On Wednesday 12 November 2003 18:56, Adam Hart wrote:
> Prehaps a novel thought but what about ODBC for asterisk? Isn't that
> the whole idea of standards and such, stop adding support for every
> db and just have odbc?
People have been talking about ODBC for Asterisk for more than a year.
Since i
Hi,
Something like
exten => _91NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1},30,tTr)
to allow both the caller and the other part to transfer
Take care that the timeout parameter is first, then the 't"
BR,
Dan
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
S
[EMAIL PROTECTED] said:
> I'm creating a test environment for Asterisk. I have Asterisk running
> on a
> PC with only a NIC card, No FXO, FXS, TDM cards. I have two Cisco
> 7960
> phones setup for SIP. Within Asterisk, the SIP SHOW PEERS, shows the
> phones. They don't appear under SIP SHOW REG
reddy wrote:
Hi,
i have configured asterisk to SIP soft phones.
Iam confused about the telephony interfaces like T1,E1,PRI interfaces and how to cnfigure them to asterisk.
If iam using a dialogic card in the asterisk server .Can i use this card with a telephone line to dialout to different n
Hi,
i have configured asterisk to SIP soft phones.
Iam confused about the telephony interfaces like T1,E1,PRI interfaces and how to
cnfigure them to asterisk.
If iam using a dialogic card in the asterisk server .Can i use this card with a
telephone line to dialout to different numebr in the
I have a Sipura SPA2000 2 line SIP FXS box with line 1 on port 5060 and line 2
on 5061. The SPA2000 is on IP address 192.168.17.6, and the asterisk box is
on 102.168.17.2. Both SPA2000 ports(5060 and
Well in pine I'm sending it out as [EMAIL PROTECTED] In asterisk in the
actual voicemail.conf file I set the From field to a valid user name like
[EMAIL PROTECTED] However for the loopback I have several names like local
host.localdomain and myhost.mydomain.com which actually is probably
unres
Use like this...
[mainmenu]
exten => s,1,Goto(sales|100|1)
exten => s,2,Goto(support|200|1)
[sales]
exten => 100,1,Answer ; Answer the line
exten => 100,2,DigitTimeout,5 ; Maximum Timeout between
digits
exten => 100,3,ResponseTimeout,10 ; Maximum Timeout aw
If you send out via pine.. Who are you sending the mail out as... Also
if * sends the mail out who is it sending it out as?
Example if you host file only has loopback with localhost then it might
be sent out as [EMAIL PROTECTED] And if Yahoo can resolve
that domain it wont accept the email..(help
Does anyone know where I can get a list of ADSI functions.. Example *70
(No Call Waiting), Flash = Flash, Hold = ???
Thank you,
-gcc
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So what do you use instead of s,1? My s extensions set things like
response timeout, digit timeout, etc. Thanks again.
AJ
On Thu, 13 Nov 2003, Master Abi wrote:
> I had experienced this problem before. I found this to be related to 2
> items. Firstly, try not to use the s,1 starting each subm
I had experienced this problem before. I found this to be related to 2
items. Firstly, try not to use the s,1 starting each submenu. Secondly,
if there are more than 20 sub menus, you will get this delay problem.
Why I do not know. I reordered and regrouped and the problem
disappeared.
-Origin
Ok thanks. I'll try to shorten the digit timeout.
On Wed, 12 Nov 2003, David Carr wrote:
> Without looking at your extensions.conf I can only guess that maybe the
> first digit(s) of your exten aren't unique and asterisk is waiting for a
> digit timeout. You can shorten your timeout or make your
Without looking at your extensions.conf I can only guess that maybe the
first digit(s) of your exten aren't unique and asterisk is waiting for a
digit timeout. You can shorten your timeout or make your extensions unique.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTE
Hi again,
I'm attempting to figure out how to transfer calls from an IAX client. I
have read and seen on the list where if you put a ,t at the end of the
dial portion in the extensions.conf file that you should be able to use
the # to park and transfer calls. I have not found this to be the cas
Our forums are opened for anykind of discussions, including business
opportunities, termination providers list, etc.
Let's move some stuff to forums, so people who is interested can find and
exchange information.
Please join, at least we will empty this mailing list from some busienss
discussions,
Hi guys
I've set up a layered menu system on one of my asterisk servers where
there is a main menu and several submenus; one for each department. Each
menu plays a background intro message giving its various options. My
problem is when I'm in the main menu and press the option to go to one of
I have ZAP channels.. so i add the lines at zapata.conf
and it does not work. When i dial *8 it return me a busy tone.
my zapta.conf is..context=home
group=2
pickupgroup=2
signalling=fxo_ks
channel=2-3
callerid="FIJO" <200>
channel=3
callerid="INALAMBRICO" <100>
channel=2
Rich Adamson wrote:
Is it
Thanks David, I had a an excellent laugh at this (actually my first of
the day, which makes it worth twice as much) ;->
Philipp
> [peons]
> exten => 1,1,TakeOutTheTrash()
> exten => 2,1,WashTheDishes()
> exten => 3,1,CleanTheToilet()
>
> [rulers]
> include => peons
> exten => 800,1,FeedMeGrapes
On my asterisk server I have placed valid email addresses in the
voicemail.conf file as to allow mailbox users to receive message
notification. My problem is it appears that the messages are attempting to
be sent but instead they are bouncing with a fatal error message like the
one below:
(re
Thu, 13 Nov 2003, Fresno CA, Adam Hart, spoke these words:
> Prehaps a novel thought but what about ODBC for asterisk? Isn't that the
> whole idea of standards and such, stop adding support for every db and just
> have odbc?
>
> ___
> Asterisk-Users ma
Returning to the original question of this thread: Have you ever looked
at an LCR implementation instead of building your own Db? I know that i4l
is not so popular around here, but still this might be of interest:
http://www.isdn4linux.de/faq/i4lfaq-3.html#ss3.26
3.26 feature_lcr: Can isdn4linu
Hi!
> Is anyone familiar with http://www.zultys.com/index.htm. Do they use
> Asterisk?
I guess not - but I found something rather interesting on their site:
"Does the ZIP 4x4 support voice encryption?
Yes. The ZIP 4x4 can use Secure RTP and AES encryption to transport voice
traffic in a secu
http://bugs.digium.com/bug_view_page.php?bug_id=504
I have been testing this patch today. Works great. Just wondered if
anyone else was intrested in such a beast.
bkw
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I don't believe that this is particularly relevant
to the Asterisk software -- perhaps another list
can be created for discussion of what commercial
services may or may not exist?
On Thu, Nov 13, 2003 at 08:02:12AM +, Isamar Maia wrote:
>
> I am looking for a brazilian VOIP terminator for
>
Thank you Gus
I found some mistake in extension.conf
exten => 1001,2.Voicemail(u1001)
This must be change to ",2,Voicemail(".
Now I use some hard phone, so I would better try another codec.
Thanks for your good advice
>Try with another codec different than G.723. Use GSM o G.711 for th
Prehaps a novel thought but what about ODBC for asterisk? Isn't that the
whole idea of standards and such, stop adding support for every db and just
have odbc?
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--- PJ Welsh <[EMAIL PROTECTED]> wrote:
> On Wed, Nov 12, 2003 at 09:22:56AM -0800, Chris Albertson wrote:
> ...
> > There are several other very good free SQL DBMSes. One of them
> > is actually supported by one of the world's largest software
> > companies, SAP.
> >
> > SAP and MySQL signed
any other gentoo users out there?
I installed from cvs earlier today, and then figured, hell I might as
well have some portage scripts to do it for me.
so, a set of cvs ebuilds for zaptel, zapata, libpri and asterisk:
http://bugs.gentoo.org/show_bug.cgi?id=33345
share and enjoy
++dg
__
On Wed, Nov 12, 2003 at 09:22:56AM -0800, Chris Albertson wrote:
...
> There are several other very good free SQL DBMSes. One of them
> is actually supported by one of the world's largest software
> companies, SAP.
>
> SAP and MySQL signed an agreement where MySQL will co-market SAPDB
> and the
On Wednesday, November 12, 2003 3:47 PM, Tom Weeks
[SMTP:[EMAIL PROTECTED] wrote:
> Good day,
>
> I am trying to setup an outbound dial plan which will time out if no
> answer. Using a X100P with the following dial command :
>
>
> exten => 101,3,Dial(Zap/1/3036972357,5) ; try the desk line - fail
Did you record the messages as gsm format?
- Original Message -
From: "Larry D. Black" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, November 12, 2003 6:33 PM
Subject: [Asterisk-Users] menu prompts and voice mail greetings.
> What program do you use to record menu prompts
Hi!
I have this test configuration:
Cisco7940(SIP)->*->GnuGK(H.323)->ATA186(H.323)
When I do call from ATA to 7940, everything is OK (exept volume level, but it
is not seriously). But when I try to call from 7940 to ATA, I got a strange
error:
=*= In CreateRealTimeLogicalChannel for c
I am looking for a brazilian VOIP terminator for
the states of Sao Paulo, Parana, Para and Bahia.
Isamar
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By the end of next week, we'll be able to offer IAX2 service for Vancouver,
Toronto, Hamilton, Montreal. End of this month or so: Calgary, Edmonton,
Ottawa, Winnipeg. Sometime in December: Windsor, Kitchener and London.
By mid next week, Charlotte NC should be on line.
Other centers, as listed
Good day,
I am trying to setup an outbound dial plan which
will time out if no answer. Using a X100P with the following dial command
:
exten => 101,3,Dial(Zap/1/3036972357,5) ;
try the desk line - fail to step 104
It dials out successfully, but never times
out. I have a basic Zapa
I prefer to record them right over the phone. Set up a macro like this
[macro-recordsound]
;${ARG1} - Sound filename
exten => s,1,record(${ARG1}:gsm,3)
exten => s,2,playback(${ARG1})
exten => s,3,playback(vm-goodbye)
exten => s,4,hangup
and then in your main context do something like this
;Tempo
Not to incite a flame war, but the only phone I like is the Cisco 7960. If
you look hard enough you can find them off-lease for around $235 with a
power brick. Cisco just came out with some new firmware that adds features
previously sorely missed (like call forwarding). Once you get used to being
a
What program do you use to record menu prompts and voice mail greetings
we tried windows recorder and it kept telling us bad file format.
Thanks.
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Title: "488 not acceptable here" message
I'm creating a test environment for Asterisk. I have Asterisk running on a PC with only a NIC card, No FXO, FXS, TDM cards. I have two Cisco 7960 phones setup for SIP. Within Asterisk, the SIP SHOW PEERS, shows the phones. They don't appear under SI
Sure. Make two contexts like these:
[peons]
exten => 1,1,TakeOutTheTrash()
exten => 2,1,WashTheDishes()
exten => 3,1,CleanTheToilet()
[rulers]
include => peons
exten => 800,1,FeedMeGrapes()
This way the rulers have everything the peons have and then a little more.
Then use your sip.conf or your
Hi All,
Using asterisk and extension.conf can i make a group dial code ?
Like this.
Ie. Let's say i have a group called directors.
Only People in this group can dial to a external number like 800.
How can i make this possible in asterisk ?
Thanks alot !
Carlos.
_
That is just beautiful...
Would you mind if it got into the wiki, or onto a webpage here or there?
-
Andrew Thompson
- Original Message -
From: "Carlton J. O'Riley" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, November 12, 2003 3:48 PM
Subject: RE: [Asterisk-Users]
I have read all the mailing list posts regarding rxfax receiving a fax and
outputing an 8 byte tif file (tif header only). This is the problem I can't
seem to get past. Has anyone out there also had this problem and found some
workaround for it?
___
Aste
Could anybody shed some light in which device they would use in this
situation:
Remote office PBX's to be connected via a) Cisco ATA-186 or b) Sipura
SPA-2000
or c) Grandstream HT-ATA-286 to go via the net to an * box.
Pros / Cons for each device would be appreciated!
Thanks
Kris
__
Here is what I have which uses IAXTEL for 800 calling and VOIP for long
distance with a fall back to my PSTN line. I don't have any issues as far
as 1800 numbers being grabbed before the long distance numbers. My internal
context is for all extensions inside the house, whereas the fax doesn't use
Quoting Freddi Hansen <[EMAIL PROTECTED]>:
> I haven't been able to google any function in '*' that would help us
> with this so that's
> why I try the list in case I (hopefully) have overlooked something.
Just take a look at fax extenstion which basically does what you want.
___
Putting
exten => fax,1,RxFax(filename)
in your context invokes the kind of fax detection you are seeking. You can
similarly send the call to a real fax machine
exten => fax,1,Dial(Zap/1)
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Freddi Hansen
>
On Wed, 12 Nov 2003, Freddi Hansen wrote:
>Hi,
>I am looking at using the softfax that Steve Underwood has developed.
>It's very straight forward when you assign an extension for the fax.
>A function that several pbx's has is that they listen for the 'faxtone'
>for 5 seconds
>after 'answer' in the
Hi there,
I'm new to Asterisk. Installed, configured, but not really used it
yet...
I'm considering some investment on mounting a small network for voice
phones, say 20 to 30 terminals.
What hardware should I use for the telephones ? IP Phones seam too
expensive and I'm sure they do a lot of th
That's an awful lot of assumptions, my friend. What we care about is
how long it takes to get the FIRST result, not all of them together. I
mean, I only need to call ONE number, not 1000...
This comes to a statement of optimal. What is optimal? Optimal with
respect to what??? We want somethin
Actually, you may want to make your TSP use the Manager interface. Not
ALL of the TAPI primitives are supported, but such is the case with most
PBXs. Better yet, you can alter the PBX source to add additional
events/commands that can be written into your TSP.
If you need the TSP/MSP (or TAPI/WAV
Chris Albertson wrote:
The new "OpenOffice" works very well now and is completley
cross platform. It also allows one to save in any of a
serval file formats. I've been using it to produce
HTML, PDF and plain text format copies of documentation.
and I can run this same Open Office suite on Solari
Hi,
I am looking at using the softfax that Steve Underwood has developed.
It's very straight forward when you assign an extension for the fax.
A function that several pbx's has is that they listen for the 'faxtone'
for 5 seconds
after 'answer' in the menu where you can enter your local extension n
On Wed, 12 Nov 2003, Ray Burkholder wrote:
> Are the Digium T1/E1 cards capable of D channel bonding for PRI? As in one
> D channel can service two more PRI lines?
NFAS? Not that I know of.
--
Dave Weis "I believe there are more instances of the abridgment
[EMAIL PROTECTED] of th
Ernest W. Lessenger wrote:
I am having some oddness with the 11/11/2003 CVS of *. Specifically,
outgoing audio to NuFone doesn't seem to be transmitted (I can hear
the other side just fine). My firewall is set to allow all outgoing
traffic, and the IAX2 connection is definitely established corr
Chris Albertson wrote:
Testing a querry by doing 2000 identical querries and then
deviding the total by 2000 is not a valid way to
measure the time to do one querry. The result will appear
to be as much as 100X or even more to fast.
The reason is:
1) Operating system will have cached the exact d
David Gomillion wrote:
Hello. I have never run into this problem. What I would do is inserted
below:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stephen R. Besch
Sent: Wednesday, November 12, 2003 11:06 AM
To: asterisk users list
S
Hi,
Does anyone know of Canadian VoIP termination providers? I have
Canadian customers and would like to provide Canadian dial in and dial
out (canadian callerid).
Thanks!
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Title: D Channel Bonding
Are the Digium T1/E1 cards capable of D channel bonding for PRI? As in one D channel can service two more PRI lines?
Ray Burkholder
[EMAIL PROTECTED]
http://www.oneunified.net
704 576 5101
--
Scanned for viruses & dangerous content at
One Unified
and is bel
Title: Zultys.
Is anyone familiar with http://www.zultys.com/index.htm. Do they use Asterisk?
Ray Burkholder
[EMAIL PROTECTED]
http://www.oneunified.net
704 576 5101
--
Scanned for viruses & dangerous content at
One Unified
and is believed to be clean.
The new "OpenOffice" works very well now and is completley
cross platform. It also allows one to save in any of a
serval file formats. I've been using it to produce
HTML, PDF and plain text format copies of documentation.
and I can run this same Open Office suite on Solaris, Linux
and Windows.
Hi,
Citeren Michael Devenijn <[EMAIL PROTECTED]>:
> Has anyone ever worked opn TAPI stuff to make asterisk work with it ?
>
> I'm a Windoze C++ developer dig'n into asterisk (and linux at the same time)
> since a few months and i'm quite interested in creating a TAPI driver for
> asterisk.
>
Steven Critchfield wrote:
Are you trolling here, or are you just clueless about the people who
will be helping contribute to your documentation? I'm sure I am not the
only one here that goes weeks on end without touching windows. Screw
Word and its largely bloated file formats.
Unfortunately, we
David Gomillion wrote:
Test script of 1000 quieries..
Query1 ("code" field not indexed) = 47.183s
Query1 ("code" field indexed) = 45.731s
Query2 ("code" field not indexed) = 109.321s
Query2 ("code" field indexed) = 2.302s
I disagree with your disagreement :P We have to keep in mind the b
Testing a querry by doing 2000 identical querries and then
deviding the total by 2000 is not a valid way to
measure the time to do one querry. The result will appear
to be as much as 100X or even more to fast.
The reason is:
1) Operating system will have cached the exact disk sectors
required r
I am having some oddness with the 11/11/2003 CVS of *. Specifically,
outgoing audio to NuFone doesn't seem to be transmitted (I can hear the
other side just fine). My firewall is set to allow all outgoing traffic,
and the IAX2 connection is definitely established correctly. Also, I can
watch UD
Thanks everyone for your help on this..
For those who are interested I have done some speed tests on these two
queries (below) on my server and the results are..
Test script of 1000 quieries..
Query1 ("code" field not indexed) = 47.183s
Query1 ("code" field indexed) = 45.731s
Query2 ("code" field
Hello Gavin,
Sorry for so long time in my reply but I was very busy on other tasks.
I attached to this message my working test files for mgcp.
Best regards,
Daniel
Daniel ANDRE wrote:
Gavin Hamill a écrit:
On Tue, 2003-11-04 at 10:14, Daniel ANDRE wrote:
Hullo
Hi David,
For those who are interested I have done some speed tests on these
Test script of 1000 quieries..
Query2 ("code" field indexed) = 2.302s
OUCH! those times are lng!
I agree the first three are long, but the last one works out to just
over 26000 queries per min.. I didn't think
Hello. I have never run into this problem. What I would do is inserted
below:
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Stephen R. Besch
> Sent: Wednesday, November 12, 2003 11:06 AM
> To: asterisk users list
> Subject: [Aster
Hey,
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of WipeOut
> Sent: Wednesday, November 12, 2003 10:28 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] OT : For the SQL gurus..
>
> Andy Powell wrote:
>
> >>Thanks everyone
Codecs are g711ulaw, on both Cisco5300... Dial Peer config is showed
below
Los codecs que uso son G711ulaq, en los dos Cisco5300, te muestro los
dialpeers...
GW that not work - GW que no funciona
translation-rule 1017
Rule 0 8002666333 1000
dial-peer voice 1016 voip
destination-pattern 800266
During calls using an extension off of the
TDM40B out through a X100P I sometimes get a echo or cave sound if you
will. It is random sometimes I have it sometimes not. Sometimes it
starts with the beginning of a call sometimes you can be in the middle of a call
and it starts. It only happ
I have a couple of X100P's in my system and
while on calls they just randomly hang up for no reason.
I have tried messing with the busydetect
and callprogress setting them to yes and no same and still random
hangups. Is there another setting I should be looking at?
My zap config looks li
--- Mark Spencer <[EMAIL PROTECTED]> wrote:
> I'll try to call them tonight.
>
> Mark
>
> On Wed, 12 Nov 2003, costas wrote:
>
> > I guess people are pissed off with them and are looking at the
> alternatives. I think they are charging too much money for it. Also
> they must compete against MS
It looks like my conversion of the STUN server to a GNU Autotools
build system will go into Vovida.org's CVS system soon. My next
task will be to do the same for Asterisk. Third task is to get
Asterisk to use STUN.
Back to BSD: I think GNU Autotools is the right way to fix this.
But until t
I have an interesting dilemma with sequencing in the dialplan. Up to
now, I have assumed that the extensions in the dial plan were tested in
the order that they appear in extensions.conf. In other words, I have
the following fragment which was designed to dial toll free on the PSTN
and all ot
At 11:07 AM 11/10/2003, you wrote:
Thanks everyone for your help on this..
For those who are interested I have done some speed tests on these two
queries (below) on my server and the results are..
Test script of 1000 quieries..
Query1 ("code" field not indexed) = 47.183s
Query1 ("code" field index
Has anyone ever
worked opn TAPI stuff to make asterisk work with it ?
I'm a Windoze C++ developer dig'n into asterisk
(and linux at the same time) since a few months and i'm quite interested in
creating a TAPI driver for asterisk.
so if anybody did any research in that way please
inform me
Roy Sigurd Karlsbakk wrote:
Thanks everyone for your help on this..
For those who are interested I have done some speed tests on these two
queries (below) on my server and the results are..
Test script of 1000 quieries..
Query1 ("code" field not indexed) = 47.183s
Query1 ("code" field indexed) =
Andy Powell wrote:
Thanks everyone for your help on this..
For those who are interested I have done some speed tests on these two
queries (below) on my server and the results are..
Test script of 1000 quieries..
Query1 ("code" field not indexed) = 47.183s
Query1 ("code" field indexed) = 45.731s
Hi Gavin
i have the same error when i try to run DIAX with Wine.
thanks
Dimitri
On Wednesday 12 November 2003 15:23, Gavin Hamill wrote:
> On Wed, 2003-11-12 at 15:07, Dan wrote:
> > DIAX 0.9.3 is available for download from the same place:
>
> Hi Dan :)
>
> Do you know if anyone has succe
On Wednesday 12 November 2003 10:01, Roy Sigurd Karlsbakk wrote:
> > >Thanks everyone for your help on this..
> > >
> > >For those who are interested I have done some speed tests on
> > > these two queries (below) on my server and the results are..
> > >
> > >Test script of 1000 quieries..
> > >Que
Hi,
- Original Message -
From: "Ariel Batista" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, November 12, 2003 5:50 PM
Subject: Re: [Asterisk-Users] DIAX 0.93 with some sound improvements and not
only...
.
> Thank you for the update! I have the following problems wit
Title: Mensaje
Fijate en los 'voice codecs' de los
dial-peers.
- Original Message -
From:
Sebastian Nocetti
To: [EMAIL PROTECTED]
Sent: Wednesday, November 12, 2003 12:41
PM
Subject: [Asterisk-Users] Media
Negotiation Failed
Hi, I have this
scenario
> >Thanks everyone for your help on this..
> >
> >For those who are interested I have done some speed tests on these two
> >queries (below) on my server and the results are..
> >
> >Test script of 1000 quieries..
> >Query1 ("code" field not indexed) = 47.183s
> >Query1 ("code" field indexed) = 45.7
-- Original Message --
From: "Dan" <[EMAIL PROTECTED]>
>Hi all,
>
>DIAX 0.9.3 is available for download from the same place:
>http://www.laser.com/dante
>or
>http://www.geocities.com/tdanro
Thank you for the update! I have the following problems with it! W
Title: Mensaje
Hi, I have this
scenario
Cisco 5300 (public
ip. 200.47.xx.xx) <---> Asterisk (public ip: 64.76.xx.xx) <--> Cisco
3600 (public ip: 64.76.xx.xx , same network than * )
When a calls comes
in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome
message and
Hi,
- Original Message -
From: "Gavin Hamill" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, November 12, 2003 5:23 PM
Subject: Re: [Asterisk-Users] DIAX 0.93 with some sound improvements andnot
only...
> Do you know if anyone has successfully run DIAX on Linux with W
On Wednesday 12 November 2003 09:47, Mark Spencer wrote:
> it's implemented on the zap side (which is now configurable with
> "jitterbuffers=foo" in zapata.conf.
Will this work on a SIP to SIP call?
What does the parameter jitterbuffers=XXX represent? Is it memory allocation
or milliseconds of
On Wed, 2003-11-12 at 15:07, Dan wrote:
> DIAX 0.9.3 is available for download from the same place:
Hi Dan :)
Do you know if anyone has successfully run DIAX on Linux with Wine?
After installing the VB6 runtime DLL, I ran diax.exe and got
fixme:ole:CoRegisterMessageFilter stub
fixme:ole:OLEPic
Romulo,
Without a little more information this is not so
easy to solve. So let me see if I can go through a couple of scenarios and
see if we can figure out your particular problem.
If you have X-Lite behind a NAT router AND you are
not connecting to an * server or your * server is outside
Hi all,
DIAX 0.9.3 is available for download from the same place:
http://www.laser.com/dante
or
http://www.geocities.com/tdanro
The new DLL contain the latest updates made by Steve in the iaxclient
library.
Still just IAX1 is supported (for the moment).
What's new in 0.9.3?
- accept blank p
I'll try to call them tonight.
Mark
On Wed, 12 Nov 2003, costas wrote:
> I guess people are pissed off with them and are looking at the alternatives. I think
> they are charging too much money for it. Also they must compete against MS free
> Personal Server (SQL Server but not optimized) and P
it's implemented on the zap side (which is now configurable with
"jitterbuffers=foo" in zapata.conf.
Mark
On Wed, 12 Nov 2003, Matteo Brancaleoni wrote:
> mmmh... I'm not sure ig chan_sip has jitter buffer.
> I think that there isn't a jb in sip,
> but correct me if I'm wrong.
>
> Matteo.
>
> Il
Hi iax doesn't use zaptel for timing.
only iax2 uses it, but when using trunking=yes.
(not your case,so)
so the distortion could be caused by loss of packets.
Matteo.
Il mer, 2003-11-12 alle 15:31, nathan ha scritto:
> Hi All,
>
> I'm currently running Asterisk with SIP phones and an ISDN card
Hi All,
I'm currently running Asterisk with SIP phones and an ISDN card using
chan_capi. I've just started to use IAX (GSM codec)over the Internet and
the sound is adequate. However, there is an occasional 'glitch' in the
audio resulting in lost sound or distortion. Is the distortion because
I'm u
Try with another codec different than G.723. Use GSM o G.711 for this.
You could disable G.723 in your sip.conf
disallow=all
allow=gsm
allow=alaw
allow=ulaw
Hope this helps,
Gus
- Original Message -
From: "Hachy" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, November 12,
A phone system providing this kind of directory service usually asks for the
first three letters of the person's last name.
So, if you wanted to call me and didn't know my extension, you'd press 6 for
directory. Then you'll press 746 (for S H O - the beginning of my last name)
And the software wil
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