On Thu, 10 May 2007, Francesco Peeters (Asterisk) wrote:
On Thu, May 10, 2007 23:44, Gordon Henderson wrote:
On Thu, 10 May 2007, Francesco Peeters (Asterisk) wrote:
It gives me pause though... Maybe it's time to get rid of my fixed
line...
;-)
No ;-) needed - I have friends on cable
Francesco Peeters (Asterisk) wrote:
On Fri, May 11, 2007 07:34, Armin Schindler wrote:
On Thu, 10 May 2007, Crazy Boy wrote:
Hi Friends,
Can anybody tell me other softPBX softwares like Asterisk?
- OpenPBX
- Freeswitch
Or try Googling for something like 'open source pbx'... Sheesh! :-o
Hello everybody!
I have a problem recording voices for my Asterisk menu.
I used the Record(/home/lazkano/bienvenido:gsm) function to record the menu
voices, but when I call from outside or from an extension the voice listen so
low.
is there any software to record my voice properly and convert
There is a small (and growing!) number of small businesses (and not so
small ones either!) who are moving towards using their broadband
(typically ADSL in the UK) connection for Telephony - and even installing
a 2nd ADSL line just for VoIP.
Indeed, many of our clients are doing just that. I
Steve Totaro a écrit :
Hi Steve
Your Zap conf files would be helpful. Zttest results? Cat
/proc/interrupts. Sharing interrupts?
No. Zap con files should not be relevant as we are using ISDN.
[EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ cat /etc/zaptel.conf
loadzone = us
defaultzone=us
Friday May 11, 2007 at 12:30PM EDT
A short reminder that you can connect with others in the asterisk
community by phone or SIP (or both obviously) during these
conferences. Anyone interested in asterisk is welcome to join the
conference.
Details are found here: http://x2z.eu
Past recordings
thankyou very much, i will probe it
byee
- Original Message -
From: Gordon Henderson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, May 11, 2007 10:35 AM
Subject: Re: [asterisk-users]
On Fri, 11 May 2007, Josu Lazkano Lete wrote:
Hello everybody!
I have a problem recording voices for my Asterisk menu.
I used the Record(/home/lazkano/bienvenido:gsm) function to record the
menu voices, but when I call from outside or from an extension the voice
listen so low.
is there
Stephen Bosch wrote:
3. I thought I might save some clutter by putting these cables between
the midspan and the patch panel, but then I discovered that the male end
of the cable is keyed, just as in the default AC cables provided with
the phones, meaning that they'll only work if plugged
2007/5/11, Stephen Bosch [EMAIL PROTECTED]:
Francesco Peeters (Asterisk) wrote:
On Fri, May 11, 2007 07:34, Armin Schindler wrote:
On Thu, 10 May 2007, Crazy Boy wrote:
Hi Friends,
Can anybody tell me other softPBX softwares like Asterisk?
- OpenPBX
- Freeswitch
Or try Googling for
Thanks for the pointers, I know about the Set(CDR..) function but I need
the codec that was negotiated in the Dial (once I have that its easy to
stick it into the cdrs as you pointed out).
Ie a call comes in as G729 Dial then negotiates GSM for the outbound
leg,
I want to log both these codecs in
Hi Folks,
Just in case there are any Mitel gurus here:
1) Is it possible to convince a non-dual mode 5220 phone to 'upgrade' to the
SIP firmware? I have inherited one that's Minet only.
2) I have a 5310 conference unit and 5235 phone in SIP mode, but someone's
lost the connecting lead. Can
Hi guys,
Is it possible to allow remote peers to connect to your local DUNDi
Asterisk box, even if you don't have them listed in the dundi.conf?
Alex
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To
Version 1.4.2 but to be honest I have no reason at all to suspect
that this is a problem with the asterisk software.
I've able to replicate this from a few different client net
connections and a across a few different linksys ata's. Where when
you call into the
host and enter the
On Fri, 2007-05-11 at 07:33 -0300, Roberto Pereyra wrote:
- OpenPBX
- Freeswitch
Other: sipX
Yet another: Yate
http://yate.null.ro/pmwiki/
ciao
Luca
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To
Hi all!
Is there a way to asterisk-gui to allow underline (as such cpd_tom)
in Names? It allows to [di]enable alphanumeric, but not underline noway.
Why such restriction in asterisk-gui if even asterisk users.conf allows
(and works fine) it?
Thank you,
Tom Lobato
Hi all,
I have been using asterisk to do such kind of thing,
But I must admitt, this is not 100 % conveniant (Mainly because Asterisk
isn't a SIP Proxy).
I just wanted to know if you knew/used some kind of SBC or packages which
would deal both with SIP AND RTP !
SER/OpenSER woulc be a good SIP
- Original Message -
From: Ed Nuñez
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, April 30, 2007 1:36 PM
Subject: [asterisk-users] Confference function
I would like to know if anyone here knows the answer to the following question
I
Great !!!
Thanks a lot !!
Nitesh Divecha a écrit :
Hello,
Here is my config: -
/etc/zaptel.conf
# T1 Configuration
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
span=3,1,0,esf,b8zs
bchan=49-71
dchan=72
span=4,1,0,esf,b8zs
bchan=73-95
I'm trying to configure a TDM808P card on debian. When I modprobe wctdm24xxp
i get this error
FATAL: Module wctdm24xxp not found.
FATAL: Error running install command for wctdm24xxp
I think i have successfully compiled the zaptel drivers, and the card
appears when i do a lspci
02:06.0 Ethernet
Hi Gavin,
You don't need queues to ring two phones, you can simply use the dial
command:
Dial(SIP/1SIP/10001) -- Would dial SIP extensions 1 and 10001.
Now if you want the ability to have multiple people waiting on the line for
those two extensions, that's when you need to look at
Situation such. There is an asterisk working as office pbx. 6 fxo - 18
fxs ports. All works perfectly, but some times in a week something
occurs. Could not catch what exactly yet. But symptoms such. The
asterisk infinitely writes the message of a type to broad gullies:
WARNING [20757] chan_zap.c:
On 5/11/07, Morgan Gilroy [EMAIL PROTECTED] wrote:
At the moment to find the codecs used I have to look though the sip
trace or show channels/show channel (annoying when you have 50+
channels).
Im just trying to find an easier and quicker way to keep track of the
codecs used to help with debug
I spoke to soon. Not an hour into the day this morning and we locked
up. I'm back to sip debug enable have turned sip history on, get me
the bug number and I'll contribute there.
Thanks,
Ken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken
Hello,
Here is my config: -
/etc/zaptel.conf
# T1 Configuration
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
span=3,1,0,esf,b8zs
bchan=49-71
dchan=72
span=4,1,0,esf,b8zs
bchan=73-95
dchan=96
/etc/asterisk/zapata-channels.conf You need to
Ps. Please start new messages from scratch rather then reply to existing
ones... (a mistake I've made in the past )-:
Woops..
I was ment to remove all that before I posted...
:-(
Sorry...
Best Regards
Gavin Spurgeon
Systems Administrator
Leigh City Technology College
[EMAIL PROTECTED]
On 5/11/07, Juliano Fernandes Schroeder [EMAIL PROTECTED] wrote:
I'm trying to configure a TDM808P card on debian. When I modprobe wctdm24xxp
i get this error
FATAL: Module wctdm24xxp not found.
FATAL: Error running install command for wctdm24xxp
I think i have successfully compiled the zaptel
Hi List,
Just a simple question for the list this time..
I need to setup 2 Phones than can Both ring when an incoming
call is made to a certain number...
I have done this 3,000,000s times with CCM and have no
problems with it, But it is the 1st time I have needed to do
this with Asterisk.
I
On Fri, 11 May 2007, Gavin Spurgeon wrote:
Hi List,
Just a simple question for the list this time..
I need to setup 2 Phones than can Both ring when an incoming
call is made to a certain number...
exten = 123,1,Dial(SIP/101SIP/102)
Gordon
Ps. Please start new messages from scratch rather
i did, and looking at the output, he processed all the modules into the
kernel just fine.
On 5/11/07, William Moore [EMAIL PROTECTED] wrote:
On 5/11/07, Juliano Fernandes Schroeder [EMAIL PROTECTED] wrote:
I'm trying to configure a TDM808P card on debian. When I modprobe
wctdm24xxp
i get
Gavin Spurgeon wrote:
Ps. Please start new messages from scratch rather then reply to
existing ones... (a mistake I've made in the past )-:
Woops..
I was ment to remove all that before I posted...
Actually, what he's referring to is that posters should start a NEW
thread for a new
Hi,
I have a question of using 2 SIP providers. Let's say I have provider A and
provider B, and I would like my calls to go to A, and then B if A wasn`t
available
Something like this would work:
exten = 1234,1,Dial(SIP/providerA)
exten = 1234,2,Dial(providerB)
exten = 1234,3,Hangup
But what
Bear with me this is a bit long winded. I am having some issues making
automated outbound calls over Broadvoice from my Asterisk 1.4.2 server.
For reference, none of the below issues happen when I make the calls to
VoIP phones attached to the Asterisk server. What I am trying to do is
call,
I need to setup 2 Phones than can Both ring when an incoming
call is made to a certain number...
Right.. After a little clicking around and getting a fresh pair of eyes to
help me look-over the web interface of SAIL, I found a way of
doing what I wanted by adding an Alias and making all inbound
Hi Morgan,
Am Freitag, den 11.05.2007, 10:32 +0100 schrieb Morgan Gilroy:
Thanks for the pointers, I know about the Set(CDR..) function but I need
the codec that was negotiated in the Dial (once I have that its easy to
stick it into the cdrs as you pointed out).
Ie a call comes in as G729
hi there guys!
how can I eliminate this message?
[May 11 11:00:46] WARNING[7039]: res_musiconhold.c:506
monmp3thread: Unable to spawn mp3player
[May 11 11:09:06] WARNING[7039]: res_musiconhold.c:424
spawn_mp3: Found no files in
'/var/lib/asterisk/mohmp3'
This is on debian etch 4.0
asterisk 1.4,
Stephen Bosch wrote:
Gavin Spurgeon wrote:
Ps. Please start new messages from scratch rather then reply to
existing ones... (a mistake I've made in the past )-:
Woops..
I was ment to remove all that before I posted...
Actually, what he's referring to is that posters should
Hi, Vitaly:
Vitaly Oborsky wrote:
Situation such. There is an asterisk working as office pbx. 6 fxo - 18
fxs ports. All works perfectly, but some times in a week something
occurs. Could not catch what exactly yet. But symptoms such. The
asterisk infinitely writes the message of a type to
You don't need queues to ring two phones, you can simply use the dial
command:
Dial(SIP/1SIP/10001) -- Would dial SIP extensions 1 and 10001.
I would strongly suggest you might want to create a queue for it anyway. One of
the things I've noticed in the past (and it's not asterisk's
Is it possible to allow remote peers to connect to your local DUNDi
Asterisk box, even if you don't have them listed in the dundi.conf?
I seem to remember something in the sample config file about a [*] entry being
possible...
One would assume that would cover connections from undefined DUNDi
Jon-o Addleman wrote:
I'm using the ices command to stream a conference to an icecast server.
This is working nicely, for the most part, but the volume is very low.
The streamed ogg vorbis audio is much quieter than what I hear in a
SIP
client, for example (on the same machine with the
Hi
Does anyone have a howto on how to set one of these up on Asterisk or Trix box
please?
I can make it SIP or MGCP so whatever you have ;-)
I have found one page but it isn't really a howto setup
Thanks in advance
Paul___
--Bandwidth and
Hi,
Does anyone know of a way to get a dry copper pair (also known as an alarm
line) from Verizon for less than $20/end? I know we have been able to get
them, but they come out to $40/month for a circuit.. and there's no
dial-tone over it
___
I have actually seen this behaviour on 1.2.x. I always assumed it was just
me dialing too fast for the ATA.
On 5/11/07, Bryan Laird [EMAIL PROTECTED] wrote:
Version 1.4.2 but to be honest I have no reason at all to suspect
that this is a problem with the asterisk software.
I've able
On Fri, May 11, 2007 at 02:36:46PM -0400, Matt wrote:
Does anyone know of a way to get a dry copper pair (also known as an alarm
line) from Verizon for less than $20/end? I know we have been able to get
them, but they come out to $40/month for a circuit.. and there's no
dial-tone
You might be able to try ordering it from a CLEC that can provision it over
UNE and sell it for considerably less. Depending on your area, their
interconnection agreement, tariffs, etc. So, your mileage may vary.
On Fri, 11 May 2007, Matt said something to this effect:
Hi,
Does anyone
Hi,
Does anyone know of a way to get a dry copper pair (also known as an alarm
line) from Verizon for less than $20/end? I know we have been able to
get
them, but they come out to $40/month for a circuit.. and there's no
dial-tone over it
around here (Canada) its a tariffed service
Let me see. Dry pair, $40 for the circuit.
Hardware for each end, $0.
Not paying verizon for DSL or PTP T-1 service? Priceless.
It's a BANA circuit, btw, in Verizon territory.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Balashov
Sent:
Nigel,
You cannot upgrade a non-dual mode 5220 to SIP.
If you are referring to the cable that connects the 5310 to a 5235, that
is a standard CAT5 straight-through cable.
Barry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nigel
Kendrick
Sent:
Yeah tried that. The CLEC said that one end of the line has to end on
their equipment.
On 5/11/07, Alex Balashov [EMAIL PROTECTED] wrote:
You might be able to try ordering it from a CLEC that can provision it
over
UNE and sell it for considerably less. Depending on your area, their
Who said I wanted to run DSL over it :)
On 5/11/07, Smith, Rick [EMAIL PROTECTED] wrote:
Let me see. Dry pair, $40 for the circuit.
Hardware for each end, $0.
Not paying verizon for DSL or PTP T-1 service? Priceless.
It's a BANA circuit, btw, in Verizon territory.
-Original
How far is the run? I'm wondering what you mean by $0 for hardware? I
typically use Ethernet extenders, but it has been a crapshoot on the quality
from Verizon.
What is a BANA circuit?
Finding someone who will even sell it to you has been somewhat of a game as
well.
Thank you. I will go through these softwares.
Luca Corti [EMAIL PROTECTED] wrote: On Fri, 2007-05-11 at 07:33 -0300,
Roberto Pereyra wrote:
- OpenPBX
- Freeswitch
Other: sipX
Yet another: Yate
http://yate.null.ro/pmwiki/
ciao
Luca
___
Who said I wanted to run DSL over it :)
no one - I'm sure you really just want to run 110baud modem over it :)
and I'm sure you probably don't want a handful of them between the same 2
locations either.
btw - here is an interesting strategy to get fibre or something better
than you have at
On Fri, May 11, 2007 at 05:32:33PM +0300, Vitaly Oborsky wrote:
Situation such. There is an asterisk working as office pbx. 6 fxo - 18
fxs ports. All works perfectly, but some times in a week something
occurs. Could not catch what exactly yet. But symptoms such. The
asterisk infinitely writes
From: Mike [EMAIL PROTECTED]
Date: Fri, 11 May 2007 11:06:35 -0400
Hi,
I have a question of using 2 SIP providers. Let's say I have provider A
and
provider B, and I would like my calls to go to A, and then B if A wasn`t
available
Something like this would work:
exten =
Hi,
In my opinion, you should keep your Asterisk, probably with PSTN Cards,
inside your network and just setup an OpenSer or even simpler another
Asterisk server on your DMZ.
This way you will enable ENUM and URIs for your Clients, and will prevent
much better any DoS, intrusion or any other
Matt et al,
Can you still do homebrew PTP T1 in the U.S. this way? I thought this was
nixed by the ILEC/CLECs years ago.
John Treble
Ottawa, Canada
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: May 11, 2007 2:37 PM
To:
On Fri, 11 May 2007, John Treble said something to this effect:
Can you still do homebrew PTP T1 in the U.S. this way? I thought this
was nixed by the ILEC/CLECs years ago.
It's logically possible. But if you're trying to do T1 over a single
pair, you'd have to break it out using
Hello,
I have very annoying problem with asterisk 1.4.4:
Every evening when I have peak load asterisk crashes, peak load is only
over 20-30 sip-to-h323 simultaneous calls. Nothing special in logs after
crash. Load average never was higher than 0.3, asterisk never uses more than
12% CPU (according
Update:
I was able to obtain another VSP to try and rule out Broadvoice. Seems
that either my Broadvoice settings, or something on their end is causing
the brief screech noise upon playing the first sound.
However, with this new VSP I still have the AMD (Answering Machine
Detect) problem
On Fri, 11 May 2007, John Treble said something to this effect:
Can you still do homebrew PTP T1 in the U.S. this way? I thought this
was nixed by the ILEC/CLECs years ago.
It's logically possible. But if you're trying to do T1 over a single
pair, you'd have to break it out using
So we know, and I know, that a dry copper pair has no load coils, etc.
Generally sells for about $20/line.. sometimes less.
Is there something that iLEC will sell that has load coils in it? Like say,
if I wanted to run voice over it, and didn't care about data?
IE.. I know this is VoIP, but
On Fri, 11 May 2007, Matt said something to this effect:
Is there something that iLEC will sell that has load coils in it? Like
say, if I wanted to run voice over it, and didn't care about data?
I don't know that they'd necessarily sell you anything with load coils
*per se*, especially
On 5/11/07, John Treble [EMAIL PROTECTED] wrote:
Matt et al,
Can you still do homebrew PTP T1 in the U.S. this way? I thought this was
nixed by the ILEC/CLECs years ago.
Not according to Verizon (in my area anyhow), We tried it and it
didn't work. The verizon technician insisted it wasn't
On Fri, 11 May 2007, C F said something to this effect:
Not according to Verizon (in my area anyhow), We tried it and it didn't
work. The verizon technician insisted it wasn't real PTP copper and
therefore anything but analog voice might/should not work.
What is PTP copper? Unless it's an
On 5/11/07, Alex Balashov [EMAIL PROTECTED] wrote:
On Fri, 11 May 2007, C F said something to this effect:
Not according to Verizon (in my area anyhow), We tried it and it didn't
work. The verizon technician insisted it wasn't real PTP copper and
therefore anything but analog voice
On 5/11/07, Alex Balashov [EMAIL PROTECTED] wrote:
On Fri, 11 May 2007, C F said something to this effect:
Not according to Verizon (in my area anyhow), We tried it and it
didn't
work. The verizon technician insisted it wasn't real PTP copper and
therefore anything but analog voice
On Fri, 2007-05-11 at 18:44 -0400, Jon Pounder wrote:
On 5/11/07, Alex Balashov [EMAIL PROTECTED] wrote:
On Fri, 11 May 2007, C F said something to this effect:
Not according to Verizon (in my area anyhow), We tried it and it
didn't
work. The verizon technician insisted it wasn't
On Friday 11 May 2007 5:45 pm, Jon Pounder wrote:
again, I'm interested to know anyone whose actually done this, and what
the results were, since I have been thinking of the same thing for a
while.
I'd run about two dozen of these things using a variety of equipment.
Pairgain SDSL modems
Yeah ok. That doesn't help.
What I mean is I want a call to go out on ProviderA, UNLESS it's down and
then go to ProviderB.
I want it to ring 30 seconds and then Hangup if nobody has answers.
I DON'T want to dial both, only one or the other.
Mike
-Original Message-
From: [EMAIL
Quoting Greg Oliver [EMAIL PROTECTED]:
On Fri, 2007-05-11 at 18:44 -0400, Jon Pounder wrote:
On 5/11/07, Alex Balashov [EMAIL PROTECTED] wrote:
On Fri, 11 May 2007, C F said something to this effect:
Not according to Verizon (in my area anyhow), We tried it and it
didn't
work. The
On Friday 11 May 2007 7:46 pm, Jon Pounder wrote:
well actually there is dialtone on the unprovisioned pairs for the
most part, but you can only dial repair, the telco office or 911 on
them. I am not sure if its all pairs or just pairs that had a line
provisioned at one time. ANAC just replys
On Fri, May 11, 2007 at 07:46:18PM -0400, Jon Pounder wrote:
Wear a hardhat and toolbelt with a butt set hanging off it, and you'll
easily penetrate the collective :) I've had many a conversation with a
telco installer and for the most part if you know what you're talking
about they
What I mean is I want a call to go out on ProviderA, UNLESS it's down and
then go to ProviderB.
I want it to ring 30 seconds and then Hangup if nobody has answers.
This one's actually a bit more complicated than it first seems, since you need
to know how each provider reports status when it's
Mike wrote:
Yeah ok. That doesn't help.
What I mean is I want a call to go out on ProviderA, UNLESS it's down and
then go to ProviderB.
I want it to ring 30 seconds and then Hangup if nobody has answers.
I DON'T want to dial both, only one or the other.
Mike
Mike,
You had it correct
pedro noticioso wrote:
hi there guys!
how can I eliminate this message?
[May 11 11:00:46] WARNING[7039]: res_musiconhold.c:506
monmp3thread: Unable to spawn mp3player
[May 11 11:09:06] WARNING[7039]: res_musiconhold.c:424
spawn_mp3: Found no files in
'/var/lib/asterisk/mohmp3'
I'm no
I forgot to add that the built-in support for playing mp3s which
replaced, for some people, the mp123 program, requires asterisk-addons,
which also isn't packaged for debian! There are other possibilities
though. I think you could use mp321 plus sox to convert to the proper
sound format, for
Hi
Can somebody brief me the working of RTP mixer from MeetMe perspective.
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Drew Gibson wrote:
Stephen Bosch wrote:
Gavin Spurgeon wrote:
Ps. Please start new messages from scratch rather then reply to
existing ones... (a mistake I've made in the past )-:
Woops..
I was ment to remove all that before I posted...
Actually, what he's referring to is
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