I have taken all of the text and put it into a message in the forums:
http://geekgazette.com/phpbb/viewtopic.php?p=14#14
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hank smith
Sent: Wednesday, March 30, 2005 11:39 PM
To: Asterisk Users
Hi!
Can someone help me with a problem I have with CAPI and dialing out
or in? Installed is a B1ISA from AVM.
I have installed chan_capi-0.3.5. In modem.conf I have this entries:
[interfaces]
driver=chan_capi
type=autodetect
dialtype=tone
mode=immediate
msn=144673
device = /dev/ttyI0
device =
Hi,
I would like to create extension, so user will have to enter password, and
later he will be prompt for a number to call.
My config looks like this (ONLY THE PART OF):
exten = 888,1,Ringing(),
exten = 888,2,wait(2)
exten = 888,3,Background,welcome
exten = 888,4,Authenticate(1234|a)
exten =
dear fellows,
i want to make a call from an ip phone to a local pstn number (say
021-699-8256) using a Wireless phone service. asterisk comes in b/w
the ip phone and the Wireless phone which is connected to fxs port#1.
i just want to dial a number (above mentioned) through ip-phone and
asterisk
Isamar Maia wrote:
Technically speaking not. But Sangoma's support seems to be pretty much
better.
My understanding is that to an extent when we buy Sangoma we're putting
the dagger to Digium. They're glad to use Asterisk as a selling point
for their hardware, but unwilling to donate
Its the most crappy piece of software i've ever seen.
Not to mention that even the installation files for the latest version
are incomplete. (try importing the .sql files).
Joachim.
Kanuri, Seshu (Company IT) wrote:
it is by far the most complete billing system.
Really? Are you using it now or
On Thu, 2005-03-31 at 10:01 +0200, Andreas Meyer wrote:
REASON=0x3302
This means Protocol error layer 2. Are you able to make outgoing calls
any other way using this card? Do you see anything relevant in 'dmesg'
when you make outgoing calls, or when incoming calls occur?
You don't need to
On Thu, 31 Mar 2005 02:11:46 -0600 (CST), Bartosz Wegrzyn - asterisk
[EMAIL PROTECTED] wrote:
Hi,
I would like to create extension, so user will have to enter password, and
later he will be prompt for a number to call.
Did you look at DISA? This does almost exactly what you're looking for.
When you say wireless service to you mean a cellular service or a cordless
phone? Why not just have an analog phone line plugged into the asterisk
server that can connect to the PSTN to reach the local number?
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
I love soekris boxes, but in my humble opinion the answer would be be no.
Just for yucks set up a 2-3GZ bix and compare it with a 4801, perhaps
you will com to a different conclution that I.
If some kind sole could port the codecs to use Serin's minipci
encryption card, than it might be a
This bit me too. Had to turn nat off on the 7960Gs
Kristian Kielhofner wrote:
Peter J VERNON wrote:
Guys..
I have Asterisk CVS-NHEAD-03/19/05-21:56:28 running on a box here and
have a couple of Cisco 7960s and a Grandstream phone.
I can make calls from the 7960. When I get a call placed to
Forwarding a call from Asterisk to Microsoft Live Communication Server 2005
via SER (to translate from UDP to TCP), I get a 'one-way' communication
(WMessenger user can hear voice but PSTN phone user cannot).
Running SER in debug mode, I found:
DEBUG: RFC3261 transaction matching failed
DEBUG:
i dont have anolog phone at my office. anyway i will arrange it for
instance. but can u do me a favour? plz resolve y queries...
i have connected a anolog phone to Fxs port-1 at asterisk machine now
plz send me the configuration of extension.conf to make outbound
calls. i had configured
Hello.
I have two hired pstn numbers with the same voip provider.
I want to distingish in the sip.conf file, what of two phone numbers was
dialed, but i don't know how to do the match, because the sip client and the
sip host are the same for both numbers.
How can i match in sip.conf by the
Isamar Maia wrote:
I don't understand this *love* for Digium. Digium is a commercial
institution, period.
Yes, but. They are a commercial institution which took an enormous risk
by giving away for free what is undeniably their most valuable product.
It was a gamble, as it were, of the family
--- [EMAIL PROTECTED] wrote:
At the moment all I know is that they have Siemens
PBX system. They will give me
more details soon.
since HiPath4x00 V1.0 you can use oh323 and HG3550
(STMI board in the HiPath) for interconnection between
Siemens HiPath4x00 PBX and Asterisk.
domé
Isamar Maia wrote:
I don't understand this *love* for Digium. Digium is a commercial
institution, period.
Yes, but. They are a commercial institution which took an enormous risk
by giving away for free what is undeniably their most valuable product.
So, if Linus Torvalds had a
Hi all,
Has anyone got the call screening sample to pickup DTMF correctly ? I
have tried with the latest HEAD release and the dial macro gets executed
all the way up until the Read command where it sits until the timeout is
triggered no matter what DTMF tones you send it. Asterisk responds with
Hi,
If I want a user to, while waiting for a transfer after responding to an IVR,
to listen to music instead of a ring sound, what is the change should i do in
extensions.conf? Is it on the IVR menu or on the optional extension
Txs,
Robson
___
On Thu, Mar 31, 2005 at 01:58:21PM +1130, Craig wrote:
From my investigations, I can't find any carrier in Au that does allow
it to be set outside the allocated range, can't even get one to set it
to our 1300 number, have been told the ACA doesn't permit it in Au, but
not certain on that.
On Wed, Mar 30, 2005 at 11:36:18AM -0800, Kenneth Porter wrote:
Excellent, thanks for the info!
I was mostly worried about opening privileged ports, but an initial test
showed only high ports opened.
I'd guess that only files asterisk needs to write need to be owned by the
asterisk
On Thu, Mar 31, 2005 at 02:35:07AM -0500, Kris Edwards wrote:
Well, I'm certainly not selling xten.. Perhaps my enthusiasm extends
from my disgust with everything else. In particular, kphone, and
sjphone. I have noticed latency with xten in meetme, but if I just dial
somebody it works better
Hi,
I have discovered something strange with my asterisk box. After having
figured out how to get incoming calls to work on my TDM22B (TDM400P) card,
i am having an issue in that when the outside caller hangs up the phone
before the phone is answered on extention end, the extention rings till
In article [EMAIL PROTECTED],
Rod Bacon [EMAIL PROTECTED] wrote:
I have noticed whilst being connected to the console of my * server,
that my PRI interface (Digium TE410P) periodically reinitialises itself.
This server is currently not actively used, and each time the reset
happens the
Hi,
is there a configuration in iax.conf to specify that if a call goes to
that peer, a second call should not be allowed.
Specifically, I do this:
Dial(IAX2/iaxcomm) # in extensions.conf for a specific extension
in iax.conf:
[iaxcomm]
type=friend
mailbox=20
On Mar 31, 2005 1:00 PM, Robson Ribeiro [EMAIL PROTECTED] wrote:
Hi,
If I want a user to, while waiting for a transfer after responding to an IVR,
to listen to music instead of a ring sound, what is the change should i do in
extensions.conf? Is it on the IVR menu or on the optional
On Mar 31, 2005 12:31 PM, Marc SCHAEFER [EMAIL PROTECTED] wrote:
Hi,
is there a configuration in iax.conf to specify that if a call goes to
that peer, a second call should not be allowed.
Specifically, I do this:
Dial(IAX2/iaxcomm) # in extensions.conf for a specific extension
in
hello from germany,
i'm using a TE110P in E1-mode with asterisk as a VOIPPSTN gateway. i can
dial out with the sip-phones and everything is ok, but when i dial a wrong
phonenumber, with a normal phone i will hear a message telling that, but
asterisk passes no audio to the phone, like it worked
We
require Asterisk configuration and support consultants to concentrate on the
"other" business issues. The expected functions of the consultant(s)
are:
* Help
in configuring setting up complete Asterisk system (h/w setup and LINUX setup is
handled by us). System may include Digium PRI,
I've setup * with TDM400P w/1 FXS, 1 FXO modules.
I've one analog phone connected to TDM400P FXS module, 1 PSTN line to
one of the FXO module(ZAP) , and IP phone connected to asterisk on
LAN.
The calls between SIPs and zap phone (fxs) are OK. But 2 issues
cannot be solved:
1. To dial to PSTN
Checking the oh323 configuration on asterisk console
gives the following result below. I'm editing the /etc/asterisk/oh323.conf file
to correct the parameters, but the result doesn't change. I didn't receive any
error massages during the installation of asterisk-oh323-0.7.1 channel driver.
In which directories I should install asterisk, chan_capi, and modem driver?
And did I forgot something to get asterisk functional?
what is best way to test quick is the pbx working, at this point I only have HFC
card for external isdn lines?
I have RH9 so Linux kernel should be fine?
Thank
Did you copy the oh323.conf file from the asterisk-oh323 package ?
Could you show what it looks like ?
Are the file permissions ok ?
Yves
Cenk Yabas wrote:
Checking the oh323 configuration on asterisk console gives the following
result below. I'm editing the /etc/asterisk/oh323.conf file to
You dont have any codecs configured in your oh323 conf. also FastStart with H245 tunneling should be enabled to get the
best call-setup out of h323.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cenk Yabas
Sent: Thursday, March 31, 2005 7:18
AM
To:
Checkout http://www.voip-wiki.org as it relates to asterisk. There are a
number of useful guides on how to setup and run asterisk. Btw, all the
config files should be located in /etc/asterisk. RH9 should be fine to run
asterisk.
Alex
-Original Message-
From: [EMAIL PROTECTED]
Simon,
I am not sure if I understand you question
properly. However, you can configure password for each user (peer or friend) in
corresponding channel configuration file (i.e. sip.conf)
HTH
Alex
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simon
After I changed from leastrecent I did reload asterisk and waited about
an hour and nothing changed. So I restarted asterisk and waited another
hour, but it was still calling the agents in the order that they are
listed in the agents.conf file.
-Original Message-
From: [EMAIL PROTECTED]
True. I think Digium's USA bias is clearly demonstrated by their lack
of a BRI ISDN product. Most of the rest of the world use it in
abudnace yet Digium do not see fit to service this market because it
is not big in the US. very poor...
On Thu, 31 Mar 2005 18:32:40 +0900 (JST), Isamar Maia
Hello,
I need to correct myself on one of the points I made in my reply last night.
As a very polite developer from Sangoma stated to me(with evidence I might
add)they have in the past and continue to today contribute code to GPL
Asterisk. It doesn't say so on their website but their developers
Hi!
David Woodhouse [EMAIL PROTECTED] wrote:
On Thu, 2005-03-31 at 10:01 +0200, Andreas Meyer wrote:
REASON=0x3302
This means Protocol error layer 2. Are you able to make outgoing calls
any other way using this card? Do you see anything relevant in 'dmesg'
when you make outgoing calls,
What about pricing of the Sangoma compared to Digium, is it comparable?
Can Sangoma card handle modem data incoming calls at all?
Selon mattf [EMAIL PROTECTED]:
Hello,
I need to correct myself on one of the points I made in my reply last night.
As a very polite developer from Sangoma
Has anyone had any luck with being able to register a DDI with SMS in
the UK ? If so, how have you done it ?
Anything I send to 0 with a reset message gets sent back to my
main ISDN number, even though I have specified a callerid within my DDI
range.
If I use the same code to dial a
Cross posted on purpose
FYI, just upgraded from cvs-head from March 23 to this morning (March 31).
All compiles and installs completed normal.
Loading asterisk via safe_asterisk (or asterisk -cdvvv) fails with the
standard oche... message. Piped the output to a text file and it
appears the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
hi,
we have the ser sip-proxy for registration and we forwarding
the call to our cisco gateway and it works.
but now we will forwarding the calls to the asterisk and
the asterisk shoud forward the calls to our gw (via sip not h323).
how must i
Hello,
I am trying to configure Asterisk to be used by two (or more) different remote companies, sharing the same instance of Asterisk on my host.
By setting specific entry contexts for each sip user, I can repeat extensions among companies.
My question is: is it possible to have repeated
On Thursday 31 March 2005 02:43, Brian Capouch wrote:
Isamar Maia wrote:
Technically speaking not. But Sangoma's support seems to be pretty much
better.
My understanding is that to an extent when we buy Sangoma we're putting
the dagger to Digium. They're glad to use Asterisk as a selling
On March 30, 2005 10:26 pm, Kristian Kielhofner wrote:
It is obvious that Asterisk/TDM support from Sangoma is (and has been)
secondary. Their cards support data like no other. Excellent. Voice,
on the other hand, appears to be immature.
I respectfully disagree. Sangoma's voice
--On Thursday, March 31, 2005 10:20 AM -0300 Dov Bigio [EMAIL PROTECTED]
wrote:
My question is: is it possible to have repeated users on sip.conf being
identified by their different passwords? I tried to do that but got an
authentication failure. Is there a way to do this? Or I should always
Rich Adamson wrote:
Cross posted on purpose
FYI, just upgraded from cvs-head from March 23 to this morning (March 31).
All compiles and installs completed normal.
Loading asterisk via safe_asterisk (or asterisk -cdvvv) fails with the
standard oche... message. Piped the output to a text file
I have recently installed Asterisk @ 8.0 and
loading it fine and setup the ip addressing and change the default
password. But when I access the gui from a computer on the network I
can pull up the gui but the amp link doesn't work.
http://192.168.1.x/admin doesn't work any leads on this.
My question is: is it possible to have repeated users on sip.conf
being identified by their different passwords? I tried to do that but
got an authentication failure. Is there a way to do this? Or I should
always have different usernames?
I think it would be less crazy best if you had a naming
March 31, 2005
Hello Asterisk-users:
I am interested in automatic configurations based on typical legacy
phone systems configurations.
In other words, after installing Asterisk- I don't want to have to
learn the mechanics of the PBX, I just want it to work out of the box.
Given that most
Hi!
I have this setup at a customer:
PRI - (port 1) TE410P (port 2) - PABC
|
Asterisk
Before the Asterisk part was inserted the customer claims that their
PABC automatic changed the clock acourding to daylight saving time
from the PRI.
Now the
On Thu, 31 Mar 2005, Eric Bishop wrote:
True. I think Digium's USA bias is clearly demonstrated by their lack
of a BRI ISDN product. Most of the rest of the world use it in
abudnace yet Digium do not see fit to service this market because it
is not big in the US. very poor...
Why on earth
Good day all
I'm looking for someone with good knowledge of the way the snom220
transfer
I want to know how to do a consultative transfer on the second call
I.o.w if a call come in,A and another call come in B and B asks to be
transfered to exten 200,I want to speak to 200 1st and the transfer B
Hi,
how to use Asterisk where I need to have lets say 40 analog lines. Any ideas?
Thanks,
David
___
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
Hi Maron,
Thank you for your answer! I use a simple cisco router 2621XM
as call manager with the following configuration:
interface
Loopback79
description
ALT-VoIP-Gateway
ip address 10.xxx
255.255.255.255
h323-gateway voip
interface
h323-gateway voip
id Ldnxxx ipaddr 10.xxx
Hi List,
Can I use asterisk to enable call conferencing? I'm using ser for the UA's to
register, can I do something like if they dial a certain digits, it will
forward it asterisk and use asterisks meetme feature? can i do meetme using
only sip?
Sorry for my terms, hope you understand my
Hi All,
Is it possible to have only one SIP account that is shared by several
users ? I am currently setting up one asterisk box for a small company
(around 7 users). Can all of them make simultaneous call using only one
SIP account for termination or I have to setup individual account for
all
On Thu, 2005-03-31 at 14:10 +0100, Asterisk wrote:
Has anyone had any luck with being able to register a DDI with SMS in
the UK ? If so, how have you done it ?
Anything I send to 0 with a reset message gets sent back to my
main ISDN number, even though I have specified a callerid
[EMAIL PROTECTED] wrote:
I have recently installed Asterisk @ 8.0 and loading it fine and setup
the ip addressing and change the default password. But when I access
the gui from a computer on the network I can pull up the gui but the
amp link doesn't work. http://192.168.1.x/admin doesn't work
You could easily write some perl scripts that create you the right
config files depending on selectable configurations ... Just an idea.
Yves
Mark R. Watson wrote:
MWCNETWORK.COM
*
March 31, 2005*
Hello Asterisk-users:
I am interested in automatic configurations based on typical legacy
cpu load on te4xxp cards is very low, and now that they have echo
cancellers as add-ons cards, it will be even lower.
I can't speak on hardware compatibility as i never tried a sangoma card.
(But i can say that in the last year i've never had an issue with digium
cards and we have 8 in use.) The
On March 31, 2005 08:53 am, David Hajek wrote:
how to use Asterisk where I need to have lets say 40 analog lines. Any
ideas?
A pair of TE110Ps or a TE405P and an Adit600. This will get you any
combination of up to 48 ports, in groups of 8.
-A.
___
Hi Maron,
Thank you for your answer! I use a simple cisco router
2621XM as call gateway with the following configuration:
interface
Loopback79
description
ALT-VoIP-Gateway
ip address
10.xxx 255.255.255.255
h323-gateway
voip interface
h323-gateway
voip id Ldnxxx ipaddr 10.xxx
Hi,
In short : cannot register SIP phone (403 forbidden)
In long :
I am rather new to asterisk (and linux)
One month experience fighting my way in the doc wiki.
I worked before with the static '*.conf' files.
That worked but I need real-time.
I did compile a cvs head 29 mach 2005.
MySQL is
Eric would you be so kind as to assist me in setting up an acd que in astersik?
ALso I am interested in your domain name rocketgaming.com is that an
organization your involved with.
On Tue, 29 Mar 2005 21:50:38 -0600, Eric Rees [EMAIL PROTECTED] wrote:
I have a simple 4 person ACD queue using
On Mar 31, 2005, at 12:11 AM, Bartosz Wegrzyn - asterisk wrote:
Hi,
I would like to create extension, so user will have to enter password,
and
later he will be prompt for a number to call.
My config looks like this (ONLY THE PART OF):
exten = 888,1,Ringing(),
exten = 888,2,wait(2)
exten =
Hi ron,
Of course you can make meetme, what you need is a zaptel device or, if you
haven't any hardware, the ztdummy device. Install it (google), compile
asterisk again, define an extension and it should work, more or less ;-))!
Greetings,
Mario
-Original Message-
From: ron
Does anyone want to get involved in sednign traffic from North America
to Australia?
I want to provide a service were expatriates of Australia that live in NA ?
I hope I can post this type of question here?
___
Asterisk-Users mailing list
Hi,
I'm trying to compile channel_capi with current Asterisk CVS.
Asterisk compiled successfully but channel_capi (patched with all patches
needed, as suggested from some nice people on IRC #Asterisk) compilation
fails with:
app_capiFax.c:34:34: asterisk/channel_pvt.h: No such file or directory
You need a T1 card and a channel bank.
http://www.voip-info.org/wiki-Asterisk+Channel+Bank
Cheers,
Jon.
On Thursday 31 March 2005 07:53 am, David Hajek wrote:
Hi,
how to use Asterisk where I need to have lets say 40 analog lines. Any
ideas?
Thanks,
David
On Thu, Mar 31, 2005 at 05:05:21PM +0500, Muhammad Haris wrote:
The calls between SIPs and zap phone (fxs) are OK. But 2 issues
cannot be solved:
1. To dial to PSTN via zap phone, the setup in extensions.conf with
the following
exten = _Nxx, 1, zap/1
doesn't work.
I think
On Wed, 30 Mar 2005, C F wrote:
I think this bug is what you describe:
http://bugs.digium.com/bug_view_page.php?bug_id=0003067
Hope this helps.
I think so, but if I'm reading this correctly, the patch is already part
of the CVS version?
--Sean
Hi all,
i have installed X-Lite (xlite-linux-22)Suse 9.2
(2.6.8-24) and i have one way audio. The calling
number can hear me but i don't hear the called number.
Calling my mailbox works fine, i am able to hear my
messages. I use a usb handset from Tedas AG.
Another strange thing is that the pc,
Hi,
(re-posted since I did not see my original one after some time)
In short : cannot register SIP phone (403 forbidden)
In long :
I am rather new to asterisk (and linux)
One month experience fighting my way in the doc wiki.
I worked before with the static '*.conf' files.
That worked but I
I hope I can post this type of question here?
No you cannot. Please post this on Asterisk Biz List. That is the right
forum.
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon Walsh
Sent: Thursday, March 31, 2005 9:18 AM
To: Asterisk Users
[EMAIL PROTECTED] wrote:
I have recently installed Asterisk @ 8.0 and loading it fine and setup
the ip addressing and change the default password. But when I access
the gui from a computer on the network I can pull up the gui but the
amp link doesn't work. http://192.168.1.x/admin doesn't work
hank smith [EMAIL PROTECTED] writes:
do you know if it is gtk2?
It appears to be:
$ ldd xlite-linux-22
... blah ...
libgtk-x11-2.0.so.0 = /usr/lib/libgtk-x11-2.0.so.0
... blah ...
Regards, Bruno.
___
Asterisk-Users mailing list
-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
My understanding is that to an extent when we buy Sangoma
we're putting the dagger to Digium.
If anything puts the dagger to Digium it'll be their own inability to
engineer reliable hardware.
I appreciate what Digium
Is 586671 your ddi number ?
I've got a ISDN-32 if that makes any difference
Thanks for your help.
Julian
David Woodhouse wrote:
On Thu, 2005-03-31 at 14:10 +0100, Asterisk wrote:
Has anyone had any luck with being able to register a DDI with SMS in
the UK ? If so, how have you done it ?
Anything
Stephen,
You should be able to setup what you want. For example, asterisk sip peer
will register with your provider. The IP/analog phones will attempt outbound
calls which will be sent to this provider. What you need to determine is how
your provider bills for the calls. If they bill flat, then
I am having similar issue with Build 1.0.7
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, March 30, 2005 9:54 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Problem with Music on Hold. Please help
Folks!
I want to let everyone know that I have been trying to migrate from
1.0.6 to 1.0.7 last few days and I have come across serious issues in
the build 1.0.7. What I found are listed below. I would recommend
everyone to hold off any upgrade till the next build.
1)Voicemail - No Audio.
My understanding is that to an extent when we buy Sangoma
we're putting the dagger to Digium.
If anything puts the dagger to Digium it'll be their own inability to
engineer reliable hardware.
I appreciate what Digium has done for Asterisk, but reliability expectations
for phone
Eric Bishop wrote:
True. I think Digium's USA bias is clearly demonstrated by their lack
of a BRI ISDN product. Most of the rest of the world use it in
abudnace yet Digium do not see fit to service this market because it
is not big in the US. very poor...
And your EU bias is clearly
Hi,
I am not sure about PRI, but I noticed that * does send date/time info
on BRI. Perhaps installing bristuffed Asterisk would solve your problem
(bristuff patches libpri, so whatever applies to BRI probably applies to
PRI as well).
As for syncing PC clock from ISDN line, I haven't noticed any
Hi,
I am new to Asterisk and the first thing I have noticed about Asterisk
and Pingtels open PBX's is that they are using this dinosaur method of
running forums. It is a real pain getting every message in the forum and
essentially keeping my own database of issues. With that said are there
any
Guys.
While working with agents and queues, you have settings like timeout for
agents that dont answer within certain times but I have a question, if you
use autologoff, for example, setting the timeout for 15 seconds and
autologoff for 30 seconds, then, the agent wont be logged off since you
Rod Bacon wrote:
I have noticed whilst being connected to the console of my * server,
that my PRI interface (Digium TE410P) periodically reinitialises
itself.
This server is currently not actively used, and each time the reset
happens the card is idle.
Is this normal behaviour, or does it
Brian Capouch wrote:
I'll be glad to stand corrected, but if that assertion is in fact
true, we should be careful to do things that actually damage Digium's
ability to leverage their development of Asterisk with their hardware
sales.
It sucks that its such a fine line. On the one had, it is
Hi,
I am new to Asterisk and the first thing I have noticed about
Asterisk
and Pingtels open PBX's is that they are using this dinosaur
method of
running forums. It is a real pain getting every message in
the forum and
essentially keeping my own database of issues. With that said
Anyway, you should have this as your
first line in the
script.
#!/usr/bin/perl
___
I had #!/usr/bin/perl5 -w
I changed it to #!/usr/bin/perl and now it works.
Thanks for the help
__
Do you
On Mar 31, 2005 1:05 PM, Muhammad Haris [EMAIL PROTECTED] wrote:
I've setup * with TDM400P w/1 FXS, 1 FXO modules.
I've one analog phone connected to TDM400P FXS module, 1 PSTN line to
one of the FXO module(ZAP) , and IP phone connected to asterisk on
LAN.
The calls between SIPs and zap
I run http://geekgazette.com which has a forum, how-to guides, etc.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn
Sent: Thursday, March 31, 2005 7:27 AM
To: Linux - PBX, Asterisk
Subject: [Asterisk-Users] Are there online forums
Shaoul Jacobson - TELLINK wrote:
(re-posted since I did not see my original one after some time)
You must be more patient than that. Sometimes my posts take a good hour
to show.
I can access the database from a remote windows pc with access via
odbc locally with sql admin sql browser.
Here's an idea, Digium buys Sangoma with the massive amounts of cash they
are getting from venture capitalists and just integrate Sangoma designs into
their boards. Not sure how Sangoma would feel about this idea though.
MATT---
-Original Message-
From: Matthew Boehm [mailto:[EMAIL
On Thu, 31 Mar 2005, Steve Underwood wrote:
Eric Bishop wrote:
True. I think Digium's USA bias is clearly demonstrated by their lack
of a BRI ISDN product. Most of the rest of the world use it in
abudnace yet Digium do not see fit to service this market because it
is not big in the
Hi All,
On my * server I am getting echo on internal SIP calls. I.E. Sip 2
Sip. Calls going over the T1 via the T100p are fine.
I have used ulaw and gsm, gsm has less echo but it is still noticable.
All phones are snom 190s. Any ideas on what i can do to cancel this.
Thanks,
On Thu, 2005-03-31 at 00:32 -0500, Tim Bass wrote:
Hello All,
This is my first post. Sorry to post under such sad circumstances. Here is
the situation:
We installed a TE410P (today) in a SuperMicro 1U server today (Motherboard
X5DE8-GG), which was running great until installing this
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