
>
> em
Additionally you may want to check http://www.pkgsrc.org/ to see if
there is a package for NetBSD
[pkgsrc related tangent]
Sadly there is not one yet for interix the posix subsystem that is not a
sandbox (cygwin) not an emulator (bochs) not a virtual machine (vmware,
virtual pc) it runs side by
Tim,
Aside from the firewall logs in /var/log/messages, what tools did u find
most helpful for seeing SIP/RTP traffic?
What are some of the key things to look for to see if there's a problem?
Aaron
On Sat, 2005-21-05 at 14:04 -0600, Tim Pushor wrote:
> I have (had) a similar setup at one time.
On 5/21/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> I was reading on the wiki that Asterisk runs very solid on NetBSD.
> Can anyone comment? What is the definition of solid? Who is running
> Asterisk on NetBSD and which version of Asterisk are you running?
>
> Also, I know there is limited s
hello how do I set up asterisk to play music on
hold to callers while it rings my phones?
I am using the amp portal to configure the asterisk
pbx just to let you all know.
thanks
hank
email:[EMAIL PROTECTED]gmail:[EMAIL PROTECTED]msn
messenger:[EMAIL PROTECTED]aim:hanksmith5skype:hanksmith
The firewall I'm using is a Linksys BEFSR41 V3 it says that it uses "The
router uses NAT and TCP/IP port inspections" not stateful inspections.
Joel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Francisco
A. Lozano
Sent: Saturday, May 21, 2005 10:23 AM
To:
I have (had) a similar setup at one time. I'm running freebsd/pf on each
nat box. Asterisk is behind one, an xten softphone behind the other.
I watched the SIP traffic on both the incoming and outgoing interfaces
(pre/post nat) of each box. You can then generally see whats wrong, and
as a huge
Okay sounds like a stupid question but just to be clear do you have some
sort of timer on both machines?
Joel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adnan Ahmed
Sent: Saturday, May 21, 2005 10:30 AM
To: Asterisk Users Mailing List - Non-Commercial D
Try to use macro's I am not the one to ask about them, I couldn't give you
an example off the top of my head. But read up on them on the wiki, and i'm
sure they can do what you want very easily.
Joel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
St
All,
I have my * box NAT'd with all ports forwarded that are SIP related
(based on Wiki). I also have nat=yes, externalip=WAN address of
firewall, internalip=LAN network of *.
I have my Xten soft phone on a PC which is NAT'd behind firewall with
ports forwarded. I have also followed instruction
I was reading on the wiki that Asterisk runs very solid on NetBSD.
Can anyone comment? What is the definition of solid? Who is running
Asterisk on NetBSD and which version of Asterisk are you running?
Also, I know there is limited support for Digium cards on NetBSD, but
is there any support
Hello!
I got an interesting task to make with asterisk:
pstn--- * ---sip--- * pstn
This sounds common till now. What I have to make is:
1.the call is routed through PSTN to asterisk1 (#1) which has ISDN PRI
interface(s) - leg1
2.#1 doesn't pick up the call, neither rejects, it just place into st
On Sat, 2005-05-21 at 13:17 -0500, Guillermo Salas M wrote:
> On Sat, 2005-05-21 at 12:51 -0500, Guillermo Salas M wrote:
> > On Sat, 2005-05-21 at 11:08 -0500, Moises Silva wrote:
> > > other tip. you can use a text to speech software like
> > > naturalvoices from AT&T to develop your own soun
On Sat, 2005-05-21 at 12:51 -0500, Guillermo Salas M wrote:
> On Sat, 2005-05-21 at 11:08 -0500, Moises Silva wrote:
> > other tip. you can use a text to speech software like
> > naturalvoices from AT&T to develop your own sounds. Asterisk will try
> > to look for them in /var/lib/asterisk/soun
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Paul
> Sent: Thursday, May 19, 2005 4:10 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] LOOKING TO HIRE
> > Or rather, let me take that back. If you do
On Sat, 2005-05-21 at 11:08 -0500, Moises Silva wrote:
> other tip. you can use a text to speech software like
> naturalvoices from AT&T to develop your own sounds. Asterisk will try
> to look for them in /var/lib/asterisk/sounds/ if the default language
> is set to english, for spanish you can
Robert Goodyear wrote:
So: knowing that the X11 window GUI is a resource hog, is it appropriate
to use the GUI to install and configure various components, then set
RUNLEVEL to 3 once all is nicely set up and running cleanly? Would this
give the same effect as doing a minimal install or is the
You need to instal the module 'libipt_ipp2p.so'
--- Tom Fanning <[EMAIL PROTECTED]> wrote:
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > chawki hammoud
> > Sent: 21 May 2005 05:32
> > To: Asterisk-Users@lists.digium.com
> > Subje
This is a _very_ green question, but I am just beginning to explore and
learn Linux. Have to admit I avoided it for years due to other
obligations but discovering Asterisk has forced my hand.
So: knowing that the X11 window GUI is a resource hog, is it
appropriate to use the GUI to install and
> > > On Fri, 2005-05-20 at 12:06 -0300, Juan Luis Moyano wrote:
> > > > Hi, I've merged asterisk-0.9.0 on a gentoo (kernel 2.4) system using
> > > > portage ebuilds. I've just got NVFaxDetect .c files from Justin Newman
> > > > and I'm about to install them. I want to know which is the best way to
other tip. you can use a text to speech software like
naturalvoices from AT&T to develop your own sounds. Asterisk will try
to look for them in /var/lib/asterisk/sounds/ if the default language
is set to english, for spanish you can use the folder
/var/lib/asterisk/sounds/es/
More info is avai
Hi Guillermo. Currently im using several sounds that you can download in:
http://voip-info.org/wiki-Asterisk+sound+files+international
there are other good links there to explain how to make it work. But
actually is very easy.
If you have troubles making it work, let me know.
Best Regards.
- m
I was tracking down an error in my dialplan, .. but at the end it
showed, that I called a mobile phone in USA, which had a successfull
ENUM lookup, ...
Could that be?
[trunkUSA]
;
; USA & Canada long distance through trunk
;for ENUM drop 9
;exten => _91Z.,1,NoOp(trunkUSA)
exten => _91Z.,
Hi, It is possible to change the operator voice announcements to
spanish? What files do i need to replace o record new ?
Regards,
--
Guillermo Salas M.
Telconet S.A. Manta
Calle 15 y Av. 24 Esq.
Phone : 593 5 262 8071
Mobile: 593 9 985 5138
SIP : [EMAIL PROTECTED]
e-mail: [EMAIL PROTECTED]
www
Is there any way to have the user confirm the extension they are
looking to go to before transfering?
i.e.
"You pressed 5 4 3 3 2. Is this correct?"
1 - GoTo extensionPressed
2 - Enter extension again
Thanks!
Michael
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A
All, I was reading over the chanisavail command in the wiki and was
wondering a couple things.
First and foremost, what does this command do to determine if SIP is
available? All I could tell from a debug is that it simply checks to see
if the peer's port is open and doesn't run any callflows. Is
Hello ,
I want some tips guidance i am sure this topic discuss alot in list,i
try my best to solve it by myself try googling looking wiki everywhere
but no luck question is iax-iax trunking not working setting,trying
each n every option
server2 iax.conf:
[general]
bindport=4569
bandwidth=low
disal
Maybe you connections pass through a stateful firewall , and these states
die after some inactivity time... Check it.
- Original Message -
From: "Joel Duffield" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturday, May 21, 2005 3:45 PM
Subj
Hey steve
I remember a tip somewhere where they used a conference room and added all
the users into that conference muted, then kicked them out at the end of the
call. Sorry I can't remember at all where this was but it looked like it
could work. How did you get the autoanswer to work, I have trie
My * box keeps losing its registration to all the servers it is registering
to, the only way to fix it is to restart asterisk and then it works fine for
another 2 hours or so. I'm on a static IP, but this happens like clockwork
every time. I have seen other people that have this problem but never a
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> chawki hammoud
> Sent: 21 May 2005 05:32
> To: Asterisk-Users@lists.digium.com
> Subject: [Asterisk-Users] Boosting Internet Bandwidth for VOIP
>
> There was errors when I tried to start the scri
Hi all,
I am using a Sangoma with two PRI’s. As far as land phones,
the calls are fine but it refuses all cellphone calls:
My configuration in Zaptel is
span=1,0,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=2,0,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-61
chawki hammoud wrote:
There was errors when I tried to start the script
recommended by Andrew to boost bandwidth for voip
./rc.tc start
RTNETLINK answers: File exists
RTNETLINK answers: File exists
Looks like you are already running some type of QoS script, you'll need
to stop it did bef
> We have applied for LiveVoip termination.
> Although the account is now available via Web, I have not got any
> information how to set it up.
>
> Can anybody guide me to make my paid advanture a success?
You should have received an email with a sample config in it.
__
On Sat, 21 May 2005, Companity wrote:
> The sip phones and the internal phones on the PBX see the number of the
> calling party correctly (e.g. 040-987654321). Cause we can´t set a
> callerid to the public phone network (to show the calling party number),
> we want to show an extension of our numb
You can choose "Refresh Extensions" from the file menu in IPSwitchBoard,
that will delete all extensions and read all from your server again.
Thorben
"Ronald Wiplinger" <[EMAIL PROTECTED]> skrev i en meddelelse
news:[EMAIL PROTECTED]
>I had to cancel Broadvoice, but IPswitch does not like to d
> The *seller* determines the degree of "where to shipness" for Paypal.
> There are various degrees of paranoia possible on the seller's part.
Call you and raise you one!
Yup, in the end, the seller decides, period. The decision is based on
whether they think the people doing the transacation (P
Version 0.117 - 21. may 2005
* CAPI support has now been added to IPS
* Save all you speed dial number in the asterisk server and retrieve them
from any other intance of IPS. This way you can easily share all your speed
dial numbers between all users of IPS.
* All speed dial number can be saved wi
The two-channel ISDN is called Basic Rate ISDN, and it's a smaller cheaper
version of the Primary Rate ISDN (PRI).
With a BRI, you have two B-Channel (each 64kbps) and one D-Channel (16kbps).
It's very common in Germany, Spain and other european countries, as reliable
low-cost low-bandwidth da
We have applied for LiveVoip termination.
Although the account is now available via Web, I have not got any
information how to set it up.
Can anybody guide me to make my paid advanture a success?
bye
Ronald
___
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Asterisk
Hi all,
I try to setup Asterisk TDM400P (1fxs/1fxo) and
Hylafax as gateway to PSTN:
PSTNASTERISK+TDM400P-modem---HYLAFAX
|
SIP
>From modem I can dial number to fxs card however when
Hylafax server send fax to modem Asterisk failed !?
What's wrong ?
Regar
Hi:
The phone company here use Europe telephone system.
They offere ISDN lines with multiple channels,
starting with two and they add channels on demand.
Am I write to understand ISDN of more like E1 or T1
line. If not, what's the difference.
Hi Andy,
I have been trying to get 7910's work with *. I have tried with both
skinny and chan_sccp. Could you please instruct me about the
configuration? I have found some detalis about 7920 with sccp in
voip-info.org. But I haven't find any document for 7910. Please help
me to get them work proper
Hello,
On Sat, 21 May 2005, Marcin wrote:
> Hi,
> Is there a simply way to allow dialout from ISDN modem to
> outside number through Asterisk?
> I've got an server with an Asterisk and the following cards:
> 1. TE110 -- to telco
> 2. TE400P with one FXS to analog phone
> 3. Two HFC-S based cards
Hi,
Is there a simply way to allow dialout from ISDN modem to
outside number through Asterisk?
I've got an server with an Asterisk and the following cards:
1. TE110 -- to telco
2. TE400P with one FXS to analog phone
3. Two HFC-S based cards in NT mode
I'd like to connect ISDN modem to one HFC-S ca
Hi,
we are using
asterisk with Junghanns QuadBri and some sip phones. 2 channels are configured
in NT mode (ISDN PBX connected, internal ) and 2 channels are connected to the
public ISDN network (bri-cpe). We use Bristuff 0.2.0 RC8C from
Junghanns.
When a call comes
in from the public
Wilson Pickett wrote:
Just a quick note, if you typically ship to a different address than your
credit card billing address, you can file that address with your credit
card company. Most cards allow you to have mulitple addresses on file so
that your "Address Verfication" goes through correctly.
On 5/21/05, Chris Coulthurst <[EMAIL PROTECTED]> wrote:
> You can use the unavail.gsm sound file for more than voicemail. Why not
> have a Background() statement in its own context with the dtmf options
> you want from there:
>
> [leavenomessage]
> exten => s,1,Background(unavail.gsm)
> exten =>
Hi,
I am using ACD, i.e. application Queue(). Is there a way to use mysql
for the configuration file?
Thanks,
Richard
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I'm still relatively a novice with asterisk and am having issues with echo.
The calling party that calls a PSTN number doesnt hear the echo, but the
answered
side via sip or forwarded to another PSTN number over voip hears
excessive echo that
makes it difficult to communicate.
I've been playin
You can use the unavail.gsm sound file for more than voicemail. Why not
have a Background() statement in its own context with the dtmf options
you want from there:
[leavenomessage]
exten => s,1,Background(unavail.gsm)
exten => 1,1,Dial(LA/la/land)
exten => *,1,Goto(the-main-menu)
Man is that a r
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