I don't think creating a network without a single point of failure is
unreasonable.
-Original Message-
From: [EMAIL PROTECTED] on behalf of Stephen Bosch
Sent: Sat 8/4/2007 3:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Teliax Quality
On 30 Jul 2007, at 14:54, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hello,
I have been reading up on the capabilities of the Asterisk-Java
library. I believe that this library can act as an interface
between a Java GUI(custom softphone) and the Asterisk server.
Seems like the
In the O'Reilly Asterisk book it suggests that it is important to allow
BIOS specification of the PCI slot IRQs -- the Tyan won't let us do that
I don't think. Is this an issue with the Sangoma card?
Also comments about how suitable this machine is would also be
gratefully received.
Rory
On
Hello,
As we know, to connect Asterisk to PSTN network, we must use a PCI card
containing FXS and FXO modules like Digium TDM400P.
Now to connect Asterisk to a Frame Relay network what is the PCI card that
we need? Is the Ethernet adapter only is enough? or i have to buy another
type of PCI
Matt wrote:
Hi,
I have a client who has a system with a Sangoma 1 port PRI card with
echo canceling in it.For some reason, when the system comes up the
PRI will stay up for about 4-5 hours, then drop. zap show status
shows everything as ok, but we can't make or receive any calls until
I have verified it is EXACTLY 5 hours. At 5 hours, the PRI stops
working until I issue a restart on the wanrouter interface. I have a
call into Sangoma and Verizon to figure out who's problem it is. Can
anyone offer any thoughts?
On 8/5/07, Matt [EMAIL PROTECTED] wrote:
Hi,
I have a client
Same site, just a few lines later:
... you could run Asterisk and Hylafax with T38modem (by
www.openh323.org) on the same box and terminate T.38 calls ...
Gunnar
Hello,
From http://www.voip-info.org/wiki/view/Asterisk+fax you can read:
*Update Jul 2007:* For a T.38 gateway you can use
I am aware that Sangoma has amazing support, that's why we use their
cards =). It's just the most bizarre thing.. exactly 5 hours after
wanrouter starts (ie the PRI comes up) it will lose signaling until a
restart of wanrouter.
On 8/5/07, John Novack [EMAIL PROTECTED] wrote:
Matt wrote:
Hi,
On 4 Aug 2007, at 14:04, Michael Munger wrote:
IAX is not encrypted. What you're seeing in wireshark is likely the
authentication method you've chosen. (RSA or MD5)
In IAX that doesn't look like encryption - the challenge and
response are in hex strings.
You can encrypt it with a VPN as
On 5 Aug 2007, at 06:54, Douglas Garstang wrote:
I don't think creating a network without a single point of failure
is unreasonable.
It's impossible. I can't think of a single example where this
actually exists.
Getting even close is hideously expensive.
Tim, speaking for himself :-)
Hi
some problem with chan_alsa. Depending on the configuration I don't
get any sound output (output_device not set in alsa.conf - same as
output_device=default) or very strange output (output_device=hw:0,0)
when dialing into something like
exten = 10,1,Answer
exten = 10,n,Playback(soundfile)
I've RTFM and Googled but can't seem to get sip_autoreg to work (or
perhaps I'm just completely missing the point of it).
(what I'd like to do is avoid having to put explicit entries for every
SIP phone into extensions.conf).
Asterisk is creating entries in the (virtual) context sip_autoreg:
After a recent update I now get messages from one IAX channel about
the wrong format of the sound files when using Playback. I have tried
to force ulaw on that channel, but that doesn't stop the problem. Am I
missing something, like maybe g729 files?
On 5 Aug 2007, at 16:46, Russell Brown wrote:
I've RTFM and Googled but can't seem to get sip_autoreg to work (or
perhaps I'm just completely missing the point of it).
(what I'd like to do is avoid having to put explicit entries for every
SIP phone into extensions.conf).
Asterisk is
Tim Panton wrote:
On 5 Aug 2007, at 16:46, Russell Brown wrote:
I've RTFM and Googled but can't seem to get sip_autoreg to work (or
perhaps I'm just completely missing the point of it).
(what I'd like to do is avoid having to put explicit entries for every
SIP phone into
On Sun, 5 Aug 2007, Rory Campbell-Lange wrote:
In the O'Reilly Asterisk book it suggests that it is important to allow
BIOS specification of the PCI slot IRQs -- the Tyan won't let us do that
I don't think. Is this an issue with the Sangoma card?
Probably not. Once the system is built, have a
I had something similar happen recently with a new Sangoma 2 port PRI
card with HWEC and a new PRI provider. Ours would drop carrier about
once a week.
Sangoma had me upgrade the card firmware (not the drivers) which
fixed our problem. That is covered on their wiki.
Their support is
Note to Digium
I wish I could upgrade my wct4xxp drivers locally. I still have the v1
firmware on my card.
It is kind of hard (next to impossible) to pull it from a production
machine and ship it to Digium. That might take a week if all goes well.
Thanks,
Steve
Tom wrote:
I had something
Should have read firmware rather than drivers...
Steve Totaro wrote:
Note to Digium
I wish I could upgrade my wct4xxp drivers locally. I still have the v1
firmware on my card.
It is kind of hard (next to impossible) to pull it from a production
machine and ship it to Digium. That might
The site lists openh323.org, but that's the old t38modem I think. The
new one should be coming out here sometime:
http://www.voxgratia.org/downloads.html
However, right now I think that the SIP-compatible version is only in CVS:
cvs -z9 -d
:pserver:[EMAIL PROTECTED]:/cvsroot/openh323 co
Hi Gordon. Very many thanks for your comments.
On 05/08/07, Gordon Henderson ([EMAIL PROTECTED]) wrote:
On Sun, 5 Aug 2007, Rory Campbell-Lange wrote:
In the O'Reilly Asterisk book it suggests that it is important to allow
BIOS specification of the PCI slot IRQs -- the Tyan won't let us do
John Novack wrote:
The fact that ASCAP goes on campaigns doesn't make it any less absurd
(or, for that matter, any more likely that the average business is going
to be taken to task); the reality is that thousands upon thousands of
interconnects install PBX systems with radio ports on them
The thread is about music on hold. Things such as playing local radio
stations in a waiting room are not related. I don't think there is
anything illegal about using normal over the air radio and TV for such
purposes as long as it stays in the local market area.
Stephen Bosch wrote:
John Novack
Let's assume for a moment that it's impossible. That does not mean adding
additional servers and additional networking equipment does not add value, or
is a worthless endeavour.
-Original Message-
From: [EMAIL PROTECTED] on behalf of Tim Panton
Sent: Sun 8/5/2007 5:01 AM
To: Asterisk
Paul wrote:
The thread is about music on hold. Things such as playing local radio
stations in a waiting room are not related. I don't think there is anything
illegal about using normal over the air radio and TV for such purposes as
long as it stays in the local market area.
It is ALL
Stephen Bosch wrote:
Doug wrote:
Kewwl! How do you get the .wav files into the Polycom?
If it's not obvious, I'd be interested in this information too.
Most people seem to think you can't change the ringtones on the Polycom
sets.
This is the info I used:
On Sat, 04 Aug 2007 19:52:21 -0400
Matthew Rubenstein [EMAIL PROTECTED] wrote:
I currently have an AGI that calls the Festival text2wave app to write
a wav file that my dialplan plays into a call with the Background()
command. But the voice sounds terrible: like SAM, the 1980s 6502 voice
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Hash: SHA1
Tzafrir Cohen wrote:
What do you need that for?
'!' is pointless with asterisk -rx: with asterisk -r, '!' runs a local
command in a subshel (or starts a new subshell) by the local cleint
asterisk. It does nothing by the server.
So you might
John Novack wrote:
Paul wrote:
The thread is about music on hold. Things such as playing local radio
stations in a waiting room are not related. I don't think there is anything
illegal about using normal over the air radio and TV for such purposes as
long as it stays in the local
Worthless comes in many forms, Doug. If you're talking specifically
about the monetisation of hardware/effort, then it may indeed be
worthless by the simple fact that the cost may outweigh the net gains in
profits gained from the purchasing, configuration, and deployment.
Businesses are about
I found the firmware files on Sangomas website...but could not find
the upgrade procedure...can you advise on how to do it or provide a
link?
On 8/5/07, Tom [EMAIL PROTECTED] wrote:
I had something similar happen recently with a new Sangoma 2 port PRI
card with HWEC and a new PRI provider.
You know the problem is that most consumers think that it is possible to get
the best and the most reliable for almost nothing.
They go out with this expectation and get the cheapest, then when it bites them
a few times, they scream why me.
-- Original Message
ᎣᏏᏲ,
I am looking for VOIP (SIP/IAX) providers that support sending me RDNIS
info on forwarded calls. Are there any providers out there that support
this?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users
Hi,
I'm in a bit of a bind here, and I'd appreciate anyone who can help me
quickly. I have a customer who bought a Linksys 224P switch (PoE enabled,
with 24 ports). He also has 4 Polycom 501 phones, PoE bundle. Bizarrely,
the switch doesn't seem to power up the phone, even though all ports
Moises Silva wrote:
The latest versions of unicall (0.0.5) work with the latest spandsp
(0.0.4), but I have done nothing about making either of them work with
Asterisk.
Minor changes were needed to chan_unicall. Anyone interested in using
it can find it here:
If the provider is selling the service and you are paying for the
service the provider should give you the best service.
If the provider can't give you the BEST service at that price then the
provider SHOULD charge more and not waste my time.
The providers are charging LOW PRICES to get
At 07:02 PM 8/5/2007, you wrote:
I found the firmware files on Sangomas website...but could not find
the upgrade procedure...can you advise on how to do it or provide a
link?
I used this.
http://wiki.sangoma.com/sangoma-hardware
On 8/5/07, Tom [EMAIL PROTECTED] wrote:
I had something
On 8/5/07, Michael Joyner wrote:
ᎣᏏᏲ,
I am looking for VOIP (SIP/IAX) providers that support sending
me RDNIS info on forwarded calls. Are there any providers out
there that support this?
I have a hunch that les.net may offer this in their service
They don't do Jacksonville, FL USA / area code (904)
Baji Panchumarti wrote:
On 8/5/07, Michael Joyner wrote:
ᎣᏏᏲ,
I am looking for VOIP (SIP/IAX) providers that support sending
me RDNIS info on forwarded calls. Are there any providers out
there that support this?
I have a hunch
On Sun, 2007-08-05 at 20:32 -0500,
[EMAIL PROTECTED] wrote:
Date: Sun, 5 Aug 2007 19:08:25 -0300
From: Jo?o Paulo Vanzuita [EMAIL PROTECTED]
Subject: Re: [asterisk-users] text2wave Voices Improvements?
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type:
Working for two different VOIP providers I have had no success getting
this from upstream providers such as Level3, Time-Warner and Verizon.
-Jonathan
Baji Panchumarti wrote:
On 8/5/07, Michael Joyner wrote:
ᎣᏏᏲ,
I am looking for VOIP (SIP/IAX) providers that support sending
me
Steve Totaro wrote:
Note to Digium
I wish I could upgrade my wct4xxp drivers locally. I still have the v1
firmware on my card.
It is kind of hard (next to impossible) to pull it from a production
machine and ship it to Digium. That might take a week if all goes well.
The only way
On Sun, Aug 05, 2007 at 07:28:05PM -0400, SIP wrote:
Lots of information around about people who've had issues with
rebroadcasting the radio in their business establishments. However, it
is rare that ASCAP et al go after anyone but the big moneymakers. The
old Bloom County rule still holds
All,
In the design of an Asterisk system using Cisco 7900 series SIP phones
we are struggling with giving the reception folks (3) hardware that can
tell them the status of everyone in the office (10 or so) (On the phone,
out of office etc) Something that would register each of the extensions
we
On Mon, Aug 06, 2007 at 10:44:47AM +1200, Matt Riddell wrote:
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Hash: SHA1
Tzafrir Cohen wrote:
What do you need that for?
'!' is pointless with asterisk -rx: with asterisk -r, '!' runs a local
command in a subshell (or starts a new subshell) by
Jay R. Ashworth wrote:
On Sun, Aug 05, 2007 at 07:28:05PM -0400, SIP wrote:
Lots of information around about people who've had issues with
rebroadcasting the radio in their business establishments. However, it
is rare that ASCAP et al go after anyone but the big moneymakers. The
old
On Sun, Aug 05, 2007 at 07:08:25PM -0300, João Paulo Vanzuita wrote:
On Sat, 04 Aug 2007 19:52:21 -0400
Matthew Rubenstein [EMAIL PROTECTED] wrote:
I currently have an AGI that calls the Festival text2wave app to write
a wav file that my dialplan plays into a call with the Background()
easiest way of connecting multiple Asterisk boxes are trough IP network.
I know Digium cards supports HDLC encapsulation but i'm not sure about
framerelay.
On 8/4/07, Michael Munger [EMAIL PROTECTED] wrote:
What modules do you want on it?
Yours,
Michael Munger, dCAP
404-438-2128
is there any way to send busy tone to the calling party measn when i call 2
some body and phone would be busy then i got busy tone so i can guess party
still talking 2 somebody...
Steve Totaro [EMAIL PROTECTED] wrote: Sounds like you have call waiting on
the phones. You can disable this
on
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