I am authenticating sipura device as a sip user to my asterisk server.
Things work fine and then suddenly asterisk console tells me:
Oct 26 23:09:17 WARNING[5096]: chan_sip.c:4826 check_auth: Stale nonce
received from 'Sipura1PSTN sip:[EMAIL PROTECTED]'
as soon as that happens if i try to call
hi,
is there anyway to just enable faxdetection in voicemail?
thanks,
paradise dove
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Hi,
I'm not really sure if this helps you, but as far as I remember, the
diastring with chan_capi-cm-0.6 is not
CAPI/g1/0299546476:b${EXTEN},30,r but
CAPI/ggroup/destination[/params] or in your case
CAPI/g1/${EXTEN}/b,30,r.
To set your CallerPresentation, use the SetCallerPres() in your
Hello,
can somebody recommend me any hard or may be even softphones which support
ADSI. I would like to work with Asterisk voicemail application using ADSI.
Thanks,
Dmitry
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Any problems with bristuff ?
2005/10/26, Julian J. M. [EMAIL PROTECTED]:
You can try this patch
(www.maxosystem.net/asterisk/asterisk-stable-polarity.html), if your
telco sends your polarity reversals on answer and hangup.
Julian J. M.
On 10/26/05, Giovanni Miano [EMAIL PROTECTED] wrote:
Hi,
Isit possible to
make the skinny working over a odbc/mysql/oracle db?
what i have to put
in the extconfig.conf and how must the tables look like?
Hope somebody can
help me..
thx
rene
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No... It applies without problems (just a little offset)
Julian.
On 10/27/05, Giovanni Miano [EMAIL PROTECTED] wrote:
Any problems with bristuff ?
2005/10/26, Julian J. M. [EMAIL PROTECTED]:
You can try this patch
(www.maxosystem.net/asterisk/asterisk-stable-polarity.html), if your
I don't know if this will relate to your specific
issue, but I had problems with system not responding
to numbers I pressed right away when dialing
internally-i.e. the dialtone did not stop like it
should when system reads numbers pressed (DTMF).
I found that adjusting the rxgain and txgain in
Hi Angus,
I have the same problem but on a Debian distro I do not know very well...
When I boot the machine only wcfxs and zaptel modules are loadedhow
can I load qozap before wcfxs?
TIA
Giorgio
Angus Comber wrote:
Hello
I am sure this is a very basic Linux question.
But every time
On Thu, Oct 27, 2005 at 10:42:18AM +0200, gincantalupo wrote:
Hi Angus,
I have the same problem but on a Debian distro I do not know very well...
When I boot the machine only wcfxs and zaptel modules are loadedhow
can I load qozap before wcfxs?
echo qozap /etc/modules
I figure that
One last check...won't ask again, promise
:)
Does someone know a solution to my problem
below?
Best Regards
Goran
- Original Message -
From:
Goran
Tornqvist
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 26, 2005 10:33
AM
Subject: Asterisk
I would suggest that you are missing something like:
iptables -A INPUT -m state --state ESTABLISHED,RELATED -i eth0 -j ACCEPT
This will mean that if a UDP packet is sent by * from sport:2345,
dport:5060, then the response (sport:5060, dport:2345) will be allowed
in, whereas at present that is
Now that Skype and Ebay are one, I feel that they will
be cherry-picking all the promising open-source
voip/asterisk development and calling it their own.
There is a company called gNumber that relies
completely on Asterisk that has also teamed up with
ebay for cell phone notification of ending
On Thu, October 27, 2005 12:10, Neil Skowronek said:
Now that Skype and Ebay are one, I feel that they will
be cherry-picking all the promising open-source
voip/asterisk development and calling it their own.
There is a company called gNumber that relies
completely on Asterisk that has also
Hi,
I have Asterisk between PBX and telco line. PBX is reporting number in
overlap dial manner.
I'd like to early connect to telco line as soon as I get for instance two
numbers, that distinguish telco calls. But the problem is if I receive 3
numbers at once, then two numbers dialplan rule
Hi ALL;
I have users with Sipura/Linksysphones
regsitered behind Nat( useing STUNat phonenot
portforwarding) in my Asterisk box, when I try to call them
with another phone i got:
Got SIP response 404 "Not Found" back from
217.6.190.4
SIP/217.6.190.4:5060-666d is
circuit-busy
Isabove
Kerry Garrison wrote:
During a PSTN call the status screen correctly displays the caller ID
information.
Well, if the SPA-3000 is picking up the CID, and PSTN CID as VOIP CID is
set, and the caller ID isn't being passed to Asterisk, it looks as if
the SIP INVITE is being passed to Asterisk
I don't think the problem is NAT-related. Looks like To header in SIP INVITE message do not match to User ID in sipura settings.
On Thu, 2005-10-27 at 01:57 +0330, mohammad mirzaee wrote:
Hi ALL;
I have users with Sipura/Linksysphones regsitered behind Nat( useing
I've had an absoloutely fantastic run with the new KB1 patch currently
on mantis - http://bugs.digium.com/view.php?id=5520
The Digium guys are looking for feedback, please apply and test - If we
can get some positive feedback, it might make it into 1.2!
--Rob
Hello all.
I have installed the Asterisk 1.0.9. But I am facing problems compiling the
zaptel for asterisk. I am getting lots of errors stating dereferencing pointer
to incomplete type.
The error appears in the zaptel.c file. Could anybody please let me know if
they have come across
Chris HARIGA wrote:
Gary Reuter wrote:
On 10/26/05, *Chris HARIGA* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I have a show parked calls php script for my Polycom IP600
phones. If
U are interested let know and I can email it.
Even if Sean doesn't want it, I do! All
Hi list, i'm having a problem with asterisk+pstn termination, i just
bought a TDM400p and connect my phone line(bellsouth) now when im
using the pstn through asterisk there's an echo, i don't know if this
is already have been resolved. If it does please point me to the
instruction
Is it possible to somehow read spandsp / txfax exit codes?
What I mean, I never know if the fax sent through the Asterisk box was
sent successfully, or not (i.e., a real person picked up the phone
instead of a fax machine).
A possibility of reading an exit code, or a log file would allow to
I'm looking for that one too. I had not been succesfull up to now.
Bob.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomasz
Chmielewski
Sent: Thursday, October 27, 2005 1:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
On Wed, 26 Oct 2005, Dean Collins wrote:
Thought this may be of interest to some people on this list.
https://studio.tellme.com/skype/submissionprocess.html
Bullet point 4 translates for me into If you live in South Africa or
another country where Paypal won't take customers, go away now
Hi
I
have a pri connection working on asterisk; I would like to send the MWI on the
PRI link
Libpri
code clearly says that it is there, but there is no document in asterisk says
anything about this.
The
current mailbox config also doesnt work
Anyone
has any idea about this?
That shouldnt be complicate, but it
looks like you re not registering with your provider. However, without
the configuration files, it is not much to do for help you.
Carlos Alperin
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pikoro
Sent: Wednesday, October
Faris Raouf wrote:
Chris HARIGA wrote:
Gary Reuter wrote:
On 10/26/05, *Chris HARIGA* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I have a show parked calls php script for my Polycom IP600
phones. If
U are interested let know and I can email it.
Even if Sean
On Wed, Oct 26, 2005 at 03:48:19PM -0500, John HIll wrote:
I have a small test system -- 6 phones. It is a dual processor server. I
noticed that asterisk spawns 12 child processes. Can this be controlled? I
would think 2-4 would be plenty for this test site.
Asterisk generally spans a
I've been given the charge of finding out if anyone has gotten vmail.cgi to
work with asterisk realtime, pulling the voicemail users from the db...
I thank you all for any input you may have
Sherwood
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On Thu, Oct 27, 2005 at 04:43:21PM +0530, Bharat M. Sarvan wrote:
Hello all.
I have installed the Asterisk 1.0.9. But I am facing problems
compiling the zaptel for asterisk. I am getting lots of errors stating
dereferencing pointer to incomplete type.
I have a number of such
Hi Tony
Thanks for the reply and for
posting the code. I added the code and recompiled Asterisk, but unfortunately
it did not resolve the issue. It basically trapped all of the incoming audio
and wrote to the error log instead of outputting it. So basically it never
seemed to go to the
Hi there,
I have 2 ISDN modems (HFC chipset). I use bristuff
from junghanns. Its
possible to load one cards as NT (T-Bus) and the other as S-Bus.
When I do make load the 2 cards loads
as S-Bus and when I do make loadNT the 2 cards loads as T-Bus.
Can someone help me??
Thanks in
Hi,
we have strange problem on our new card. Sometimes it reports all channels
busy, so call cannot be made (it doesn't even appear in log).
We've contacted Digium support, but received no usable answer (they've told
us that this card should work on stable Asterisk version - AFAIK this is
http://www.voip-info.org/wiki-Asterisk+zaphfc
look this
Giordano
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] Per conto di Pedro Nunes
Inviato: giovedì 27 ottobre 2005
16.23
A: asterisk-users@lists.digium.com
Oggetto: [Asterisk-Users] Bristuff
question
Hi
Here is my version
Software Version: 3.1.5(GWb) Hardware Version: 2.0.1(42a8)
I had mentioned this before, I am downloading 3.1.7 right now.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Daragon
Sent: Thursday, October 27, 2005 3:50 AM
Senad,
We welcome competition of any kind.It just makes us improve and aim higher.
Ans: Before aiming higher, why dont you guysjust deliver a working software for your current clients, who paid the money and never got anything in return.?
Good luck to A2Billing in its pursue although our
[EMAIL PROTECTED] wrote on 10/27/2005
08:22:04 AM:
If you do an fxotune and all of the coefficients are 0, does
this
mean that fxotune is not
making
any changes?
Based on what Matt has mentioned previously, fxotune only sets the
impedence
to proper values today. He has not
Look on the wiki which is located at:
http://www.voip-info.org/
On 10/27/05, Dmytro Mishchenko [EMAIL PROTECTED] wrote:
Hello,
can somebody recommend me any hard or may be even softphones which support
ADSI. I would like to work with Asterisk voicemail application using ADSI.
Thanks,
Dmitry
At 08:38 AM 10/27/2005, you wrote:
http://www.voip-info.org/wiki/view/Polycom+XML+browser+scripts+for+asterisk
Best regards,
Chris HARIGA
Thanks.
Is it possible for someone to provide a basic explanation of how to
implement this for us less technical minded people?
From what I can tell, it
Upgraded to 3.1.7
Excerpts from Asterisk Log:
Oct 27 07:43:50 DEBUG[1531]: cdr_mysql: SQL command as follows: INSERT INTO
cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration
,billsec,disposition,amaflags,accountcode) VALUES ('2005-10-27
07:43:50','\Garrison Kerry\
Are you having a problem? Have you even tried to do it?
We are using asterisk realtime with MySQL voicemail integration.
vmail.cgi works just fine. I think I had to tweak a variable in it to
tell it to look in the database instead of a file. Open the CGI up
and take a look at it.
On 10/27/05,
Kevin Bockman wrote:
This is just a feature of PRI service. Of course all of the call info
is still available even if you 'block' it. The call still has to be
traceable. Magic huh? I thought that was cool too the first time I
found out about it.
It depends on whether you are purchasing
Giordano,
Thanks, stupid question.
Ive look to that page 100 of times but I do not remember that part of
the page about loading more than one card :S.
Thanks again
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Giordano Grandis
Sent: quinta-feira, 27 de
Hi guys. Please disregard this. I'm testing connectivity after being
down due to Hurricane Wilma.
Thanks,
Waldo
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Try Voice Over Ethernet. Asterisk cannot do that since it only
supports Voice Over IP.
On 10/25/05, Maps [EMAIL PROTECTED] wrote:
Dear Supporters!
Does any one know how to set the asterisk to allow the phone to register to
asterisk using the MAC address?
Thanks!
Lan Phan.
I had to turn on the aggressive echo cancellation in the zaptel drivers
for mine. Which is much better, but we still get occasional pops.
The funny part is only the asterisk side of the connection hears the
echo.
Jared Armstrong
-Original Message-
From: Rich Adamson [mailto:[EMAIL
Steven Langley wrote:
Hi Tony
Thanks for the reply and for posting the code. I added the code and
recompiled Asterisk, but unfortunately it did not resolve the issue. It
basically trapped all of the incoming audio and wrote to the error log
instead of outputting it. So basically it never
Seshu,
So, now you are not Seshu Kanuri any more but Pbxware Swithware?
Since you are not working or associated with our company I need to ask you
not to use Pbxware, Switchware in your email client From field.
PBXware SWITCHware wrote:
Senad,
We welcome competition of any kind. It just
Kerry Garrison wrote:
Upgraded to 3.1.7
Excerpts from Asterisk Log:
Oct 27 07:43:50 DEBUG[1531]: cdr_mysql: SQL command as follows: INSERT INTO
snip ...
Here is a link to a screenshot of the SPA3000 settings:
http://techdatapros.com/temp/spa3000.gif
I get connection refused at that URL.
I have the first edition,
does anyone know if it's worth getting the second too?
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Tzafrir,
Thanks for the reply.
This is a 2.6.13 kernel. Runs very well.
It really is not hurting anything memory usage is ok and it is responsive.
Just my old school resource attitude.
Shana Tova
--john
--
This mail was scanned by AntiVir Milter.
This product is licensed for non-commercial
StevenThere are issues being looked at, see: http://bugs.digium.com/view.php?id=3599http://bugs.digium.com/view.php?id=4252
Always worth while checking through bugs.digium.comRegardsRobOn 10/27/05, Eric ManxPower Wieling
[EMAIL PROTECTED] wrote:Steven Langley wrote:
Hi Tony Thanks for the reply
I've had a very similar thing on my SPA-3000 and they only way to fix
it was a full default reset on the SPA and reconfigure it from scratch
8-(
Matt.
On 27/10/05, Kerry Garrison [EMAIL PROTECTED] wrote:
Upgraded to 3.1.7
Excerpts from Asterisk Log:
Oct 27 07:43:50 DEBUG[1531]: cdr_mysql:
I sure like my Aastra 390, the voicemail ADSI app works pretty well (only a
couple of incompleted functions, like not exiting by hanging up the
speakerphone, rather than go to a reorder tone.
As for the 'look at the wiki' comment, I'm not trying to get on anyone's
badside, but Dmitry was
Take this silly argument off-line please.
On Sun, 27 Nov 2005 16:28:27 -, Senad Jordanovic wrote: Seshu, So, now you are not "Seshu Kanuri" any more but "Pbxware Swithware"? Since you are not working or associated with our company I need to ask you not to use "Pbxware, Switchware" in your
I just tested it from a different location without any problem.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Daragon
Sent: Thursday, October 27, 2005 8:44 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re:
I'm using an IAXy witha current CVS-head build of Asterisk.
The IAXy has an extensions.conf entry somethng like this:
exten = 1,1,Ringing
exten = 1,2,Answer
exten = 1,3,Voicemail(u1)
exten = 1,4 Hangup
This works fine for calls routed to extension 1. But if a second call is
routed to the IAXy
[EMAIL PROTECTED] wrote:
At 08:38 AM 10/27/2005, you wrote:
http://www.voip-info.org/wiki/view/Polycom+XML+browser+scripts+for+asterisk
Best regards,
Chris HARIGA
Thanks.
Is it possible for someone to provide a basic explanation of how to
implement this for us less technical minded
George Gardiner wrote:
Take this silly argument off-line please.
Yap.. you are right.. it should not be here... apologies!
Senad
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RELLLY???
Hell, I can do that. Anything is worth a try at this point. I have it fully
documented so restoring the settings shouldn't take but a few minutes. I am
just not going to be in the office for about 5 hours now and not going to
ask my wife to do it. I will certainly try it, its had
[EMAIL PROTECTED] wrote on 10/26/2005 06:53:51 PM:
I have two TE110P cards.
If I stop the Zaptel service, the whole server hangs.
I have had this issue with 1.0.7, 1.0.8 ,1.0.9 and 1.0.9.2.
The server is a Dell 1750 with all unnecessary BIOS options off (USB,
Serial, Second NIC, etc)
It
Hi
Thanks for the reply
I do actually use the |q
option to disable the enter/exit sounds.
Steven
Message: 15
Date: Thu, 27 Oct 2005
10:25:32 -0500
From: Eric
\ManxPower\ Wieling [EMAIL PROTECTED]
Subject: Re:
[Asterisk-Users] Re: delays with IAX2 and Meetme
To:
I
currently have the following network configuration:
Internet--Firewall
--- DMZ
--- Company A
--- Company B
--- Company C
Each
company has its own network address
I
want to install asterisk and use SIP hardware phones that will be located in
all the
Hello Phil ,
On Thu, 27 Oct 2005, Phil Pritchard wrote:
only new to asterisk, but have had some hardware exp.
stay away from irq9 its tied to irq2 and will always be shared, Paul has
the go.. in bios disable serial and or usb (if not using) and make sure irda
is not enabled.
I'll give it a shot.
Do you compile it with Zaptel running or diasable it and reboot first?
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -- - - -- - - - --- - -- -
- --- - - --
In article [EMAIL PROTECTED],
Steven Langley [EMAIL PROTECTED] wrote:
Hi Tony
Thanks for the reply and for posting the code. I added the code and
recompiled Asterisk, but unfortunately it did not resolve the issue. It
basically trapped all of the incoming audio and wrote to the error log
Just added a TDM400p
with 2 fxs ports to asterisk so that I could hook up our fax lines, ztcfg shows
the card being detected and configured correctly (fxo_ks signalling)
Zapata.conf
I would like to be able to monitor my QoS.. I see that Qwest is using this
QoS Manager (Firehunter)
http://www.home.agilent.com/cgi-bin/pub/agilent/Product/cp_Product.jsp?NAV_ID=-536885714.536882909.00LANGUAGE_CODE=engCONTENT_KEY=49888ID=49888COUNTRY_CODE=US
I have some buddies who work at
On Oct 27, 2005, at 12:38 AM, [EMAIL PROTECTED] wrote:
My question is, what is the direction in relation to analog boards and
such?
Right now, it looks like the current fad of the asterisk group is
hardware echo
cancelation. However, there is work that is occurring on the software
echo
On Oct 27, 2005, at 8:04 AM, Mustafa N. Deeb wrote:
I have a pri connection working on asterisk; I would like to send the
MWI on the PRI link
Libpri code clearly says that it is there, but there is no document in
asterisk says anything about this.
The current mailbox config also doesn’t
On Oct 27, 2005, at 12:18 AM, [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote on 10/26/2005 05:09:30
PM:
On Oct 26, 2005, at 3:40 PM, Mark Quitoriano wrote:
Hi list, i'm having a problem with asterisk+pstn termination, i
just
bought a TDM400p and connect my phone
Hi,
I apologize in advance if this is a stupid question, but I have not been able to find an answer by searching the web.
I would like to add an FXO port or two to my Asterisk setup, and I am
wondering if there is any good reason to spend $120 on a TDM01B or $180
on a TDM02B instead of paying
On Oct 27, 2005, at 12:02 AM, Mark Quitoriano wrote:
i tried doing the instruction from voip-info[1] anyway here's my
comment with that instruction.
when i tried doing /usr/src/zaptel/fxotune -i 4 it gives me this
Tuning module 1Failure!
Tuning module
Fxotune doesn't appear to work with the latest TDM boards. I have a
TDM400P rev I card and receive the following when running fxotune :
# ./fxotune -i 4
Tuning module 1
Skipping non-TDM / non-FXO
Failure!
Tuning module 2
Skipping non-TDM / non-FXO
Failure!
I didn't see anything obvious in the
On Oct 27, 2005, at 10:25 AM, Jared Armstrong wrote:
I had to turn on the aggressive echo cancellation in the zaptel drivers
for mine. Which is much better, but we still get occasional pops.
The funny part is only the asterisk side of the connection hears the
echo.
If you have bad echo
FWIW, I've noticed on v3.1.7g that after dialing (via the spa3k), about
one of three attempts will cause the pstn line to drop. Not sure as yet
what the problem is, but the spa3k did not do that before upgrading
firmware.
Here is my version
Software Version:
These appear to be a common problems, but after spending half a day
reading the wiki and list archives I have not gained much useful
knowledge beyond the fact that these are a common problems. I am
hoping for some suggestions or pointers to further info.
I have an ivr in my incoming
If you do an fxotune and all of the coefficients are 0, does this
mean that fxotune is not
making
any changes?
Based on what Matt has mentioned previously, fxotune only sets the impedence
to proper values today. He has not implemented the code to set the
coefficients
as yet,
[EMAIL PROTECTED] has the Zaptel from head. Just need to update the zaptel
drivers from CVS head you don't have to upgrade the asterisk.
Matthew Fredrickson wrote:
On Oct 27, 2005, at 12:02 AM, Mark Quitoriano wrote:
i tried doing the instruction from voip-info[1] anyway here's my
comment
[EMAIL PROTECTED] wrote on 10/27/2005 10:20:05 AM:
I'll give it a shot.
Do you compile it with Zaptel running or diasable it and reboot first?
Either way should work fine... Of course you will hang one more time
trying to unload the current zaptel drivers.
Bill
The recent suggestion on the list was to not use 1.0.9 zaptel
Chris Miller wrote:
Fxotune doesn't appear to work with the latest TDM boards. I have a
TDM400P rev I card and receive the following when running fxotune :
# ./fxotune -i 4
Tuning module 1
Skipping non-TDM / non-FXO
Failure!
Tuning
Is it possible to play or generate some white noise, down an Asterisk call? Some
calls I am making are terminating if there is an RTP timeout.
Is there some file I can play during the call to fix this?
/Obelix
This message was
Anyone know what's causing this:
-- SIP read from x.x.x.x:56800:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKba5c00f818387D67
From: Xxx xxx sip:[EMAIL PROTECTED];tag=69375B3E-ACF6C78B
To: sip:[EMAIL PROTECTED];user=phone;tag=as57402fc2
CSeq: 1 ACK
On 10/27/05, Chris Coulthurst [EMAIL PROTECTED] wrote:
I sure like my Aastra 390, the voicemail ADSI app works pretty well (only a
couple of incompleted functions, like not exiting by hanging up the
speakerphone, rather than go to a reorder tone.
As for the 'look at the wiki' comment, I'm not
First you could adjust the rtp timeout (I think its in sip.conf),
second the white noise (or CNG, Cofort Noise Gen) is something that
was added to the bug tracker not too long ago, although only as an
application right now, but if more ppl test it and report (even just
that it works) it will push
What a creative way to test. GL.
On 10/27/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
Hi guys. Please disregard this. I'm testing connectivity after being
down due to Hurricane Wilma.
Thanks,
Waldo
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Hi,
Is there any possibility to 'point' a FWD, Callware or IPKall number/virtual
number to an Asteriks server bypassing the PSTN network to connect cu
Asteriks? I need to do a setup like this:
1 local VoIP provider for calls within the country. Calls will be made
directly from the endpoint.
I was sitting at my buddies house, and noticed a little sign that he
has on his desk, and thought, these are great words for the Asterisk
community to live by.
Service Policy:
We provide service which is CHEAP, FAST PERFECT.
You can only have two.
If you want it CHEAP and FAST,
It won't be
Folks!
I want to know if anyone in the list is using OpenSER,
which appears to be a fork of SER. If so can you post
Your comments on its functionality?
The location where this is available is here:
http://openser.org/index.php#about
Some of the the features I am impressed with being...
[EMAIL PROTECTED] wrote on 10/27/2005
03:11:11 PM:
Are these settings persistent across reboots? The README
for
fxotune seems to mention that you
need to do a fxotune -s in order to reload the card
with the
analyzed settings (rather than
take the 20 minutes it seems to take on my 6
On Oct 27, 2005, at 3:35 PM, [EMAIL PROTECTED] wrote:
However, my fxotune.conf contains only 0's for all 8 of each of 6
lines. I'm wondering does that mean that fxotune had no effect, or
that whatever effect it does have is A) Persistent within the card
between reboots and B) Not reflected
Hi all,
this is probably a asterisk-devel question but I'll try it here first.
Is there a way to delay a ReInvite on SIP? I have an issue where my
provider's server is only ~1 ms RTT away and for about 1/3 of the
incoming calls I get a 482 Loop Detected error because the ReInvite
is processed by
Are these settings persistent across reboots? The README for
fxotune seems to mention that you
need to do a fxotune -s in order to reload the card with the
analyzed settings (rather than
take the 20 minutes it seems to take on my 6 lines). However, if
fxotune.conf is all 0's, I
Juan Salas wrote:
yes,
I tested too and it's works.
The Problem is when we want to add (or delete)
registers without reload the asterisk.
We are using it like a border server wich
is registered like many users in a SER machine
and the real endpoints are registered in the
asterisk.
I
Luki wrote:
Hi all,
this is probably a asterisk-devel question but I'll try it here first.
Is there a way to delay a ReInvite on SIP? I have an issue where my
provider's server is only ~1 ms RTT away and for about 1/3 of the
incoming calls I get a 482 Loop Detected error because the
The argent office does not support DASS2 so I suspect your circuit will be
ISDN30e anyway.
Neil
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy
Sent: 26 October 2005 15:38
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
We're having a rather serious echo problemusing the Grandstream GXP-2000's with Asterisk 1.0.9. I'm wondering if there is something I'm overlooking that might be an easy fix. The echo seems to be worst on internal SIP to SIP calls but you do get it every once in a while on outgoing calls through
Hi
I´ve just installed a2billing using PHP Version 5.0.4, MySQL version
4.1.12 and Asterisk CVS-v1-0-06/27/05, verified database installation and can see webpage,
login, create cards, etc, but I cant hear anything when I call the extension:
extension.conf
; use 6608600 as access number to enter
I need to detect when a fax machine answeres an outgoing call. NV_FaxDetect
and the zaptel fax detect seem to only work in calls originated FROM a fax
machine, not for calls ANSWERED by a fax.
Thanks
--
Larry Host
NuWorld Telecom, Ltd.
858-334-9355 Cell
tfbunm AOL and Yahoo IM
[EMAIL
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