Adam Hart wrote:
Try http://lists.digium.com/mailman/listinfo/asterisk-users
- Original Message -
From: Frank Latini [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 30, 2003 2:18 PM
Subject: [Asterisk-Users] unsubscribe
Please unsubscribe me from this list
people
It's in line 1 but I also tried line 2 just for
kicks but same problem...
When you did your tests with different ports and different phones, did you
use the same line cord?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Paulo Mannheimer wrote:
Hi All-
I'm working on a project that will have remote (internet)access to an *
server through SIP phones, either soft or hard ones.
Does anyone have any experience to share about which SIP product they
are using under similar conditions, as well as which codec is being
http://lists.digium.com/pipermail/asterisk-users/2003-October/024968.html
Any idea when these hacks will appear in CVS?
We should all hope never. That's why you call it a hack
because it works for only one very specific case and would break
SIP under Astrisk for most people. It even
At 22:00 29-10-2003 +0100, you wrote:
Just remember part of the design of the TE410P is that you can use T1
channel banks (you only get 24 ports) , if you buy these in America they
are significantly cheaper than E1 channel banks. Just assign one of the
incoming ports on the TE410P to be T1
Hi Wim,
It doesnt show the host (at least) until the phones have registered with
asterisk, because you've set the host to dynamic in your config. Either
verify if the phones register with asterisk, or set the host to their
static IP-adresses.
Best regards,
Florian
At 19:51 29-10-2003 +0100,
Thanks very much !!
I thinks it could be very useful for me
Regards
Rattana
- Original Message -
From: Peer Oliver schmidt [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 29, 2003 7:14 PM
Subject: Re: [Asterisk-Users] SIP client
Christopher Stephens schrieb:
Is
Hello all
I setup the Asterisk without Line Card.
But UA could not speak each other.
Error log was as follow
Asterisk Ready.
WARNING[5126]: File chan_sip.c, Line 444 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for
seqno 102 (Request)
Please give us suggestion to
Hi Walker,
I've put that up on
http://bugs.digium.com/bug_view_page.php?bug_id=464
Paul
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 30, 2003 11:50 AM
Subject: Re: [Asterisk-Users] Campon feature
On Thu, Oct 30, 2003
Hi there,
I'll need some help with this: Trying to establish an IAX2 link between
two servers works in one direction (SIP client with ulaw), but not in the
other (SIP client with GSM). The client used for this is X-Lite behind
NAT while both servers have a public IP (I playback an anouncement
Are these changes already done?
- Original Message -
From: Mark Spencer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Sent: Tuesday, October 28, 2003 10:32 AM
Subject: [Asterisk-Users] Upcoming Major CVS Changes
Just as a heads up, soon, I will be merging Thorston
hi all
Where can I find documentation about how to setup ZapRAS?
What I want to do (optimally) is to allow for automatic dial-up to
external sites, each having an ISDN router. Today we use a small ISDN
router for this, but it'd be a lot better, IMHO, to have asterisk do
this (functioning as a
Steve Underwood wrote:
I am taking note of people's messages about soft fax, even if I might
appear to be ignoring them. I am getting V.27ter finished off right
now, to flesh out the facilities in the software. V.27ter is used for
4800bps and 2400bps faxes - not critically important, but
Hello,
I have connected router Cisco 2600 to Asterisk with
H323 protocol.
Everything is workingfine
except...
Earlier when I was calling to router, router pick
up call and pass it to asterisk IVR system.
Then the voice says "Enter your extension" so I
enter my extension number, phone is
I am currently using Asterisk with G.711 codecs and in-band DTMF for
several Cisco 7960's
and an Audiocodes GW. When allowing out-of-band DTMF, I could use
voicemail menus and
anything else on Asterisk that required DTMF but I could not get the
DTMF relayed out of the
GW. Has anyone verified
Hello,
I have te following setup:
IAX client -(iax)- Asterisk -(h323) Cisco AS5300
At the present moment GSM codec is used betwee IAX client and Asterisk. G729
is used between Asterisk and Cisco AS5300.
I am thinking that switching from GSM to G729 between IAX client and
Hi,
trying to get the make progdocs to work.
Got
doxygen 1.2.18.3 installed, but during make progdocs I get lots of
sh:
line 1: dot: command not found
And
error:
problem running dot. Check your installation
Need
to know how to overcome this, and how to use the documentation
Yes... And I have tried different line cords just rull anything out
Does this make sence why this is doing this.. Could it be the phone it
self is broke?
Geoff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Crick
Posted At: Thursday, October
Paulo Mannheimer wrote:
That's weird. I've done some testing both with GS and Xten products, and
my iptraf readings show much more than your numbers.
It depends on how you did your tests..
If you ran iptraf an PC1 and made a call from PC2(X-Lite) to GS and your
sip.conf entry for either have
Is anybody aware of a database containing the types of nat
implementation in todays soho/consumer routers? I think it would make
sense for the community to have this database in order to avoid
symmetric nats.
If one such thing does not exist how about starting this database?
A stunclient for
Title: Polycom SoundPoint IP 500
The
SIP version of the IP500 runs the same firmware, etc as the IP600. The
config files are the same. The only difference is that the IP500 has three
lines instead of six. I believe that the model number is the same for all
IP500 phones, its just the
Might want to make sure your binaries are in the right place, or at least,
where the install script is looking for them - this was my problem.
--
Phillip C. Jackson
[EMAIL PROTECTED]
-
This mail sent through IMP: http://horde.org/imp/
Yes, I would like to see the camp feature
become part of the distribution. I know a few people who worked on ROLM
systems who swear there are no replacements just because of some of those features!
-Original Message-
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf
Hi :)
My employer is looking to move a call centre to a new office, and has
been increasingly frustrated with their legacy PBX (call-logging
licensing and hardware upgrade costs). So I've stepped forth as the
Open Source Pedant and suggested Asterisk so we can do all our own
CallerID / call
Gavin Hamill wrote:
Hi :)
My employer is looking to move a call centre to a new office, and has
been increasingly frustrated with their legacy PBX (call-logging
licensing and hardware upgrade costs). So I've stepped forth as the
Open Source Pedant and suggested Asterisk so we can do all our own
On 30/10/03 14:38, Gavin Hamill wrote:
Has anyone used ISDN30e in the UK with the Digium E1 cards?
Many people.
What options are there to stick on a couple of ISDN2's on top of that
should we require some 'backup lines'.
It's more a question of how to implement the backup lines - they're fine
Finally, are my options for handsets limited to IP phones via Ethernet,
or analogue phones via a channel bank (and then to another Digium E1/T1
card), or is there the possibilty to re-use proprietary handsets from a
previous PBX?
One option you might not have considered is connect your
This is exactly what I did.
I used Xten's GSM driver to call a Zap extension. Readings where 100
Kbits/s. Using uLAW returned 80 Kbits/s !!!
I also downloaded Xten pro to test their g729 codec, readings were even
worse.
That's why I'm so intrigued.
-Original Message-
From: [EMAIL
At 14:38 + 30/10/03, Gavin Hamill wrote:
The problem begins in that I only have a very loose grasp of the telco
world. Has anyone used ISDN30e in the UK with the Digium E1 cards? What
options are there to stick on a couple of ISDN2's on top of that should
we require some 'backup lines'.
I
Hi,
I'm trying to get asterisk to work with the Cirpack Softswitch. All I need for
now is that asterisk should forward all calls to the Cirpack. My sip.conf files
looks like:
[general]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Title: Message
Hi
Matt,
Thanks
for the reply, it helps alot. I saw your post on 10/20/03 to this list on
the review of the Polycom IP 600 review. That was extremely
helpful...thanks!
Questions:
1) Did
you get the call parking issue figured out? From what I could tell from
the post it
Hi,
I have been working with the Dial application and Gnophone. I would
like when the call is placed to the IAX client, an url is passed using
the Dial application. I cannot however seem to get the context right to
have the url passed onto the GnoPhone answering station.
Anyone have a
About every 10th call coming into my x1000p is not getting the audio it
should. You can see the messages scrolling on the console as they usually
would, playing the thankyou, then and menu messages. internal phones ring,
but when answered there is no audio. The caller gets a full volume echo
Paulo Mannheimer wrote:
This is exactly what I did.
I used Xten's GSM driver to call a Zap extension. Readings where 100
Kbits/s. Using uLAW returned 80 Kbits/s !!!
I also downloaded Xten pro to test their g729 codec, readings were even
worse.
That's why I'm so intrigued.
That is odd..
Hi, after hammering out a message, due to several hours of fighting
format. I have it resolved.
Now, Is there a variable in Extensions that can be used as the incoming
callerID from the calling party.
i.e. I would like to pass the url, with an attached CallerID string to
lookup in our
Should it work to have a multi-homed asterisk server with grandstream
phones on the internal network and another grandstream phone on the
internet and be able to call between them? I set the bindaddr to the
external IP and pointed the internal and external grandstream phones to
that address.
I am thinking of coding a solution using variables, Cut, and ChanIsAvail.
here is what i'm thinking of doing
Create a variable that contains the string SIP/gs1SIP/gs2SIP/gs3 ...
etc
check each phone with ChanIsAvail, and use Cut to remove its representation
in the string (if its not avail)
then
See README.variables in the Asterisk source directory.
On Thu, 2003-10-30 at 10:13, James Coberly wrote:
Hi, after hammering out a message, due to several hours of fighting
format. I have it resolved.
Now, Is there a variable in Extensions that can be used as the incoming
callerID
Hi
I have problem with Asterisk an 12sp+ phone. Asterisk's skinny
implementation doesn't correctly processes 'onhook' event from phone, so
voice channel stays opened and no new calls can be received by phone.
What i'm doing wrong? :)
--
Denis
Hi,
I have scanned
through the archives of this list and found a number of question about hardware,
but I just can not find the answer to my question. I am new to phone
systems, I got "drafted" to come up with a new phone system for our company (I
guess they figure since I know computers I
the simplest would be to get a t100p card and a 16fxs + 8 fxo channel bank
u can find them on ebay quite often, I got mine for 500$ (Carrier access CAC-I
with 12fxs and 12fxo)
cheers
Michael Bielicki
On Thursday 30 October 2003 6:00 pm, Just ME wrote:
Hi,
I have scanned through the archives
On Thu, 2003-10-30 at 11:00, Just ME wrote:
Hi,
I have scanned through the archives of this list and found a number of
question about hardware, but I just can not find the answer to my
question. I am new to phone systems, I got drafted to come up with
a new phone system for our company (I
run a tcpdump -s0 -x tcp port 2000 and send me the results offlist.
Jeremy McNamara
Denis Chapligin wrote:
I have problem with Asterisk an 12sp+ phone. Asterisk's skinny
implementation doesn't correctly processes 'onhook' event from phone, so
voice channel stays opened and no new calls can
Dave,
Should it work to have a multi-homed asterisk server with grandstream
phones on the internal network and another grandstream phone on the
internet and be able to call between them? I set the bindaddr to the
external IP and pointed the internal and external grandstream phones to
Hi,
I'm
having two problems.
First
I'm using the xten x-lite program to communicate with asterisk, and
everything works fine except that DTMFs are not transferred.
I've
set DTMFMODE to inband on both the sip.conf file and the x-lite configuration,
and still it doesn't work.
Anyone
You can only use inband dtmf if you are using the ulaw or alaw codecs.
On Thu, 2003-10-30 at 10:46, Shoval Tomer wrote:
Hi,
I'm having two problems.
First I'm using the xten x-lite program to communicate with
asterisk, and everything works fine except that DTMFs are not
transferred.
I think I understand the technical side of this, I'm after
opions...
For a low density Asterisk system (say 3 to 5 extensions)
what is the more preferable way to connect analog phones, a small
set of Cisco ATA-186 units or a couple Digium TDM400P PCI cards?
The criteria are, reliability, sound
Thanks, but no go.
I already used these. And it still doesn't work.
Anything I can do about the horrible echo in x-lite?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Thursday, October 30, 2003 8:55 PM
To: [EMAIL PROTECTED]
Subject:
You will want either a T100P, or a T400P. Then you will want a channel
bank that is modular enough to add a FXO card to it. With 5 lines of
FXO, the Adtran units will be a good choice as they are in units of 6
lines.
hmm what adtran unit is that the most popular adtran cb's used with *
are the
Dear,
I changed the host to a fixed ip address (host1=192.168.10.12 and
host2=192.168.10.13) now the ip address shows up in the 'host' field = ok.
Try to call, no succes, nothing happens!
What's wrong?
Wim
- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
To: [EMAIL
One other idea is to go 100% VOIP.
Get rid of the incomming analog lines. You can subscribe to
a VOIP service that will give you a POTS phone number and
route incoming calls to you using SIP.
In the office you buy 16 IP hard phones.
Now everything is done over Ethernet and you've not got
Hi,
For me, in order to get the same sound level as for a direct IP/IP call I
have the following values:
rxgain=10
txgain=15
Unfortunately, with this setting there is a little bit of echo.
To get a very small echo but with a lower audio level, the following values
work for me:
rxgain=0.8
I have 6 750s attached to my pbx server. The 850s have a lot of
functionality you don't really need.
-sb
-Original Message-
From: TC [mailto:[EMAIL PROTECTED]
Sent: Thursday, October 30, 2003 1:07 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Newbie hardware question
You
--- Peter Zeltins [EMAIL PROTECTED] wrote:
Well, I happen to be one of those very specific cases... ;) and looks
like
will have experiment with it myself. Although I'd hate to re-invent
the
wheel.
Peter
Checking e-mail this morning it looks like we have two independent
fixes that
I have a SwissVoice IP10S but can not seem to get it to have dialtone or
dial on *. Calls to or from 3001 don't work.
Any ideas are appreciated.
Robert
mgcp.conf is:
[general]
port = 2427
bindaddr = 192.168.0.110
[ip10]
host = 192.168.0.5
context = from-sip
line = aaln/1
The portion of
On Thu, Oct 30, 2003 at 03:00:09PM +, Alastair Maw wrote:
On 30/10/03 14:38, Gavin Hamill wrote:
Has anyone used ISDN30e in the UK with the Digium E1 cards?
Many people.
That's reassuring to hear :)
What options are there to stick on a couple of ISDN2's on top of that
should we
On Thu, Oct 30, 2003 at 03:24:24PM +, Fearghas McKay wrote:
I would just get another ISDN30 and enable extra circuits as required,
rather than add a couple lines here and there with ISDN2/BRI.
I think the point is that we've just about reached capacity on our 30
channels, and won't be in
While compiling Asterisk from one month
ago
cvs checkout -D "last month" asterisk
I got compiling error:
term.c:55: conflicting types for
`term_color'include/asterisk/term.h:47: previous declaration of
`term_color'term.c:98: conflicting types for
`term_prompt'include/asterisk/term.h:49:
On Thu, Oct 30, 2003 at 03:10:09PM -, Linus Surguy wrote:
One option you might not have considered is connect your existing PBX
to the back of Asterisk and thereby use it as a channel bank itself.
Very interesting :)
There *is* an 'S-bus' (which is the same as an 'S0-bus'?) I'm told,
Q931 is the RJ45 version that you just plug in to the line card.
Q931 describes the protocol and not the line presentation. However, you do
want to ensure that you ask for Q931 as although DASS/2 is an ISDN protocol,
it isnt the same as Euro-ISDN and not supported by Asterisk.
Linus
On Thu, 2003-10-30 at 20:28, Bartosz Jozwiak wrote:
While compiling Asterisk from one month ago
cvs checkout -D last month asterisk
I got compiling error:
term.c:55: conflicting types for `term_color'
include/asterisk/term.h:47: previous declaration of `term_color'
term.c:98: conflicting
It's only worth doing if you're going to route them directly to some
other kit, though, so Asterisk support for ISDN2 hardware is largely
irrelevant here.
I don't quite understand what you mean by this - we want to terminate
the ISDN30e ourselves, and have a couple of ISDN2s also there
I just did it.
When I call from H323 router and the call is answered I got then
segmentation fault.
- Original Message -
From: Dave Cotton [EMAIL PROTECTED]
To: ASTERISK USERS [EMAIL PROTECTED]
Sent: Thursday, October 30, 2003 4:41 PM
Subject: Re: [Asterisk-Users] Compile problem with
Is there any one out there using an MSI 240 Global Station with Asterisk? I didn't
see it listed on the hardware page but figured I would ask just in case.
Thanks,
Patrick
winmail.dat
At 19:24 + 30/10/03, Gavin Hamill wrote:
On Thu, Oct 30, 2003 at 03:24:24PM +, Fearghas McKay wrote:
I would just get another ISDN30 and enable extra circuits as required,
rather than add a couple lines here and there with ISDN2/BRI.
I think the point is that we've just about reached
I've done some reviewing of the archives for G729 and H323
experiences. The landscape of that query isn't pretty - lots of
pleas for help, and nor do I see too many answers. I have a
pending bid that requires some data before I can implement * on this
particular solution.
My question is
Hi Wim,
Citeren Wim Venneman [EMAIL PROTECTED]:
I changed the host to a fixed ip address (host1=192.168.10.12 and
host2=192.168.10.13) now the ip address shows up in the 'host' field = ok.
Try to call, no succes, nothing happens!
What's wrong?
That's a bit difficult to determine without
Citeren rnc Info Lists [EMAIL PROTECTED]:
I have a SwissVoice IP10S but can not seem to get it to have dialtone or
dial on *. Calls to or from 3001 don't work.
Were you able to configure the phones through their webinterface ?
You could try entering 'mgcp debug' and then power up your phone
Mixture of 7960's and ATA's for cordless phones... thats what I would do.
bkw
On Thu, 30 Oct 2003, Chris Albertson wrote:
I think I understand the technical side of this, I'm after
opions...
For a low density Asterisk system (say 3 to 5 extensions)
what is the more preferable way to
On Thu, Oct 30, 2003 at 07:42:39PM -, Linus Surguy wrote:
I think the person who replied meant that if you are having the lines
as backup in case of failure, you should also be considering failure
of the Asterisk equipment and therefore the backup lines should route
to a different
It's my understand that they are db levels. (And, if I remember my
electrical engineering classes from college, a 3db increase effectively
doubles the volume.) I hope that helps...
Jared Smith
On Thu, 2003-10-30 at 11:28, Dan wrote:
Hi,
For me, in order to get the same sound level as for a
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Ariel Batista wrote:
| Ok I have a question. I have Xten-lite working with our Asterisk
system and I am able to make and get calls. But the main problem is the
sound is very choppy and sometimes it cuts off words. I have tested it
with ulaw and alaw
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Just
MESent: Thursday, October 30, 2003 11:00 AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Newbie
hardware question
Hi,
I have scanned
through the archives of this list and found
On Thu, 2003-10-30 at 20:53, Bartosz Jozwiak wrote:
I just did it.
When I call from H323 router and the call is answered I got then
segmentation fault.
I haven't got any H323 only SIP and analog, I've had no seg faults.
--
Dave Cotton [EMAIL PROTECTED]
Citeren rnc Info Lists [EMAIL PROTECTED]:
I have a SwissVoice IP10S but can not seem to get it to have dialtone or
dial on *. Calls to or from 3001 don't work.
Were you able to configure the phones through their webinterface ?
You could try entering 'mgcp debug' and then power up your
Why not just use appqueue?
Is that the integrated quesolution that I config in queue.conf?
But as I've understood it might be a little tricky to get the users the
possibility to log in/out of groups in an easy way (each extension will
maybe be the member of up to four groups, and it must be
Can't
get asterisk to understand DTMF from x-lite.
Used
proposed configuration on the web. Still doesn't work.
Using
inband dtmfmode, still no go.
Help?
Vmail.cgi
doesn't work as well, error says Premature end of script headers: vmail.cgi
Shoval
You could also look at products like
http://sales.netxusa.com/vegastream/vega50.php
- Original Message -
From:
Andy
Hester
To: [EMAIL PROTECTED]
Sent: Thursday, October 30, 2003 3:46
PM
Subject: RE: [Asterisk-Users] Newbie
hardware question
I am thinking of coding a solution using variables, Cut, and ChanIsAvail.
here is what i'm thinking of doing
Create a variable that contains the string SIP/gs1SIP/gs2SIP/gs3 ...
etc
check each phone with ChanIsAvail, and use Cut to remove its
representation
in the string (if its not
James Coberly wrote:
Hi, after hammering out a message, due to several hours of fighting
format. I have it resolved.
Now, Is there a variable in Extensions that can be used as the incoming
callerID from the calling party.
i.e. I would like to pass the url, with an attached CallerID
Well, found the answer for the DTMF problem, and
guys, the voicemail is G R E A T !!!
The answer was use rcf2833 for dtmfmode,
not inband as suggested earlier
If someone can help me resolve the cgi problem, I'd
be forever indebted
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
It is shortly explained in README.variables, But for the general
non-readers .
exten =
1112,1,Dial(IAX/[EMAIL PROTECTED]|||http://localhost/bcs/callerid.php?phone=${CALLERIDNUM})
This pops an url to the IAX clients, that queries our customer database
for the client info .
There is also
Jason A. Pattie wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Ariel Batista wrote:
| Ok I have a question. I have Xten-lite working with our Asterisk
system and I am able to make and get calls. But the main problem is the
sound is very choppy and sometimes it cuts off words. I have
Hello all,
can someoen advise what is the exact syntaxt format for the latest OH323 in
extensions.conf.
we had error when use the chan_oh323. It seems it is a syntaxt error. But we
cannot figure out.
Please advise if you could.
Thanks,
Senad Jordanovic wrote:
Scenario one:
One asterisk server, 200+ calls/channels through it. Judging by related
posts this scenario will work fine.
Scenario two:
1+ calls/channels with one registration URL. I heard that Voyage has
50,000+ clients now. I am talking about that sort of scenario.
I had this problem, I believe I fixed it by upgrading openh323, it couldn't
parse string for some reason. Unforunately, time has eroded my memory of
exact solution/reason.
- Original Message -
From: G Lin [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, October 31, 2003 9:54 AM
Is anyone using Asterisk as the gatekeeper/proxy for videophone calls?
Thanks,
--Ernest
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
On Thu, Oct 30, 2003 at 07:29:17PM +, Gavin Hamill wrote:
On Thu, Oct 30, 2003 at 03:10:09PM -, Linus Surguy wrote:
One option you might not have considered is connect your existing PBX
to the back of Asterisk and thereby use it as a channel bank itself.
Very interesting :)
Hi,
Here is a general question, not applying to asterisk so much, but in
the application of asterisk. I have purchased a few IAX DID's through
VoicePulse and am interested in a service provider who has the ability
to provide me with one number (reliable, as I wish to publish), and the
At 10/30/03 12:21 PM, Jared Smith [EMAIL PROTECTED] wrote:
It's my understand that they are db levels. (And, if I remember my
electrical engineering classes from college, a 3db increase effectively
doubles the volume.)
As a slight aside on the subject of gain
It seems that most people
OK, I've breifly looked at STUN and what it is and can do.
First off it is NOT a way to punch UDP through a firewall.
STUN offers a method to determine the firewall environment
and find out just what is out there. But leaves it to
Asterisk to determine what to do.
The way it could be used within
I've managed to gather that the cgi problem as
appears in the httpd error_log is that it can't do setuid.
I've searched the web for the last couple of
hours and tried almost everything I could find, and I still can't get suexec to
work.
Can anyone help, please?
I know this
Hi,
'g' -- goes on in context if the destination channel hangs up
I need the completely opposite of this, something like goes on
in context if the calling party hangs up.
The situation is as follows, i got a call from outside which is
Dial'ed to somewhere else. If the calling party drops the
John Todd wrote:
I've done some reviewing of the archives for G729 and H323
experiences. The landscape of that query isn't pretty - lots of pleas
for help, and nor do I see too many answers. I have a pending bid
that requires some data before I can implement * on this particular
solution.
Chris,
snip
OK, I've breifly looked at STUN and what it is and can do.
First off it is NOT a way to punch UDP through a firewall.
snip
Bottom line: STUN could save the user much configuration
hassel but does noting that a very knowagable person could
not figure out and then put into a
Quoting Jeremy McNamara [EMAIL PROTECTED]:
Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be
terminated on Asterisk systems and sent out Zap interfaces?
IMHO as for today No,
For incomig I couldnt even get it working with g711 and ciscos 72xx and as5300.
Calls were dropped from
why start this with redhat? I'd say it's the worse
linux dist to attempt to make a small footprint. Try gentoo. If you
wantasterisk with knoppix, then start with that or debian (of which it's
based)
- Original Message -
From:
JR
Richardson
To: [EMAIL PROTECTED]
Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be
terminated on Asterisk systems and sent out Zap interfaces?
IMHO as for today No,
For incomig I couldnt even get it working with g711 and ciscos 72xx and
as5300.
Calls were dropped from cisco side after two udp packets from
JR Richardson wrote:
Im trying to get the total Linux/* installation size as small as
possible. Im wondering if anyone has looked at the installed packages
list from the Redhat installation [rpm qa] and has parsed out all
packages not needed for * to run. I follow the custom install guide
Hello,
Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be
terminated on Asterisk systems and sent out Zap interfaces?
A while ago, I only manage to get g729 call works when terminating in Cisco
AS5300 from Asterisk but was unable to terminate call in Asterisk from Cisco
AS53000
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