On Sun, 23 Jan 2005, Andrew Kohlsmith wrote:
> Why would the heads come in contact with the platters on a powerfail? The
> arms are very rigid -- the heads only float a few thousandths of an inch over
> the platters -- something that I don't believe has anything to do with the
> platters spinn
On Sun, 23 Jan 2005 01:51:56 -0500, Andrew Kohlsmith > I have *no*
issues on inbound quality with sixTel. They *had* a problem where
> the first second of audio was cut off upon connect (Wait() did not help) but
> that seems to have been fixed.
I see this problem intermittently, typically during
I'm looking for a prepaid calling card platform
that:
* easily scales to multiple servers with a common database
for: redundancy, capacity, and performance
Looking to start with capacity to handle 100 simultaneous
calls and be able to easily scale to 1000+ simultaneous
calls.
* in additi
HI John,
It also depends which H323 channel you will use for this translation.
I can recommend you to use the chan_oh323 from inAccess Networks -
according to our experience it's much stable and bug free channel.
Our Asterisk based translation system is running much stable with
chan_oh323 ..
What handset? Some such as the Planet dont work.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED]
> Sent: Monday, January 24, 2005 1:10 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Music On-Hold problem
>
> My problem is
See www.websitemanagers.com.au/asterisk/
It will allow anyone to contribute their tools/etc... Of course, some
things are more suitable to the wiki (eg, dialplan snippets/info/etc).
Regards,
Adam
On Sat, 2005-01-22 at 13:12 +0200, Mike Tkachuk wrote:
> Hello,
>
> I dunno if it's really needed, w
On 24/01/2005, at 3:26 PM, Andrew Kohlsmith wrote:
On January 23, 2005 10:30 pm, Nick Bachmann wrote:
HDDs don't fail because they lose power.
Unless the heads crash, which can happen if power fails. I know HDD
manufacturers have done "head unloading" and such recently, but the
risk
is still high
Dear Fellow Ast-Masters
Has any one here experienced the following issue (or similar).
Setup Description:
- Asterix Server - 192.168.1.10
- Analog Phone connected through FXS to Asterisk Box - (ext) 2000
- SoundPoint IP 300 - 192.168.1.2 - (ext) 4004
Problem:
I have a very basic sip.con
Hi! All
I would appreciate if someone could advice me on how stable is sip-h323 &
h323-sip translation as well as how many calls can it handle when doing such
translation.( assuming single 2.8Ghz intel processor & 1GB RAM)
Regards,
John
--
_
I'm trying to take advice to use "zaprtc from bristuff" (from both a
posting here and references on voip-info) because I have a 2.4 kernel
SMP machine.
I've downloaded and installed bristuff-0.2.0-RC3a and now have the
modules zaphfc and zaptel loaded.
Running meetme says the extension is inva
On January 23, 2005 10:30 pm, Nick Bachmann wrote:
> > HDDs don't fail because they lose power.
> Unless the heads crash, which can happen if power fails. I know HDD
> manufacturers have done "head unloading" and such recently, but the risk
> is still higher if power is suddenly lost during a writ
Title: RE: [Asterisk-Users] Music On-Hold problem
It should work right off the install..
Make sure you have MPG123 installed and running.
_
From: Computer Onsite Support [mailto:[EMAIL PROTECTED]]
Sent: Sunday, January 23, 2005 3:10 PM
To: asteris
> And, in fact, some drives *do* have problems with sudden outages. Some
Relative to the cost of a cheap UPS, downtime is much much much more
expensive. You can power pretty much any single server you want for ~$150 CDN,
and shut it down cleanly when the power goes out. Compare $150 with the cos
On Jan 23, 2005, at 7:30 PM, Nick Bachmann wrote:
> As I understand, if HD activity is minimal, the probability of HD
> failure is significantly reduced.
HDDs don't fail because they lose power.
Unless the heads crash, which can happen if power fails. I know HDD
manufacturers have done "head unlo
Did you ever get DTMF to work reliably with LiveVoip. I am having the
exact same problems.
On Mon, 17 Jan 2005 20:22:30 -0500, Jess Coburn <[EMAIL PROTECTED]> wrote:
> Hello all,
>
> What my app does is accepts a call in on a Dial-In Number (DID) via
> IAX, and then prompts the caller for the to
On Mon, 24 Jan 2005 14:57:06 +1300, Matt Riddell wrote:
>Howard Lowndes wrote:
>> Is it possible to get the Festival command to read the text from a
>> system file rather than having it input as a text string?
>>
>> Is this a case of having to use AGI, or is there a simpler way?
>
>Most people wo
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello everyone,
As you know, we released Asterisk 1.0.4 earlier this week. However,
there was a callerid bug in chan_zap that has caused us to go ahead and
make another release. Asterisk 1.0.5 is available at all of the usual
locations.
I'm sorry for
Andrew Kohlsmith wrote:
On January 23, 2005 04:04 pm, Mike Sander wrote:
> Is the harddisk activity on a standard asterisk install such that I
> don't really have to worry if the power cuts??
Not typically; there isn't much writing going on, this is true. Are
you that cash strapped that a $75 UP
i have had some problems with music on hold. some of the handsets havnt been
able to put people on hold...
When using a grandstream 101 i push hold it puts the other end on hold but
doesnt play the music. although when i do it with a x-lite client it does put
the other end on hold and stats the
BTW they also an iax2 ATA
Try here for the iax2 phone
http://www.ngtel.de/products.php#1
Do you have a contact email for these guys? I couldn't see anything listed
on their site anywhere. Seems the site is in current development.
Matt
Hi Matt,
I was just getting ready to try to order a IP phon
Hi,Adi,
We provide the USB phone you wanted, it can access Asterisk natively. It
can support Skype,X-Lite,X-PRO,eyeBeam,StanaPhone,SJphone,Net2Phone,Firefly and
MSN too. To get more information about that, contact with me offline or goto
our website please.
Regards.
David at iaxta
There are heaps but why not use a headset
If you insist on usb handset then there are 3 listed here.
http://www.telecoms.co.uk/catalog/default.php?cPath=583_829_830
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adi Linden
Sent: Sunday, January 23, 2005
Adi,
Yes there are...
You can probably use the exact same skype USB phone with X-Lite or one of
the many other windows SIP softphones.
It is not a matter of being compatable with Asterisk so much as being
compatable with your Asterisk softphone..
In the X-Lite menu, system settings - USB Setting
You are very welcome!
All the Best!
Sergey.
Robert Augustyn wrote:
Thanks for your help Sergey.
robert
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
On Behalf Of Sergey
Kuznetsov
Sent: Sunday, January 23, 2005 8:00 PM
To: Asterisk Users Mailing List - Non-Com
There are a number of Skype USB phones available. Are there any when
connected to a Windows PC can access Asterisk?
Adi
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Howard Lowndes wrote:
Is it possible to get the Festival command to read the text from a
system file rather than having it input as a text string?
Is this a case of having to use AGI, or is there a simpler way?
Most people would use AGI for that (combined with the text2wave or
whatever program).
Thanks for your help Sergey.
robert
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sergey
KuznetsovSent: Sunday, January 23, 2005 8:00 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Can anyone recomentT1/PRI providerin
SouthOntar
I wanted to let everyone know that we have a new security list covering,
Security, of all things.
You can sign up at:
http://lists.digium.com/mailman/listinfo/asterisk-security
The idea here is to have a less busy list for security issues, as the -users
is hard to keep up with already. Fortuna
Henry Devito wrote:
BTW they also an iax2 ATA
Try here for the iax2 phone
http://www.ngtel.de/products.php#1
Do you have a contact email for these guys? I couldn't see anything
listed on their site anywhere. Seems the site is in current development.
Matt
_
Looking at the error I tried moving chan_modem* out of the modules
folder and asterisk started and its working again...
Not sure what changed in the chan_modem_i4l.so but removing it from the
folder fixed my problem.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
Hello,
is there anybody reading this who has experience with VoIP (IAX or not) on
Macintosh computers? If so, have you ever seen or heard of (even an
experimental, i.e., never marketed) VoIP application for any of the older
Mac OSs, such as 9, 8, or 7?
I can't quite believe that VoIP is such a re
BTW they also an iax2 ATA
Try here for the iax2 phone
http://www.ngtel.de/products.php#1
- Original Message -
From: "Erik Espinoza" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, January 23, 2005 6:40 PM
Subject: Re: [Asterisk-Users] IAXy'
Monday, January 24, 2005, 12:20:46 AM, Jay Milk wrote:
> I don't want to be a kill-joy, but after reading your various messages
> over the last few days, I think you're in over your head on this one. I
> suggest you first get your own * system up and running. Then,
> re-examine your goals. So f
Try here for the iax2 phone
http://www.ngtel.de/products.php#1
- Original Message -
From: "Erik Espinoza" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, January 23, 2005 6:40 PM
Subject: Re: [Asterisk-Users] IAXy's apparantly failing in th
Hello group
I just update to the newest CVS now I'm not able to get asterisk to
start. No error during the make or make install
I did a make clean before the make;make install
Any help would be great
Here is the output
asterisk -vgcd
Parsing /etc/asterisk/asterisk.conf
Parsing /etc/a
I got my PRIs from ISPtel as an add-on to my colo with MCI and thru
MCI. I'll try to find ISPtel web-site (if it's exists) thru
MCI's customer service. Actually Allstream's PRI will cost you around
700-750 CAD per month. It's not that bad.
I got just few PRIs with set of DIDs I need. This is e
Hi there,
I have setup Asterisk with a couple of Sipura SPA-841's and Grandstream
ATA's.
The problem is that with both of these devices the Unattended call
transfer process seems to be just like Attended but instead you hang up
as soon as you have dialled the number of the party your are
trans
I really appreciate your comments regarding the challenges that face a new
VoIP service. While it is true that there is always still much to learn in
both the VoIP arena and with a true business model that will try to bring
something new and exciting to the existing community, I still strongly
cont
Hi All: I am new to Asterisk so if my question sounds too
newbeeish then pleasebear with me. I have about 10 remote
locations which are collecting some data. Iwould like to upload that
data every night. All remote locations have56K modem. I was
wondering can Asterisk be used
I have one of these phones. I bought it off of eBay. Not sure where to
get them direct. You will need to load the proper image, in that I
believe it ships with SIP by default. Each protocol has its own image.
Erik
On Mon, 24 Jan 2005 00:54:48 +0400, Jean-Michel Hiver
<[EMAIL PROTECTED]> wrote:
>
Ken Godee wrote:
I'm currently playing with a Digium T100P card and 2 Grandstream
phones, things are working well. I wanted to move on to linking our
Definity G3R Rev 8.2 to the T100P. Everything that I've read so far
shows that you need a TN464 to accomplish this. We have a TN767E
availabl
I don't want to be a kill-joy, but after reading your various messages
over the last few days, I think you're in over your head on this one. I
suggest you first get your own * system up and running. Then,
re-examine your goals. So far, you don't seem to be adding anything new
to the VOIP communi
What is the CLI output you are getting?
Do you have a timer source installed?
On Sun, 23 Jan 2005 18:10:27 -0500, Computer Onsite Support
<[EMAIL PROTECTED]> wrote:
> My problem is: No matter what machine I install and configure Asterisk on it
> I just can't get the music on-hold to work. Is any
Thanks
You sure have to have experience ...:)
Do you know how I can contact ISPtel?
Sprint quoted me a realy high number.
btw: what do you get with your PRI
service?
robert
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sergey
KuznetsovSent: Sunday, January 23, 2005 5:54 P
Tom Ivar Helbekkmo wrote:
Me too, and I'd certainly use it in the original poster's stead.
However, he specifically said that he must have an IPSEC tool, and
OpenVPN is not IPSEC.
-tih
We are currently using OpenVPN too with good success. I'm not sure why
you would require IPSEC. I thought th
Is it possible to get the Festival command to read the text from a
system file rather than having it input as a text string?
I suppose I could put the text string into an Asterisk variable and
reference that in the Festival command, but then, how do I get the
contents of the file into the Asterisk
Hello * Users.
I need to be able to generate a Sip Notify message using PHP
AGI but have no idea how I can do that.
What I need to send is the balance of the prepaid card and
display it on the soft phones display.
Does anyone know how to do this?
Thanks in advance.
KF
600 is for the US only.
FXS impedence for
UK 370+620||310nF
Europe CTR21 270+750||150nF
Chris
- Original Message -
From: "Remco Barende" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturday, January 22, 2005 10:12 AM
Subject: Re: [Asterisk-Use
My problem is: No matter what machine I install and configure Asterisk on it
I just can't get the music on-hold to work. Is anyone of you out there have
such problem? If so what have you done to fix the problem? I've tried so far
three other computers and none of them I was able to get music on-hol
My wife brought to my attention just yesterday that this is happening on
all my inbound PSTN calls. I am using a ZAP interface, not IAX.
Adi
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Ken Godee wrote:
Does anyone know how to monitor * to see if they are receiving timing
slips
on a span connected to a T100P card? I am seeing b-channel restarts
quite
often and also getting "No D-channels available" warnings from time to
time.
Yesterday I had all the b-channels crash during a M
Sorry, I completely forgot. You have to have an experience how to use
the CRTC site =)
If you will click to "Public Proceedings" at the top of the main page
you will be redirected to
the page witch will show you the most of the useful information.
At that page in the "Telecommunications" Part o
On January 23, 2005 03:42 pm, Reid Forrest wrote:
> When dialing out using a SIP or IAX provider (Broadvoice, SimpleTelecom,
> VoicePulse Connect) I often find that after the call is answered the first
> few seconds of audio are cut off (i.e. I don't hear the called party). This
> usually results i
Sergey,
Thanks for the input.
I looked at the crtc site did few searches but I guess I do
not know what to look for because I did not find anything related to
tariffs.
On the same note I am not able to find a Isptel web site
either I guess it is not my day today :)
robert
From: [EMAI
Does anyone have any experience with making an
adsi phone appear to have more than one line.
It seems like this would be a very simple and very useful
thing to be able to do. Ideally, it would be nice if you
could make the 6 soft buttons appear as lines 1-6 and
if you press one of the soft buttons
On January 23, 2005 04:04 pm, Mike Sander wrote:
> Is the harddisk activity on a standard asterisk install such that I don't
> really have to worry if the power cuts??
Not typically; there isn't much writing going on, this is true. Are you that
cash strapped that a $75 UPS with a serial port is
Yes I did. The same. It looks like there is some packet loss on the way
to my VoIP box.
Is there any optimal settings for jitter buffer for * ?
All the Best!
Sergey.
Andrew Yager wrote:
Hi Sergey,
Have you tried phoning from X-Lite to your PSTN line, or your PSTN
line to X-Lite? How is the audio
Why risk it? Just go snag a cheap UPS from your local store. Just
get something with enough run time to shut the system down gracefully.
On Mon, 24 Jan 2005 08:04:36 +1100, Mike Sander
<[EMAIL PROTECTED]> wrote:
> I'd considering an UPS backup system for my Asterisk server. I understand
> this
Nils Segerdahl wrote:
Im running bristuff-0.2.0-rc2b with Florians patch.
4 Billion hfc cards in ptp mode.
Works like a charm.
4 billion hfc cards! Wow that must be some server :)
Oh a brand name - I guess I missed the capital letter.
hehe
--
Cheers,
Matt Riddell
__
MCI does not provide voice trunks T1/PRI by itself. They resell it as a
add-ons to their IP solutions.
Sprint is expensive. Bell is quite expensive as well. Allstream quite
better in price. ISPTel is the least expensive
one but their customer support is not one of the best.
The best way to fin
I'd considering an UPS backup system for my Asterisk server. I understand
this is a linux issue, not a * issue, except for the following...
Is the harddisk activity on a standard asterisk install such that I don't
really have to worry if the power cuts??
As I understand, if HD activity is minimal
On Sun, 23 Jan 2005, Stuart Hirst wrote:
> Has anyone had any success using the Florz patch for zaphfc ?
>
> I have a * system with 2 HFC cards which is working fine with 2 PTP ISDN
> lines however the users are complaining of crackles on the line which I am
> assuming is related to the IRQ issues
Hello All,
Well, my explorations in to the world of VoIP is proving fruitful and in
the near future I am hoping to have my small VoIP online service up and
running ready to help promote the industry and hopefully gain a few
customers in the process.
Additionally, I will soon have my IAX and SIP s
For me this worked straight out of the box with [EMAIL PROTECTED] 0.3
Mike Sander
Operations Manager
Suite 4 / 38-48 Waterloo St
Surry Hills N.S.W 2010
Phone:(02) 8307 8877
Fax:(02)93182254
Mobile:0401 010 289
Email: [EMAIL PROTECTED]
Website: www.corporatebankinginternational.com
-Original
Michael Giagnocavo wrote:
Yes, the IAXy has faults, but until other IAX2 devices ship, it's the
only game in town. I know that the Farfon device will be out soon and
we'll be able to judge its quality at that time.
Or any PA168 phones, which are already out, and support IAX2, SIP, H323,
MGCP a
I posted this question a while back, and I'm posting again in hopes that
someone has some ideas. Sorry if you've already seen this.
When dialing out using a SIP or IAX provider (Broadvoice, SimpleTelecom,
VoicePulse Connect) I often find that after the call is answered the first
few seconds of aud
The subject says it all. After digging through latency and other issues
with all kinds of linux softphones, I've found that only * works alright
for me as a VoIP client.
Problem now is that, unlike other apps, chan_oss resp. chan_alsa grab
the card once and won't release it until shutdown, while
Martin Keding wrote:
Yes, The card is working fine most of the time. It just gets this message on
occasion and then Asterisk shuts down. I debating putting surge suppressors
on the PSTN lines. Could this be caused but a voltage issue from the Telco?
I was told the other day on IRC that telephone li
Hi,
I am considering A101/102/104 cards for my asterisk installations.
Has any of you used these or any Sangoma cards in such environment?
Any thoughts?
How do they stack up against Digium cards?
Any input would be greatly appreciated.
robert
___
Asteri
Steve Kann wrote:
Actually, the fatal issue is that asterisk's chan_oss or chan_alsa grabs
the sound device, so iaxclient can't do so.
I can't run it anymore (I used to could. . . ) even on a machine that
*isn't* running Asterisk.
I haven't changed anything else on my machine, so I think it's s
Has anyone had any success using the Florz patch for zaphfc ?
I have a * system with 2 HFC cards which is working fine with 2 PTP ISDN
lines however the users are complaining of crackles on the line which I am
assuming is related to the IRQ issues raised by Florz.
I have tried to use the patch bu
On Jan 23, 2005, at 9:11 AM, Bruno Hertz wrote:
On Sat, 2005-01-22 at 23:56 -0800, Kenneth Long wrote:
seem like some kind of port issue...
Actually, the fatal issue is that asterisk's chan_oss or chan_alsa
grabs the sound device, so iaxclient can't do so.
Probably. Both try to set up listeners o
On January 21, 2005 12:26 pm, Jay Milk wrote:
> G.711, at 64kpbs has a rated network load of 88kbps.
> So for each second of conversation, about 11KB are crossing the wires in
> each direction.
88kbps = 88*1024 bps / 8 bits/byte =11kB/sec, yes, in each direction.
> That means for a minute of two-
Yes, The card is working fine most of the time. It just gets this message on
occasion and then Asterisk shuts down. I debating putting surge suppressors
on the PSTN lines. Could this be caused but a voltage issue from the Telco?
Martin
-Original Message-
From: [EMAIL PROTECTED]
[mailto
Bump -- anyone?
> -Original Message-
> From: Jay Milk [mailto:[EMAIL PROTECTED]
> Sent: Friday, January 21, 2005 11:26 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Bandwidth, again, can someone check my math?
>
>
> I want to put a single voice-mail box on a remot
Kenneth Long wrote:
You really do not want to run Asterisk and X-Windows
on the same box.
That I understand... but this is not a production
machine. Loading is not an issue. I'm using icewm.
are there any other issues, besides loading, to not
run
x-windows at the same time?
Actually the issue seem
Michael K. Rodriguez User wrote:
I had a similar problem with power.
I connected Asterisk to a Belkin UPS 1200VA and the the server would boot up
and asterisk would load but the T1s on the Quad T1 card failed to come up. I
placed a loop on the card and still no change. Finally, I removed the UPS
an
Steve Underwood wrote:
Henry Devito wrote:
Is there ant chance of the Dialogic card model D/4PCI working with
asterisk ?
Word of caution: Even if you can buy the drivers and make this card
work with *, it is not meant to plug directly into a CO -48vdc talk
battery and 90-130vac ring voltage d
I had a similar problem with power.
I connected Asterisk to a Belkin UPS 1200VA and the the server would boot up
and asterisk would load but the T1s on the Quad T1 card failed to come up. I
placed a loop on the card and still no change. Finally, I removed the UPS
and the T1s came up.
Do know if th
Here is the console screen.
Starting simple switch on Zap/1-1
Executing Dial("Zap/1-1", "SIP/403") in new stack
Called 403
SIP/403-9c60 is ringing
SIP/403-9c60 answered Zap/1
Spawn extension (smvoice-incoming, 403, 1) exited nonzero on Zap/1-1
Hangup Zap/1
I have a grandstream 101
Doug Lytle <[EMAIL PROTECTED]> writes:
> I've had very good success with OpenVPN
> http://sourceforge.net/projects/openvpn
Me too, and I'd certainly use it in the original poster's stead.
However, he specifically said that he must have an IPSEC tool, and
OpenVPN is not IPSEC.
-tih
--
Tom Ivar H
Remco Barende wrote:
What would be the best / easiest VPN software solution. I would like
to install vpn software on the * server for roadwarriors to connect to
with laptops running windows. Ideally the vpn solution will not
require any additional software on the client side but will use IPSEC.
Stephan Schiessling wrote:
Try the variable PRI_NETWORK_CID instead of CALLERIDNUM
As written before, this did the trick for an installation based on the
bristuff install.sh skript. However, on another system I am running the
DEB packages provided by http://www.marlow.dk.
That system still does
> I have been getting the following message in Asterisk and it shuts Asterisk
> down, needing a reboot.
>
> "Power alarm on Module 2"
>
> I have
> (1) TDM400P with (2) FXS & (2) FXO cards
> (1) X100P card
>
> Any ideas?
Since nobody answered, I'll guess something :)
Did you plug the power on th
Kenneth Long wrote:
You really do not want to run Asterisk and X-Windows
on the same box.
That I understand... but this is not a production
machine. Loading is not an issue. I'm using icewm.
are there any other issues, besides loading, to not
run
x-windows at the same time?
Yes.
Whenever you scrol
>
> You really do not want to run Asterisk and X-Windows
> on the same box.
That I understand... but this is not a production
machine. Loading is not an issue. I'm using icewm.
are there any other issues, besides loading, to not
run
x-windows at the same time?
Could you give us the output of the console when you try the call ?
That would help us to point you in the right direction.
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Hi
I'm running safe_asterisk, but get core-files in /tmp - how do I debug
them ?
I know gdb asterisk core.12370
and bt full
But it didn't show anything usefull for me.
Can anyone help me ?
(Running asterisk 1.0.2 with ast_data
/Hans-Henrik
-
Last from bt full:
priority=200, ca
Hi!
> I can run iaxcomm by itself...and I start up Asterisk
> on it own...
>
> But if I start Asterisk first, then launch iaxcomm
> I get this error:
You really do not want to run Asterisk and X-Windows on the same box.
Cheers, Philipp
___
Asterisk-
Hi,
After using Asterisk with a SIP hardphone for a couple of weeks I've just
installed a TDM400P card.
My hardphone - a 7940 - allows me to use a dialplan to decide when a
particular extension is complete and automatically trigger dialing. This
works well with my internal extensions, which ar
On Sat, 2005-01-22 at 23:56 -0800, Kenneth Long wrote:
> seem like some kind of port issue...
Probably. Both try to set up listeners on the IAX port
(4569 for IAX2). Disable or reconfigure one of them to
bind to a different port, whichever you want to answer
on it.
Also, don't forget to disable
On Sun, 23 Jan 2005 10:33:14 +0100 (CET), Remco Barende
<[EMAIL PROTECTED]> wrote:
> I would like to
> install vpn software on the * server for roadwarriors to connect to with
> laptops running windows.
OK, take a hard look at this before you get too far. Installing VPN
software *on* the Asteris
When we first has the sip fixup enabled, it worked just as you
described. I think what what happening is as follows:
1. Phones are configured for NAT
2. Cisco PIX "handles NAT" by rewriting headers so the phone doesn't
appear to be NATted (for SIP proxies that may not support natted
devices)
3.
Thanks to everyone who provided feedback.
[EMAIL PROTECTED] wrote:
> Folks,
>
> I'm curious to know how the volume of Asterisk-Users rates as
> far mailing lists go. This list sees over 200 messages per
> day, which has GOT to put it in the top 5%, doesn't it? I'd
> love to know if anyone has kn
On Fri, 21 Jan 2005 14:24:39 -0500 (EST), Hayden Myers wrote:
>I've been trying to setup asterisk with an Internet Line Jack card for
>sometime. I've been successful in configuring asterisk to handle incoming
>calls, make calls between sip phones, call the asterisk demo, and even
>answer the phon
Hello Stephan,
Another way is to set the callerid in your extensions.conf via exten
=> 807440,2,SetCIDNum(0${CALLERIDNUM}).
>
> Try the variable PRI_NETWORK_CID instead of CALLERIDNUM
>
This did the trick. I will go and update the Wiki,,,
Thanks and have a good weekend.
--
Best regards
Peer Olive
Try the variable PRI_NETWORK_CID instead of CALLERIDNUM
Peer Oliver Schmidt wrote:
Jens, thanks for the feedback.
>>I've added a ZAPHFC card to my CAPI based system. Calls coming in via
>>ZAPHFC do not forward the caller id to the SIP phones. Calls coming in
>>via CAPI do forward the caller id to t
Hi,
Il giorno dom, 23-01-2005 alle 10:33 +0100, Remco Barende ha scritto:
> What would be the best / easiest VPN software solution. I would like to
> install vpn software on the * server for roadwarriors to connect to with
> laptops running windows. Ideally the vpn solution will not require any
Hi list!
I'm sure the topic has been discussed but I could not find what I was
looking for.
What would be the best / easiest VPN software solution. I would like to
install vpn software on the * server for roadwarriors to connect to with
laptops running windows. Ideally the vpn solution will not
zaprtc does not work with smp systems, unfortunately. There is some
discussion on the wiki about the bristuff zaprtc module working with
multi cpu systems, however. Link:
http://www.voip-info.org/wiki-Asterisk+timer
Brian
On Sat, 22 Jan 2005 22:37:42 -0800, Spencer Nassar <[EMAIL PROTECTED]> wro
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