On Wed, 24 Aug 2005, Humberto Aicardi wrote:
Hi,
I've a Fritz card which was working fine, recently I changed hardware and
my nightmare started. Now when I call someone through the chan_capi (0.3.5 or
0.4.0) it works fine but when I receive calls I always get hungup. Can someone
please
On Thu, 25 Aug 2005, Goran Dj. wrote:
But, now I cannot start chan_capi.so:
WARNING[802]: chan_capi.c:3106 load_module: CAPI not installed, CAPI
disabled!
from tty:
capiinit
ERROR: cannot open /dev/capi20 nor /dev/isdn/capi20 - No such file or
directory (2)
capiinfo
capi
When did you install it? Try running the update database function
from the configure menu.
Darren Wiebe
[EMAIL PROTECTED]
Il Neofita wrote:
My installation of ASTCC does not update the cdrs tables .
It is a problem of ASTCC or it is a configuration problem?
You did not specify anything about your network. If your network has a
big latency, echo cancellers can get into trouble. For instance, I
have echo problems just using wireless POTS phones on my sipura 2100
sip adapter/router on an otherwise unused 8Mbps ADSL internet
connection at home.
Lars
Steve Gladden wrote:
You also want to look at the registertimeout and registerattempts
Yes!!!, thank you VERY much this is what I needed.
Where are these options documented at?
I'm guessing the source code?
Or is there a better place to find this stuff?
A search on the wiki for
Matt wrote:
Am I correct in thinking that at this time the CVS-HEAD supports
Jitter Buffer for SIP on Asterisk?
No, you are incorrect.
/o
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Hi,
Just wondering if people could suggest a good VoIP provider that can
service the San Francisco Bay Area and the Los Angeles area. I've tried
race.com (recommended to me) but they're kind of hard to get ahold of.
Any other suggestions? This is for a business, so reliability is key.
I did
Try broadbandreports.com and/or whichvoip.com...
I know of one company that can offer service, I work for 'em...ViaTalk
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of jennyw
-Sent: Thursday, August 25, 2005 3:17 AM
-To: Asterisk Users Mailing List
On 8/25/05, Humberto Aicardi [EMAIL PROTECTED] wrote:
I've a Fritz card which was working fine, recently I changed
AVM Fritz Passive card? If so, then it doesn't work very well in
Pointto Point. Ask your Telco for a Point to Multipoint and change the
setting in capi.conf./
Note:
I am not
I found that only the kernel is installed. I'd avoid
2.6.12 for now as I had problem with the zaptel driver and stay with
2.6.9.
Regards
Lee
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
EstepSent: 24 August 2005 22:33To:
asterisk-users@lists.digium.comSubject:
Hi, do you have an on-site NTP server? I found that
after the firmware update NTP from the * server stopped
working.
Regards
Lee
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jesus
MogollonSent: 24 August 2005 22:11To:
asterisk-users@lists.digium.comSubject:
I am connecting my TE110P card to a National
Microsystems card...
Can it be done?
--- dbruce [EMAIL PROTECTED] wrote:
First off... go through your zapata.conf and
zaptel.conf files and actually
set your configuration for your specific hardware
and desired results.
The obvious is that
Hi all,
Our Asterisk box sends calls outbound via either SIP (through our VoIP
provider) or an E1 PRI (directly connected via a TE410P). When we dial
a number that is engaged via our VoIP provider we get the following on
the Asterisk console (numbers and IP addresses changed to protect the
Anybody using voicetronix cards? The 12 ports for example? What has been
your experience and how many cards can be put into one server?
Do they have the same IRQ problems as Digium ones?
AK
___
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Matt wrote:
Am I correct in thinking that at this time the CVS-HEAD supports
Jitter Buffer for SIP on Asterisk?
No, but attached to issue 3854 you will find patches you may be able
to apply to the current CVS-Head to acheive this.
Regards,
Richard
Hi All
I want to terminate as much POTS lines as possible to
my Asterisk Server, please advice me which Card to
choose with accessories
Thanks
Start your day with Yahoo! - make it your home page
http://www.yahoo.com/r/hs
Thank you was that the problem.
On 8/25/05, Darren Wiebe [EMAIL PROTECTED] wrote:
When did you install it? Try running the update database function
from the configure menu.
Darren Wiebe
[EMAIL PROTECTED]
Il Neofita wrote:
My installation of ASTCC does not update the cdrs tables .
It
Yep, I am connecting to some other equipemnt...its a
Clarent gateway equipped with a National Microsystems
(Quad Port)
--- El Flynn [EMAIL PROTECTED] wrote:
Hi there,
Are you getting the E1 span in from Telekom, or are
you connecting to some other
equipment?
root linux wrote:
My
Please confirm if PRI span is up
on CLI, type pri show span 1 it must be UP before you can dial through it.
regards,
Umair bari
On 8/25/05, root linux [EMAIL PROTECTED] wrote:
Yep, I am connecting to some other equipemnt...its a
Clarent gateway equipped with a National Microsystems
(Quad
Bob Goddard you are right but i said in my previous
mail that i am still getting this problem
some body replied me and i have followed this link but
still same problem and asterisk is stoping.
http://www.voip-info.org/wiki-file+descriptors
On Wednesday 24 Aug 2005 13:40, Kamran Ahmad wrote:
Hi,
Been asking google and browsing the lists but haven't found any answers
for this. I've connected a TE110P E1 using EuroISDN to a PBX (for me
at the time unknown model). All is fine _except_ when placing calls to
mobile phones (which takes too long, more than 2 seconds it seems)
asterisk seem
Hi All
I just completed a custom application for Asterisk (i
m not a C guru so i just copy codes from other
application and alter according to my needs)
attached files is the source file
this application is working fine but still i need you
people to give suggestion to improve it
Primary task
I'll jump in here with one comment. I worked with an individual in Canada
that could not get rid of the echo (some time ago with an x100p). As a
very experienced telephony engineer and two years of asterisk experience,
I logged into his system and tried many many changes without impacting
the
On Wed, 24 Aug 2005, Steve Gladden wrote:
I'm looking for some help in how to keep asterisk from doing this.
If we loose Internet or routing to our upstream provider even for only a
few short minutes asterisk quickly gives up never tries again.
I have to do a manual reload to get it to
Hi Lee:
NTP is working as expected, but it does take a couple of minutes (!) to get the date from the server
Jesus Mogollon
2005/8/25, Lee Archer [EMAIL PROTECTED]:
Hi, do you have an on-site NTP server? I found that
after the firmware update NTP from the * server stopped
working.
See it thanks... seems rather sparce on documentation... how does
one go about turning the jitter buffer on?
On 8/25/05, Richard Scobie [EMAIL PROTECTED] wrote:
Matt wrote:
Am I correct in thinking that at this time the CVS-HEAD supports
Jitter Buffer for SIP on Asterisk?
No, but
If you can get your POTS lines terminated in T1 lines, thats probably
best. Then you can get a digium Quad Card. Otherwise, if you're going
to terminate more than 8 lines, you should probably use a channel
bank.
On 8/25/05, Gulzar Hussain [EMAIL PROTECTED] wrote:
Hi All
I want to terminate as
On 8/25/2005, Gulzar Hussain [EMAIL PROTECTED] wrote:
Hi All
I want to terminate as much POTS lines as possible to
my Asterisk Server, please advice me which Card to
choose with accessories
Perhaps you should look up channel banks, there's info in the Wiki
about them. look for FXO channel
First off, thank you *very* much for this unbelievably informative post!
I've
got it saved away now along with Kris Boutilier's adjusting rxgain/txgain
post.
On Wednesday 24 August 2005 17:14, Bruce Ferrell wrote:
At the point where the phone line get's to your demarc the is supposed
On 8/24/05, Keith Caldwell [EMAIL PROTECTED] wrote:
I've never set up asterisk in Tiawan but I had a few issues like that
here in the U.S. I solved it by putting a pause in the dail command
so that asterisk could fully open the channel before it started to dial.
exten =
Well it's only worked once and I've left the phones several
hours. I've done various debugs and the phone is asking for NTP and the
server is answering but its not getting set.
Lee
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jesus
MogollonSent: 25 August 2005 12:54To:
Greetings all,
We are settng up a fair sized
call center on Asterisk, but we are having some issues with our agents not
knowing if they have logged in and logged out. Prior to beginning our
migration to VoIP the agents logged into our nortel phones and confirmation was
displayed on the
Which side of the span is actually providing the clock? The Asterisk side
or the other side (telco?) If it is the telco the clock needs to be set to
'0'.
Span=1,0,0,cas,hdb3
-Nate
-Original Message-
From: root linux [mailto:[EMAIL PROTECTED]
Sent: Sunday, August 21, 2005 10:30 PM
To:
Any Deskpro EN has that feature from the P-166's up until Carlygate. It's an
extremely common used PC, since there were a bajillion EN's leased and then
liquidated. Note: You need Celeron class or higher to work with a TDM card,
a PII's bus is not *quite* PCI 2.2 compliant and the card won't be
Note I am also using a couple of SIP phones at home, Snom 190's, and they
work fine. Your ZTTEST is good, no problems there. But changing the system
is worth a shot, all you have to do is move the TDM card over and the hard
drive. Kudzu shoud do the rest.
Might be worthwhile to hang a SIP phone
On Thursday 25 August 2005 08:00, Rich Adamson wrote:
If you mean placing a transmission test set at the customer's demarc (at
the customer's site), the -2 to -3 db is still incorrect for analog
pstn circuits. That level _will be_ the 0db generator tone minus the cable
loss from the CO to the
Hey guys,
We have a brand new Dell Poweredge 1850, Single Proc 3Ghz, 2GB RAM,
15K RPM HD's RAID 1.
We also have a Sangoma 4 port T1/PRI card.
We are not using G729. Everything is G711.
Every call is PRI - Asterisk G711 - Sip Carrier
We just filled up 2 PRI's and reached a CPU usage of
Hi all,
This is to let you know that I found out how to automate the
agentcallback logon and logoff. Only thing you need, is to have the
agentcode and pincode available in channelvariables.
I've updated the documentation on voip-info to incorporate my findings.
Patrick Tracanelli wrote:
Hello List,
This is my first message herein. I was playing around with System() and
AGI() and found out something I cound not determine my configuration
error. I added before.agi and after.agi to the agi-bin dir. Tried to
make before.agi get run before the dial
I am trying to test the T1 card in our legacy PBX but the connector to the
card is a 15 pin serial cable. I would like to make it myself so I can try
this test today. Does anyone have a pinout for it?
I just made a T1 RJ-45 loop back to test my TE110P and it tested out fine.
I'm trying to resolve
I'm using a current version of CVS-HEAD and am having an interesting problem
with a key user of a simple IVR application. He's dialing into the system
from the PSTN via a TDM400P.
The application (written in php with phpagi) plays a prompt, which can be
interrupted at any time, asking for a user
I am located in the UK, and I am using Sipura spa-2000 adapters to
connect analog phones to a voip network. The network connects to the
PSTN as well via the Sipura spa-3000 adapter.
I would like to provide surge protection for the spa-2000 and the
spa-3000 adapters.
1. For spa-2000, fxs
I don't know what the problem is, but this is what I use and it works on
my analog FXO port.
exten = _9NXXNXX,1,Dial(${PSTN}/w${EXTEN:1})
John Novack wrote:
Or, in the example below, wait before dialing?
exten = s,1,Dial(ZAP/g1/${ARG1},360) ; ARG1 is the number to be
dialed
If
I can time sync with time.nist.gov but not with any
internal servers. I read in the changelog about them fixing something
related to NTP on the same subnet but it doesn't say whether it should work or
shouldn't.
Regards
Lee
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Thanks for the responses. All is happy. For the record the correct
answers are:
Q1 - Additions/changes SIPxx.cnf take effect on reboots, deletions
do not.
A1 - Don't just comment out the line setting, change it specifically to
UNPROVISIONED.
Q2 - How to get Message button working.
A2 -
I updated 2 weeks ago and am due to update again...
So Yes I will update
It seems that the giving up forever feature is by design,
As I had seen a post about it awhile back...
But I would rather not have asterisk give up (forever) if it can't
see a sip server.
I feel retries should
Or, in the example below, wait before dialing?
exten = s,1,Dial(ZAP/g1/${ARG1},360) ; ARG1 is the number to be
dialed
If you are using analog ports, yes. Dial(Zap/g1/ww15551212).
exten = s,1,Dial(ZAP/g1/ww${ARG1},360) should work then?
Why in the world would you ever want to do that
You would? Why not just put then on a small UPS and have done with it?
The UK has some of the cleanest electricity in the world.
Unlike the US (what a big shock that was, moving here) where brown-outs
and over volts are common I've never needed to add protection devices to
the UK supply.
Asterisk User Group wrote:
Thanks for the responses. All is happy. For the record the correct
answers are:
Q1 - Additions/changes SIPxx.cnf take effect on reboots, deletions
do not.
A1 - Don't just comment out the line setting, change it specifically
to UNPROVISIONED.
Q2 - How to
Hi I had same problem, I commented
pridialplan = local
in /etc/asterisk/zapata.conf
and the outgoing calls where ok again. Remember to restart Asterisk not only
reload!
Ciao
Mauro
___
--Bandwidth and Colocation sponsored by Easynews.com --
I think he's talking about putting protection from the PSTN lines not
the incoming power.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Sent: Thursday, August 25, 2005 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial
Is anyone out there using Lucent brand equipment to handle an incomming
DS3, converting all 672 calls to SIP (as G729) and sending those to
Asterisk/SER over ethernet?
If you are and are willing to speak to my boss about your experiences
(over the phone) with it, please contact me off list.
Try suggesting to BT that their copper sucks!
Jonathan k. Creasy wrote:
I think he's talking about putting protection from the PSTN lines not
the incoming power.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Sent: Thursday,
Pin 1 = Send - Tip
Pin 9 = Send - Ring
Pin 3 = Receive - Tip
Pin 11 = Receive - Ring
Pin 2 4 = Ground
Plug Pins:
8 7 6 5 4 3 2 1 - Wide side
9 10 11 12 13 14 15 - Narrow side
Bart
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Geoff
Manning
Thanks very much for the answer! Yes, the UK has got a very good
electricity supply - so that wasn't really my worry. What I am more
worried about are lightning strikes during thunderstorms. And in our
setting the distance between my spa device and the analog phone can be
quite long with some
If you want SIP phone PBX hosting or residential partitioning, I can't help.
If you want traffic termination(National and International), we can do it.
Regards
Leon Sun
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jennyw
Sent: Thursday, August 25,
jennyw wrote:
Hi,
Just wondering if people could suggest a good VoIP provider that can
service the San Francisco Bay Area and the Los Angeles area. I've tried
race.com (recommended to me) but they're kind of hard to get ahold of.
Any other suggestions? This is for a business, so reliability
http://www.voipzoneenterrprise.com DID's in 92% plus of the USA, can
provide full Enterprise solutions from SIP2.0 to Internet access.
BRW
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leon Sun
Sent: Thursday, August 25, 2005 10:04 AM
To:
Hello,
Looking for a bit of feedback on * server being able to handle SIP 2.0 to
MGCP ? .. Need to talk with Sylantro CA224 client adaptors and would like to
find some sort of gateway solution between SIP 2.0 and the Sylantro CA.
Ideas? ..
BRW
___
I've just setup a new asterisk box (cvs HEAD) with a digium tdm411 and
a couple computers with eyebeam. I have one small. I cannot call the
eyebeam clients from the phone connected the fxs port. I can call the
phone from the eyebeem clients. And, I get both the fxs phone and
eyebeam clients to
Damm my eyes .. Correct URL is http://www.voipzoneenterprise.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Watters
Sent: Thursday, August 25, 2005 10:29 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
Bad URL... Too many R's in there... Correct...
http://www.voipzoneenterprise.com/
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Watters
Sent: Thursday, August 25, 2005 10:29 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Y'all will know me on this circuit for the past year or so, and you'll know
I've done some pretty intense testing with various aspects of Asterisk and
Zaptel drivers.
The Kris Boutilier's modifiecations to the MARK2 echo canceller are A#1. I
have always had a little residual echo on my home
Hello,
Would you please suggest me, where can I buy g723.1 liscence in cheap.
I might need a liscence for 10-50 channels.
Thanks,___
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi,
Can someone help me understand why I can't make calls to cellphones?
Last week I edited my extensions.conf and created a context for outbound calls
[outgoing]
exten = _92XX,1,NoOp(Call for ${EXTEN:1})
exten = _92XX,2,Dial(Zap/1/${EXTEN:1})
exten =
Hi,
It seem asterisk begin CDR when ZAP channel on FXO ring. I want to detect
On-Hook callee party to launch a macro.
Someone known a good solution?
Thanks
--
GSM : 00212 60 54 65 68
WEB : http://www.jeremy-salmon.org
___
--Bandwidth and
Forgive my ignorance, what encapsulation would you use on the ISP end of
the T1? This is for data also, correct?
-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 24, 2005 2:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Innocent Evil wrote:
Hello,
Would you please suggest me, where can I buy g723.1 liscence in cheap.
I might need a liscence for 10-50 channels.
You can't.
Here is the licensing priceing info for G723.1 direct from the patent
holder's web site (it's not cheap):
On 8/25/05, Lance Grover [EMAIL PROTECTED] wrote:
Now I have another, whenever I call out and make a succesful call,
about 10 seconds into the call the phone call is cut off and hung up
by asterisk, any Ideas?
I have now tried the lattest zaptel drivers for the 4 port tdm card
(wctdm) and
why dont u try this, this works fine, and i am send ing call to cisco ,
quintum by this g723 .
http://aussievoip.com.au/wiki-G723-1-Install
Bashir Ullah
Lamsre Informatics Limited
- Original Message -
From: Eric Wieling aka ManxPower [EMAIL PROTECTED]
To: Asterisk Users Mailing List
On Thu, Aug 25, 2005 at 05:50:29PM +0200, Mauro Zanin wrote:
Hi I had same problem, I commented
pridialplan = local
It's allready commented. But AFAIK this has to do with how numbers are
beeing sent to the PBX. This wouldn't explain why calling regular
numbers that are forwarded to a cell phone
On Wed, 2005-08-24 at 09:04 -0500, Eric Wieling aka ManxPower wrote:
Joseph wrote:
On Tue, 2005-08-23 at 12:22 -0500, Eric Wieling aka ManxPower wrote:
I don't know what the problem is, but this is what I use and it works
on
my analog FXO port.
exten =
On Thursday 25 August 2005 08:00, Rich Adamson wrote:
If you mean placing a transmission test set at the customer's demarc (at
the customer's site), the -2 to -3 db is still incorrect for analog
pstn circuits. That level _will be_ the 0db generator tone minus the cable
loss from the CO to
Is there a way to dial DTMF after bridging the call.
The current option D() in Dial will dial DTMF before the call is bridged
and this doesn't do the job.
I need to dial DTMF after the call is bridged and the message is played
with Background
--
#Joseph
Olle E. Johansson wrote:
We really need test input of the latest patch in this issue report. And
we need them today. If you are interested in device state notification
in SIP - stop whatever you are doing and give us feedback NOW!
Thank you for your assistance!
Not strictly a problem with Asterisk but one of my phones. A couple of days
ago I decided to update the firmware in my Optipoint 600 Office which looked
as though it went swimmingly until that is, it rebooted.
Since then the phone just boots up and displays the following:
Can't Boot!!
Does anyone have a config they'd like to share w/ the above hardware
doing termination for asterisk?
I've got one coming in tomorrow along w/ some DSP's and would like to
not have to create the config from scratch to start testing.
W. Kevin Hunt
___
Can anyone comment or share experences with using Dell 2850's with Asterisk.
Proposed config is 2850, 2 x 3.6g procs, 2 g's of ram, 4 x 36g 15k rpm
drives raid 10, Digium TE411P ( the echo cancelling cards ).
Expected load is 1 or 2 pri's (most likely 1 ) 100 Polycom phones on the
local
List,
I have begun to experience a strange echo problem on our
internal network. The problem starts when User A calls User B,
User A puts User B on hold. User B heres the on hold music.
User A returns and User B has trouble echo. I am using FC1,
Asterisk 1.0.9.
This electronic
We successfully use 2850s with Digium T1 cards, though I don't think we've
installed a TE411P. It'll handle two T1s with ease.
You don't need the second processor or the second GB of RAM for the expected
load. For your configuration we would usually use two single processor 1u
servers with
Does anyone have a working NFAS config for Zapata and zaptel
for 2 NFAS trunks? First two DS1s on tg 1 and other two on tg2?
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Hi all,
One of my clients had been sending me issues with G729 codec by
Digium. According to him,
the Digium codec is able to send calls into a Cisco AS54xx and AS53xx
gateway via SIP, however,
when calls are originated from the AS, asterisk using Digium G729 is
unable to receive the call
Eric Bishop wrote:
Hi all,
Our Asterisk box sends calls outbound via either SIP (through our VoIP
provider) or an E1 PRI (directly connected via a TE410P). When we dial
a number that is engaged via our VoIP provider we get the following on
the Asterisk console (numbers and IP addresses changed
Anyone using this in their Asterisk installation? I just ordered one to
play with. Hoping that I like it better than the Hitachi Wifi phone.
Michael
--
Michael Graves [EMAIL PROTECTED]
Sr. Product Specialist www.pixelpower.com
Pixel Power Inc.
Ok, fxotune is a work in progress so to speak. I fixed something in it about
a week ago that may help it adjust to the line better (whereas before I'm not
sure that it was at all). Try the latest CVS-HEAD version of fxotune as your
first step. (oh, after you use fxotune you should turn off
I talked to Digium about this and they are saying the best thing may be to
get the information from the manager API and display it on a PC if I cannot
find a way to get the data into the phone. I plan to keep looking into
this, I'll share whatever solution I end up with. Thanks for your
Nope, working as we expect.
Any more clues?, unable to receive...Correctly is not a easy to
understand tech. phrase.
LTenorio
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nir
Simionovich - CTO
Sent: Thursday, August 25, 2005 7:47 PM
Search for some of the configs for AS53XX out there. They are pretty the
same.
LTenorio
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of W. Kevin Hunt
Sent: Thursday, August 25, 2005 5:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi Bill,
We just built one for a customer with Fedora Core 3 and a TE210. We get PCI
parity errors and the machine shuts down. I'm sure we'll get it working,
but it hasn't exactly been the smoothest install ever.
I agree that the second CPU and GB of RAM is probably overkill and as you
know,
what kernel are you running ?
On Thu, 2005-08-25 at 17:01, Rich Adamson wrote:
Ok, fxotune is a work in progress so to speak. I fixed something in it
about
a week ago that may help it adjust to the line better (whereas before I'm
not
sure that it was at all). Try the latest
Hello,
All Im looking for is a yes/no answer here. I have
heard that the following scenario is possible (reasonably easy to implement as
well) but I just dont get it :-) if it is possible Ill
go ahead and learn on my own, I just dont want to waste time on
something that will not work.
I have a C++ wrapper class around the cagi class that is listed on the wiki.
It doesnt implement everything and to tell you the truth is still in beta.
But it works... and yours if you want to help testing it.
Rene Kluwen
Chimit
Being a lazy person, I was wondering if anyone has a c++ class
I'll do my comments in line and hope I don't offend.
Rich Adamson wrote:
First off, thank you *very* much for this unbelievably informative post! I've
got it saved away now along with Kris Boutilier's adjusting rxgain/txgain
post.
On Wednesday 24 August 2005 17:14, Bruce Ferrell wrote:
At
and just for REALLY bad form, responding to my own post with a postscript:
I had to have a lot of pictures drawn for me when I was learning this 20
years ago :)
Bruce Ferrell wrote:
I'll do my comments in line and hope I don't offend.
Rich Adamson wrote:
First off, thank you *very* much
For Asterisk to play MOH, it will need to have an RTP connection, right?
How otherwise, would you want to play MOH?
Rene Kluwen
Chimit
For canreinvite=yes to work, I think I need to remove the t argument in
the Dial(SIP/ext|60|t) application. Otherwise, Asterisk will allways
stay in the
Hello,
We have experienced problems with our Dell 2850 as well. Asterisk would
slow down and people calling in would not be able to get through (receive an
engaged signal). I had a chat to Digium and they advised that they
experienced occasional issues where the Intel gigabit ethernet module
The real question is:
*Can I have no RTP bandwidth consumed by the Asterisk server? (SIP
data allowed) Supposedly the 2 VoIP phones can talk to each other
directly through the NAT once STUN and SIP do their ***magic*** to
establish their RTP connection.*
If the NATs are NOT
WARNING: This has been cross-posted in the Asterisk-BIZ mailing list.
Hello, we are looking for a unix programmer who can write the
following type of application that works with asterisk.
1. Operator Console.
When a call comes in on a did, the did should be looked up in a
database(against a
On Aug 24, 2005, at 7:40 PM, Doug Lytle wrote:[EMAIL PROTECTED] wrote: On Aug 24, 2005, at 11:21 AM, Doug Lytle wrote: Create an extension that the user to be marked knows about, maybe even have it authenticate, mark the user and drop them into the conference.Doug If the Marked user isn't the
I just setup a Dell 1800, not a 2850, which is working awesome.. only had to
disable USB, which realistically no-one on a phone system would care about
anyways. Using 75 Cisco 7960's with it. Works like a charm. It has a Digium
TDM04P and a TE110P card in it.
Haven't had to many echo problems as
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