On 10/17/21 12:59 PM, cio-al...@playerschool.edu wrote:
I did test manually and the NFS mount works fine. I do create a
directory and it shows at the server.
I am using containers, indeed. How can it be affecting Asterisk that I
am using LXC containers?
I'm by no means an expert in
I did not explain myself well, for this I apologize.
The files never appear on the NFS mount, only in the local drive.
Restarting Asterisk with the mount on does not fix it.
Asterisk simply ignores the mount and writes to the local drive.
But the mount is fine, I can create a dir and it appears
Thanks for that info, Ben. I do like to test out the latest and most
up-to-date versions of things when I can, so I'll check those files
and see how it goes.
On 2020-05-11 17:20,
Ben Ford put forth the proposition:
> Hey Dave,
>
> In the case of 13 and 16, these are LTS versions wh
currently using 16.8.0 and wondering if I should upgrade to
16.10.0, or perhaps give 17.4.0 a try.
Are there any differences I should be aware of, like config file
syntax or similar things?
Thanks,
--
Dave
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asterisk-16.8.0
Hi
I've set up a callback script to retry a number if it's busy, but as
I watch the console output asterisk seems to rush 3 or 4 calls at
once before waiting the RetryTime of 20 seconds that I've set.
The script:
-8<--
CALLERID=$1
EXTENSION=$2
TEMP=`mktemp
>> A good Ethernet cable-pair tester can spot such things pretty quickly.
>
> I disagree.
>
> *Certainly*, incorrect pair terminations can cause the sort of problems
> described, however I haven't yet come across a cable tester which can
> identify
> that a cable correctly connected from end
> I looked at your network diagram. Try checking the configuration of the
> Ethernet ports on the firewall and the Asterisk box. Make sure they are
> set to auto-negotiate and not set to a fixed speed and fixed duplex.
> I have found in the past that if one end of a link is expecting auto-
>
> Hey all
>
> I am trying to register a PJSIP server on our office to an Asterisk 11
> chan_sip server in a datacenter.
>
> I keep getting
> WARNING[18084]: res_pjsip_outbound_authenticator_digest.c:178
> digest_create_request_with_auth_from_old: Host: 'XXX.XXX.XXX.XXX:5060':
> Unable to
http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/
--
Dave Topping
e: i...@dntopping.uk
t: 03445 888 888
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I'm not sure of the precise specifics of how Digium runs the
list, but this sort of problem has been a "known issue" with
mailing list distributions ever since SPF and similar technologies
showed up, almost a decade ago. DomainKeys and DMARC makes it more of
an issue, but the overall problem is
> Not sure maybe there's a better solution but I thought about using another
> peer with type=user for incoming connections.
That's what I've done for my connection to the service provider
I use (Vitelity), as they have different inbound and outbound
hosts/proxies. This works fine.
--
>>> so the main question is -- how to Disallow CALLS without registering
>>> on PBX
> In fact, I'm not sure that it's actually possible to disallow [authenticated]
> calls from a peer that hasn't registered!
>
> As far as I can tell, 'registration' was never intended to be part of the
>
> So does the Dial command go directly to the registered device or does
> it use the extension?
If you've given the Dial() command the SIP/user1 format, it will attempt
to dial directly to the SIP device/phone/endpoint you specify. If you
specify SIP/user1/user2&... it attempts to dial
> I am using ODBC realtime storage with Asterisk. Currently, with no password
> set, a user can dial the voicemail number to retrieve their own voicemail,
> without needing to enter a password (without hearing the password prompt).
> However, there is still a 'mailbox' prompt played, and if a
>OK. Maybe an echo canceller won't make any difference. But why does the
>remote side _always_ hear an echo if we use a local dahdi extension,
>and _never_ when we use a local SIP extension ??
The echo that the remote called hears, might be of either electrical or
acoustic origin.
If
> Thanks Jeff, just to confirm, password are not sent in plain text? I
> want to safeguard against man in the middle attacks, sniffing traffic of
> clients.
That's correct.
The way it works is:
- Both the client, and Asterisk, know what the password is.
- The client sends a SIP message
Now I have the problem for my cellphone... I need to register from almost any
IP (at least in Europe), so I can't restrict it.
Well, the password is NOT simple and random.
Now, I tried to register the user of my cellphone using a PC, as my cellphone
was already registered.
And Asterisk
1 - My SIP server (Asterisk) will have some SIP clients registered in its SIP
registrar. Let's say 6 SIP clients. In my project I have to implement a way
of a SIP client making a call to a number and all others 5 SIP clients ring.
That is, the others 5 SIP clients must receive the SIP
Someone on this list uses the address @sedwards.com
I doubt this is their actual email address as there is no MX record for
sedwards.com and I can't find registration for their domain either.
Part of my mail servers reject these emails because they cannot be
replied to, or are likely
Hmm the calls are made during the day (and sometimes very early in the
morning). Right now it looks like someone actually made these calls. If
that is the case it's somewhat comforting to know the system wasn't
compromised. However, the $25,000 phone bill still remains. Yikes. $6.25
per
On 10/23/2014 05:00 PM, Matthew Jordan wrote:
On Thu, Oct 23, 2014 at 3:32 PM, Dave Fullerton
dfullertaster...@shorelinecontainer.com
mailto:dfullertaster...@shorelinecontainer.com wrote:
Hello all,
I'm setting up a couple of test boxes and I'm running into a
problem. What I
!
--Tim
I can't help with your root problem (maybe check core show function
FAXOPT?), but the spandsp site is up. Try using www.spandsp.org.
Downloads are available here: http://www.spandsp.org/downloads/spandsp/
-Dave
-lan
send_rpid=no
send_pai=yes
direct_media=yes
tos_audio=46
tos_video=34
Is there something I'm doing wrong here?
Thanks
-Dave
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Is the destination Number like Country Code +972?
+972 59 xx(x) mobile - Jawall [moving to 7-digit subscriber numbers]
source - http://www.wtng.info/wtng-972-il.html
My SIP Proxy logs all the unauth. INVITEs and I found the a lot calls go
to the Country code +972 xxx
I've
On 07/17/2014 09:46 AM, Dave Fullerton wrote:
Hello all,
I'm running into an issue with Asterisk 12.4 and IMAP voicemail. I
have asterisk set up to connect to my Dovecot IMAP server and I can
leave and retrieve messages from my inbox and old messages. However, I
am unable to move messages
in my configuration or is this a bug?
Thank you
-Dave
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Problem with this is client needs to listen to the call recordings and my
interface will only display .wav or .mp3 so they will moan if they have to
wait until the next day for today's recordings
If you're up to writing a bit of shell script, and are running
on Linux, you could automate the
system will not send
messages via email. I think it is because of the gateway change. How do I
change the gateway address? Is this product something I could contract out to
have remote support? Thanks,
Dave
Just checking the transcoding on our Asterisk boxes and I get the
following results.
I have the g726, ilbc and lpc10 formats and codecs enabled in 'make
menuselect' so I dont understand why its showing as no translation path.
Any ideas?
Are the modules actually loaded?
Try doing a module
this is an interesting project, SIP protocol is easy to find, writing a php
script, perl script, or python would probably work. it would probably work
better if it was a daemon. what would be connecting to it that you would need a
SIP connection for?... interesting...
Dave Cantera
(856)813
In my case, I have good incoming quality and terrible quality going out.
That is, I can hear people perfectly well but they complain that my
voice drops out and is garbled regardless of who places the call.
This suggests to me that you may have congestion problems in your
upstream traffic
for thought,
Dave Cantera
(856)813-7098 mobile/txt
david.cant...@ibsonecall.com
Sent from my ASUS Pad
Steve Totaro stot...@totarotechnologies.com wrote:
On Wed, Sep 25, 2013 at 3:22 AM, Endri Stefani endri.stef...@plus.alwrote:
Hi
** **
Greeting to all you out there.
** **
I
://www.apc.com/products/family/index.cfm?id=145
I'll second the APC option. A PRM4 and two PTEL2 will protect 4 lines
with a little wiring. Make sure you have a good ground to connect to or
the whole thing is worthless.
-Dave
`/usr/src/linux-3.4.45'
make: *** [modules] Error 2
-Dave
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On 06/10/2013 11:53 AM, Shaun Ruffell wrote:
On Mon, Jun 10, 2013 at 11:33:16AM -0400, Dave Fullerton wrote:
Not sure how I should officially report this...
You should feel free to open issues at http://issues.asterisk.org.
but I'm getting a compile error with DAHDI-linux 2.7 when I define
, but configuring the digitmap to match your environment
is the best solution. Check the SIP admin guide for details on how to
set it up.
-Dave
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/support/setup_maintenance/products/voice/spip_ssip_vvx_Admin_Guide_SIP_3_2_2_eng.pdf
Read chapters 2 and 3 at a minimum. There is a lot to setting up a
provisioning system for polycom phones and it helps to have the proper
background before getting started.
-Dave
On 04/12/2013 01:50 PM, Daniel
Is there any way to force this? I have several user agents and I want to
achieve
near 100% availability for all peers. I realise that the peer will be 'woken'
up
at my qualify intervals, but can I actually force registration from the CLI?
For those peers which are at known, fixed,
Thanks Matt. The suggestion helped. No more slip erros.
Dave
Original Message
Subject: Re: [asterisk-users] dahdi timing source multiple cards
From: Matthew Fredrickson cres...@digium.com
Date: Fri, December 21, 2012 3:41 pm
To: asterisk-users@lists.digium.com
You
totchans=24
irq=50
type=digital-T1
syncsrc=0
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=B8ZS,AMI
framing_opts=ESF,D4
coding=B8ZS
framing=ESF
[root@aislecom28502 dahdi]#
Thanks,
Dave George
AIsleCom Inc.
1 473 520 1000
1 561 674 3838
Here's where I am baffled and I am hoping someone with intricate
knowledge of this implementation may be able to explain it to me. What
we had to do to get this working was to set the host= parameter to the
respective endpoint IP's of the VPN tunnel, 172.10.1.1 in my case, and
172.10.1.2 in
. It should ONLY be used to upgrade phones from SIP 3.2 or lower to
SIP 3.3 or higher.
Hope this helps.
-Dave
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Setting up a group of analog lines to use for outbound emergency calls
(911). My current dial plan and debug output shown below. It appears
that when the SoftHangup() is executed that the line does not really
hang up. In the case shown, I had reduced the group to a single DAHDI
Here is my IP-PBX setupmy setup is : sips softphone - asterisk - xorcom
PSTN gateway - pstn line to telcoi'm using xlite for windows
when I make a phone call (sip - outgoing channel),I can hear my own voice so
clear. it's very annoying mewhen talking a little loud... any solution?
Two
turn that button off when I set up my site sip.conf
to avoid any questions.
-Dave
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I must be missing something. If a phone sends a UDP packet to
192.168.1.1, how does that get routed to (arrive at) the 10.0.2.1
interface on the Asterisk server? The only way I can imagine that
happening is if a router in between the phone and the server has been
told that 192.168.1.0/24
don't have this issue when calling from a SIP phone. I only have this
issue when calling from one media gateway to the asterisk box.
Any suggestions welcome. Can I play some file in the back while
collecting DTMF?
Dave
any RTP
packet to WCM.
How can I enable the option to allow asterisk to maintain the RTP stream
during DTMF collection?
Thanks,
Dave
Original Message
Subject: Re: [asterisk-users] Loss of RTP stream during DTMF collection
From: Kevin P. Fleming kpflem...@digium.com
Date: Fri
In our app we do not forward packet immediately. After enough packet
received to increase rtp packetization time (ptime) the we forward the
message over raw socket and set dscp to be 10 so that this time
packets can escape iptable rules.
From client side the RTP stream analysis shows nearly
Ouch. That isn't going to be so easy to spot, then! You would have to guess
a bunch of likely passwords, fake up a challenge with some known nonce, and
compare the response against those you would expect with each of the various
possible passwords. (You've already got the Source Code
5. Placing ferrite cores on the phone cables.
Do either of the phone lines in question have DSL on them?
If so, a ferrite core (which will block common-mode RF
signals) probably won't help much, if at all. DSL is a
differential-mode signal, and its frequency content starts
down in the tens of
)
Sent RTP packet to x.x.x.x:22760 (type 00, seq 005633, ts 152368,
len 000160)
Got RTP packet fromx.x.x.x:22760 (type 00, seq 042371, ts 177600,
len 000160)
Dave
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Which version of asterisk are you using? I just have this in 1.4 and it
works fine:
SIPAddHeader(Alert-Info: intercom);
-Dave
On 02/14/2012 08:10 PM, Mike wrote:
In case anybody was following this thread, or someone Googles it in the
future, here is the solution:
This worked fine
voIpProt.SIP.alertInfo.2.value=page
voIpProt.SIP.alertInfo.3.class=autoAnswer
voIpProt.SIP.alertInfo.3.value=silentanswer
/voIpProt.SIP.alertInfo
/voIpProt.SIP
/voIpProt
/polycomConfig
I have also added an se.rt section to adjust the ringer and timeouts
for these ring tones.
-Dave
On Tue, Dec 6, 2011 at 4:05 PM, white hat whitehat...@gmail.com wrote:
Would you be willing to post sanitized versions of your jabber.conf,
gtalk.conf and details regarding the context you're using and how your
inbound route is configured in your dial plan?
Are you using STUN? Is Asterisk
On Sat, Dec 3, 2011 at 12:59 AM, white hat whitehat...@gmail.com wrote:
When a caller calls my google voice phone number, I must answer, wait and
press one to accept. Sometimes even that does not work.
I just need a little advice on how to write the dial plan. I still have
much to learn
I've been trying to find a solution that would allow our sip phones to
communication with walkie talkies. Our setup is that we have sip phones
setup in 2 locations, headquarters and dome. We can communication from
headquarters and dome through sip phones, but within the dome we have
).
-Dave
On 11/17/2011 11:57 AM, eherr wrote:
Doing it that was does accomplish the original question, which is cool. Thanks.
But you're also right in that we wont like it.
This setup only allows for her extension to be registered to just one line key,
unless I am missing something.
So
Is about outgoing calls from multiple devices with the same username at
aprox same time. The overwritten is for incomming calls. I want to prevent
using the same account in multiple devices at same time. The solution with
IP will not apply because users may be behind nat or will change
/asterisk-users
--
+
Dave Aibel
President CEO
Pervasive Telecommunications, Inc.
email: dai...@pervasivetelecom.com
(603)367.3512
(603)367.9942
(401)862.4203 (c)
dai...@pervasivetelcom.com
Great discussion, all of it. Thanks, people.
How much power does the home asterisk box need ?
I'm using Asus Eee Box (1012Ps) as Myth front ends in another project.
About $280 with 320 Gb hard drive and 2 GB RAM. Atom 510 processor. Built
in Wifi. Nearly silent. Runs F15 nicely.
I opened the jumpers on the card putting them in T1 mode and it worked. I
had them set to T1 using the options in the dahdi.conf file under
/etc/modprobe.d/
That worked well for over a year until it started acting up.
Dave
-Original Message-
From: asterisk-users-boun
3.3V (rev 02)
0a:03.0 Communication controller: Digium, Inc. Wildcard TE410P/TE412P
quad-span T1/E1/J1 card 3.3V (rev 02)
Thanks
Dave
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Hi,
We opened the server an checked that the cards were seated correctly and
they are. I will have the tech completely remove them tomorrow and try
again. I will post the results.
Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
,b8zs
bchan=169-192
span=9,1,0,esf,b8zs
bchan=193-216
span=10,1,0,esf,b8zs
bchan=217-240
span=11,1,0,esf,b8zs
bchan=241-264
span=12,1,0,esf,b8zs
bchan=265-288
Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
0:00.00 cqueue/0
167 root 18 -5 000 S 0.0 0.0 0:00.00 cqueue/1
Dave
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On 07/29/2011 06:56 AM, --[ UxBoD ]-- wrote:
Hi,
compiling up a new installation of Asterisk 1.8.5 on CentOS 6 X86_64 and
am seeing the following when running the make:
/usr/bin/ld: skipping incompatible /usr/lib/libpam.so when searching for
-lpam
/usr/bin/ld: skipping incompatible
They've got a bunch of Grandstreams that seem to be rock solid... until
7:00pm. At 7:00, some of the phones become unavailable, and stay down. Call
quality is solid almost all the time. But right at 7:00, things go bad.
Only
some of the phone lines go down and they stay down until the
I need to keep out all connection from 5 countries, which originate
most of the Denial of Service attacks. The entries are around 9000 if
used as xx.xx.0.0/16. I heard that there is a smarter way to do this
by using User Tables in iptables, that will keep the speed equal to
LOG(x). I
dan, elder,
I have played with scripts to generate calls and track their
completion, email me off-list if you have questions.
daveC
Daniel - Asterisk wrote:
Hello Everyone,
I wonder if someone could share a manual about using SIPp for
Asterisk's testing.
I'll be gratefull
Regards,
danny,
not that it matters, but I agree. if the design is a good design, it
would not have to be redesigned on every release. in fact, the modules
template should also follow this philosophy that way you can concentrate
on adding functions and not the design...
sometimes, it is smarter to
doug,
why are you shaking!?!?... do you have a better recommendation?
daveC
Doug Lytle wrote:
C F wrote:
model name : AMD-K6(tm) 3D processor
*shudder*
Doug
--
SJREIA South Jersey Real Estate Investors Association
Want to invest in Real Estate?
come out and join over 450 real
paul, doug,
I had several AMD athlons 64bit... no problems running centos, suse.
they seem solid on 1.4.xx... had a few intel celerons and P4s. they
were good as well. guess I was Lucky back then!
thanks for supporting the list!
daveC
Paul Hayes wrote:
On 04/05/11 17:10, || dave cantera
I've been away from asterisk for a while since 1.4.16 and only installed
1.6 once to run a test... can someone recommend what the best version to
install is and the recommended CPU/motherboard for an * box these days?
I'm just running about 20 handsets and 4-8 lines with POTS SIP mix.
I
.
Dave Cotton
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asterisk-users mailing
On 10/03/11 12:55, Gilles wrote:
On Tue, 08 Mar 2011 13:22:18 +0100, Gillescodecompl...@free.fr
wrote:
I need to write a script which prompts the callee to type a number,
and then read it back to them as confirmation:
Apparently, the right way to read a phone number back to the user is
not to
I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the
office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On
the office side, they hear an echo of _their_ speech, not mine.
The office uses sip-providers generally without any echo problem.
Where do I start to
Actually, I don't think that has been the case for quite a while. Anyone
can get the latest firmware directly from polycom. Including, 3.3.1F
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html
On 02/24/2011 03:32 PM, Mike wrote:
Sorry, I realize my tone might not go down
How about encrypt the whole hard drive?
If I built a server and give to other people, there is no easy way to
stop them reset the root password or just mount my drive to read
everything on it. But if build an encrypt OS then it will be secure.
It will be more secure. However, you
In the meantime, does anyone have a nice way to update a stable/stock lenny
installation with the updated glibc as well as the latest kernel
Scary and risky, as others have noted!
There is an official backports release kit associated with Debian,
which contains newer versions of many packages
I know this is an {*} list but does anyone know if simply adding the Squeeze
repository to my sources.lst and running an 'aptitude
upgrade/safe-upgrade/full-upgrade will just upgrade Lenny - Squeeze
without me having to rebuild the system from scratch?
In my experience: you're likely to run
I may be wrong here, but I think you can only register once. The last
registration received will overwrite the first one. You will need to
specify a second entry and register that one separately. This is the
same reason you cannot register two devices to the same extension.
Yes, that's
are thinking of combining DSL +
DSL + Cable ISP on the same box and have our USA box send traffic to all 3
IPS.
Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sherwood
McGowan
Sent: Monday, December 27, 2010 4:52
;dateformat=%F %T.%3q ; with milliseconds
Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba
Sent: Monday, December 27, 2010 5:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Need some advise or paid help on running asterisk on two WAN connection. I
need load balancing and failover support.
WAN: 1 DSL + 1 Cable ISP.
Dave
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Server will have two fix public ips.
Dave
Original Message
Subject: Re: [asterisk-users] load balance with 2 wan connections
From: Alejandro Imass a...@p2ee.org
Date: Sat, December 25, 2010 1:58 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk
peer found
[2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830
handle_request_register: Registration from '7002 sip:7...@x.x.x.x'
failed for '38.108.40.94' - No matching peer found
[2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830
handle_request_register: Registration from '7002 sip:70
Dave
: Re: [asterisk-users] sip attack.. fail2ban not stopping attack
Make sure you have
dateformat=%F %T
in logger.conf
On Sun, Dec 26, 2010 at 1:04 AM, Dave George dgeo...@teletoneinc.com
wrote:
My server is being attached all day and fail2ban is not stopping the
attack. I updated stamstamp
allow=gsm
any suggestions welcome.
Dave
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When I call from a mobile to mobile (both registered on OPENBTS) the correct
caller ID is passed. That is the callerid that I set in the callerid=
field.
When calling from openbts to the PSTN the config header is passed.
Thanks,
Dave
-Original Message-
From: asterisk-users-boun
=473520
disallow=all
allow=gsm
host=dynamic
dtmfmode=info
Thanks,
Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten
Göllner
Sent: Monday, December 13, 2010 4:44 AM
To: Asterisk Users Mailing List - Non
=473520
disallow=all
allow=gsm
host=dynamic
dtmfmode=info
Thanks,
Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten
Göllner
Sent: Monday, December 13, 2010 4:44 AM
To: Asterisk Users Mailing List - Non
:
[IMSI310410381554227]
canreinvite=no
type=peer
context=openbts
callerid=473520
disallow=all
allow=gsm
host=dynamic
dtmfmode=info
I use the following in extensions.conf to dial:
exten = _45.,1,Dial(SIP/${ext...@ss72)
Thanks,
Dave
On 08/12/10 17:53, Danny Nicholas wrote:
Thanks
Just my .02, but since you’re going to (quite possibly) have a long(ish)
timeout if internet connection or SIP provider is down, I would have an
AGI run in front of my dial that did a ping to verify internet and sip
provider
Different brand/model, but similar as they are both el cheapo,
entry-level headsets. I tried using them on a laptop, and I get
marginally better microphone output, even with its volume cranked all
the way up + automatic gain control enabled.
I guess those on-board soundcards by Realtek
I'm having the following problem when using a headset on XP
connected to an on-board Realtek soundcard on an AsusTek M2N68-AM Plus
motherboard:
- Using any sound recorder (Windows', Audacity, XLite), the level is
just too low when speaking at a conversational level, even with the
I know understand the latency due to the resending .. But if the link was
have a good speed internet, then resending will make a big latency?
Maybe this latency better than having a cutting voice?
Fundamentally, TCP's congestion-avoidance and loss-recovery logic simply
won't work well with
On 21/10/10 22:04, Hans Witvliet wrote:
For suse there is a precompiled version on the OBS (vitsoft)
Package search on the OBS shows nothing for 1.8.0 at all.
Perhaps you know where it is hidden.
Dave Cotton
On 22/10/10 11:05, Hans Witvliet wrote:
On Fri, 2010-10-22 at 09:20 +0200, Dave Cotton wrote:
On 21/10/10 22:04, Hans Witvliet wrote:
For suse there is a precompiled version on the OBS (vitsoft)
Package search on the OBS shows nothing for 1.8.0 at all.
Perhaps you know where it is hidden
Just done a clean install of rc5 on a totally new machine and found the
following:-
/etc/init.d/asterisk start
errors on line 109 - there is no 0 before $VERBOSITY as in the other lines.
More interesting is that after make samples I have no iax2 available.
Dave Cotton
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