hello,
can anyone help me? im gettitng this error when i
tried runnin make on solaris 9
rm -f include/asterisk/version.h.tmpmake[1]:
`ast_expr.a' is up to date.make[1]: Leaving directory
`/export/home/fst/chris/cvs/asterisk'gcc -g -o asterisk io.o
sched.o logger.o frame.o loader.o
I have been wanting something similar. I paid some money for a busy
detect routine from newman telecom, but it is not yet done.
We'll see what happens.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Min Hwan
Chang
Sent: Tuesday, August 09, 2005
I'm getting messages like this:
NOTICE[18552]: Scheduled event in 0
ms?
Has anyone gotten these errors before? I'm
using the get data command to try to receive input, and I'm sometimes getting
this error message.
There are also times when asterisk doesn't
recognize dtmf input and instead
Guys.
How and which tools to use to load test an asterisk install? Say for
example, you need to see how many calls can be routed thru before losing
quality and making the cpu jump to the roof?
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According to the CIA world factbook there are 800 million landlines in
use and about 6.4 billion people. This makes more sense than 800
billion. there are probably at least an equal number of cellular
telephones in use as well, but i have no idea how one would go about
getting those numbers
Joseph wrote:
On Wed, 2005-08-10 at 08:10 +0400, Jean-Michel Hiver wrote:
I do up to 10 g.729 channels over 1024 / 256 DSL without noticeable
difference in call statistics (i.e. avg length of calls). If you are
using ADSL, the maximum bandwith you'll be able to use is your upload
rate
Hi,
this SETUP message does not contain a CalledParty IE. That means your
telco does not send you the DID. You will probably get ripped off extra
for that feature by your telco.
best regards
Klaus
--
Klaus-Peter Junghanns
On Tue, 2005-08-09 at 17:20 -0500, Panitaxx wrote:
Hi,
thanks for
First, the Asterisk settings:
- sip.conf -
[general]
port=5060; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)
context=default; Default context for incoming calls
disallow=all;
Hi all,
i'm using Asterisk with several extensions with 7 PSTN lines. Is possible, for
a caller, to dial directly an extensions ? For example, dial something like
[PSTN number]*[ext number] ?
Thanks !
--
O-Zone ! No (C) 2005
www.zerozone.it
___
Hi,
I'm want to do something slighty different with call queues than the
config allows...
I wish to have things work in an 'overflowish' manner. Ie, it works just
like 'roundrobin', where it rings
on one phone, no answer, rings on the next etc etc, except I want it to
keep ringing on all
Hi there
I am in the process of setting up a production Asterisk
server, which will mainly be used for meetme conferencing. I am considering
running a firewall, but wondering whether this will slow Asterisk down if all
packets are being scanned. Any ideas?
Many thanks
Steven
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com ---BeginMessage---
Hi
When I call from 1 RTC Client to another without
Asterisk everything use to be fine but when asterisk
is there as a
__
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Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com ---BeginMessage---
Hi
When I call from 1 RTC Client to another without
Asterisk everything use to be fine but when asterisk
is there as a
I am running the latest CVS version of Asterisk.
Calls between an IAX client and SIP phones (Grandstream SP2000 and
Sipura SPA-841) works fine and so do external call over the Internet
from the SIP desk phones.
However when I call from either the Grandstream/Sipura phones to another
one I get no
There is no called party ie but sending complete ie included in the
setup message. Hence, it tries to terminate.
Best regards
Hans
Paul Belanger schrieb:
Where are your calls coming from? Are you connected to the Telco or PBX?
PB
Panitaxx wrote:
Hi,
thanks for your response. here is the
Hi
I want Queue Application not to call those agents who
are busy talking
is it possible ?
Thanks
Start your day with Yahoo! - make it your home page
http://www.yahoo.com/r/hs
Hello ,
I have a question which i am not clear that whether it is possible or
not so i want some help to clearify Sorry for very long mail:
we have eight asterisk servers across different cities connected
through IAX intenet connection is DSL broadband so for sake simplicity
and easiness for eight
Hi all,
Can someone help with with Asterisk, SER, and Asterisks Queues?
I have three servers:
Server A: Asterisk with TE410 connected to PSTN
Server B: Asterisk connected to Server A via IAX2 trunk
Server C: SER where SIP agents register/connect to
What I wanted to do is configure Server A
hi,
are there ne asterisk system vendors in india???
-ankit
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Hello all asterisk users!
Question: Does anybody know about any good USB handset that would
understand SIP and Asterisk and will run with Linux?
I have found tons of them, but they are mainly only supported in Windows
environment.
I would like to set up new phone system in our company that
Contact Dristi soft, http://www.drishti-soft.com
Thanks Regards Ritesh Jalan Senior Engineer - Test
Audit Net4India Ltd. D-25, Sector -3Noida - 201301Tel:
+91-120-5323500-29 ,Extension-220Fax: +91-120-5323520Cell :
+91-9818616329Web site: http://www.net4india.com
On Tuesday 09 August 2005 18:56, Min Hwan Chang wrote:
I'm having this problem where if the phone is ringing from
IncomingCall #1, IC#2 will be immediately sent to VM. Is there
somethign wrong with my dial plan? I currently have 4 incoming lines
going into a TDM400 with the group set to g0.
Any network device (ie: switch, router, firewall) will add a small amount of
latency. To test the latency your firewall adds, you could simply try to do
a ping www.google.com, directly in front and behind the firewall, and look
at the ms response times.
Cheers,
S.
Although I'm not 100% sure, I believe the queue app never calls a busy
agent.
The problem you might have is that your phone(s) have their ability to
receive multiple calls (on snom's it's called Call Waiting Indication)
enabled. If this feature is enabled, from the viewpoint of asterisk your
Technically, yes it will. The question should really be is the latency
tolerable?
I run (as do many folks here) mine from behind a Linksys firewall at
home and I find the latency increase to be acceptable. I use Broadvoice
as a provider and have 3 other extensions connected to it from remote
Hello,
I would like to know how can the user be asked for a pin number instead what
appears to be a Serial number. And secondly, how do I ensure that as soon
as the phone is used it asks for a card to be entered. The default has it
you need to dial 77#.
Thanks
Marios
--
No virus found in
hi,
can you recommend some pocket pc sip client with iLBC or G729?
i'm googling over a hour and found nothing
---
Marek Cervenka
===
___
Asterisk-Users mailing list
You could put the phone into a context that would only allow numbers
beginning with 77#.
As for the request for a pin rather than a serial number, isn't that
just a voice file?
Mark
Dr. Marios Moutzouris wrote:
Hello,
I would like to know how can the user be asked for a pin number instead
The Xten softphone does GSM. Why won't that do?
http://www.xten.com/index.php?menu=productssmenu=xproppc
marek cervenka wrote:
hi,
can you recommend some pocket pc sip client with iLBC or G729?
i'm googling over a hour and found nothing
---
Marek Cervenka
Thanks Mark, how do I put phone into a context that would only allow numbers
beginning with 77#?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips
Sent: Wednesday, August 10, 2005 3:57 PM
To: Asterisk Users Mailing List - Non-Commercial
I encountered a similar problem with CVS-HEAD and sip2sip calls
between our Polycom IP500s. I attempted to diagnose the problem and
there are a few patches on mantis, but none of them worked for me. I
flipped back to stable and have had no problems since.
Anyone got any ideas?
--
Tom
On
Christian wrote:
Hi,
I'm recording wave files but I cant get Asterisk to play them, only if
they are in 8000 Hz. What is the maximum sample rate Asterisk can
handle? I have been using 16-bit 44.1, 22050 and finally 8000 kHz.
To my knowledge, the only correct format for wav files to use in
Moises Silva [EMAIL PROTECTED] writes:
make sure you have the next line in /etc/asterisk/modules.conf
load = app_dial.so
Not only that, but be sure to have a sound system loaded in the
modules.conf files.
--
Esben Stien is [EMAIL PROTECTED] s a
http://www. s
I assume that its a SIP based phone.
in sip.conf do
[phone-number-of-payphone]
all the usual stuff = all the usual stuff
context = astcc-context
in extensions.conf do
[astcc-context]
exten = the astcc stuff
Hope this helps. If not I am fairly reasonable. +1 973 828 1625
Mark
Dr. Marios
Good day all
How do I get h323 and video working?
--
Thanks
Altus Snyman
Stormcorp Network Solutions
+27 11 8071141 exten 301
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To
I did try installing the 1.0.9 version but I have the same problem with
that release aswell.
On Wed, 2005-08-10 at 14:14, Tom Hayden wrote:
I encountered a similar problem with CVS-HEAD and sip2sip calls
between our Polycom IP500s. I attempted to diagnose the problem and
there are a few
Christian [EMAIL PROTECTED] writes:
only if they are in 8000 Hz
Asterisk only handles 8kHz, or so I understand. There is some work to
get it to do more. Would be very nice for us that only use asterisk
for voip between softphones and have no interest in hardware or
telephone networks what so
On Wednesday 10 August 2005 16:43, Esben Stien wrote:
Moises Silva [EMAIL PROTECTED] writes:
make sure you have the next line in /etc/asterisk/modules.conf
load = app_dial.so
Not only that, but be sure to have a sound system loaded in the
modules.conf files.
there were actually multiple
With a liberal application of RFTW
altus wrote:
Good day all
How do I get h323 and video working?
--
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
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RFTW or RTFM
On Wed, 2005-08-10 at 09:36 -0400, Mark Phillips wrote:
With a liberal application of RFTW
altus wrote:
Good day all
How do I get h323 and video working?
--
Thanks
Altus Snyman
Stormcorp Network Solutions
+27 11 8071141 exten 301
Been having problems with CVS lately. I get incoming calls from an
iaxcomm user, using a windows system. Asterisk stops sending data
after about 30 seconds. I view this with tcpdump on the same
computer. I still receive data and can hear the remote party.
This problem starting sometime mid last
Does anyone know when Asterisk is supposed to support prioritization of
the SRV records returned. I think it's accepted that right now the
Asterisk always just uses the first record returned regardless of
priority.
I also noticed, on register, that it queries for an A record first. If
it gets a
On Wed, 10 Aug 2005 08:59:59 -0400, Mark Phillips wrote:
The Xten softphone does GSM. Why won't that do?
http://www.xten.com/index.php?menu=productssmenu=xproppc
marek cervenka wrote:
hi,
can you recommend some pocket pc sip client with iLBC or G729?
i'm googling over a hour and found
Michael Lunsford wrote:
Does anyone know when Asterisk is supposed to support prioritization of
the SRV records returned. I think it's accepted that right now the
Asterisk always just uses the first record returned regardless of
priority.
I wonder if this is fixed in CVS-HEAD? I guess I could
Then perhaps you have a NAT problem or some other issue.
--
Tom
On 8/10/05, Gareth Blades [EMAIL PROTECTED] wrote:
I did try installing the 1.0.9 version but I have the same problem with
that release aswell.
On Wed, 2005-08-10 at 14:14, Tom Hayden wrote:
I encountered a similar problem
Or you could use the local channel and branch it off to as many
contexts as needed with different contexts that use different times
for the dial.
On 8/9/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On 8/10/2005, Irakli Natsvlishvili
[EMAIL PROTECTED] wrote:
Hello,
I know that using
Both phones are on our internal network so not a NAT issue.
However your email did prompt me to check iptables as I have rebooted
the machine since it last worked. Dropping the firewall has fixed the
fault so it looks like I will need to have a look at the ruleset.
Thanks
On Wed, 2005-08-10 at
On 7/27/05, Paul Traue, Jr. [EMAIL PROTECTED] wrote:
I'm experiencing rather severe problems with 1.0.9 (we've had to backrev
to our last version we know works (1.0.5).
We are running a single PRI line with a T100P card. After about 10
hours of asterisk running and the modules loaded we
I have installed a new TE205P in my asterisk server. When I reboot the
box the error ZT_SPANCONFIG failed on span 1: No such device or
address (6)
When I modprobe wct2xxp and run ztcfg -vvv loads everything
correctly. So it seems that it is not loading wct2xxp at boot.
I then added the
Ok, but thats static routing. My architecture is this:
[pbx extensions] --- [SIEMENS PBX] [ASTERISK] --- [SER] --- [sip
clients]
I can't put in Asterisks sip.conf the hundreds of pbx extensions (and
they are always changing), I must do a dinamic forward for all 74XXX calls.
I think
Not the Pocket PC version
Michael Graves wrote:
On Wed, 10 Aug 2005 08:59:59 -0400, Mark Phillips wrote:
The Xten softphone does GSM. Why won't that do?
http://www.xten.com/index.php?menu=productssmenu=xproppc
marek cervenka wrote:
hi,
can you recommend some pocket pc sip client with
On Wed, 2005-08-10 at 15:51 +0100, Joao Pereira wrote:
exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r)
Just an observation that you have an invalid address there; you have
1193 instead of 193 I believe. Fix this and I see no reason for your
problem to remain.
--
-Bryce
[EMAIL PROTECTED]
But to have a transparent integration with VoIP and legacy, I cant make
users dial twice... or having to whait for Asterisks dialtone, and dial
the number.
I whant to dial the 74XXX from a PBX extension (74118 for example) and
the IP phone rings.
Asterisk just need to forward the 74XXX calls,
Im having some problems with connecting a TNT to asterisk. The problem
is
when the call is sent to asterisk and signaling is done the RTP syncs
however
no audio is produced. Can someone give me some idea of how to
accomplish this?
I am using the standard configs and g711 and 729 do the same.
On Wed, 10 Aug 2005 10:54:20 +0200, O-Zone\ wrote:
Hi all,
i'm using Asterisk with several extensions with 7 PSTN lines. Is
possible, for a caller, to dial directly an extensions ? For
example, dial something like [PSTN number]*[ext number] ?
Thanks !
Nope.
Unless * answers the call and
I am trying to create a system as follows:
incoming call -- ivr -- sent to dummy extension 1000 -- redirects
user to ring group 2000 -- (ring group consists of extension 3000
and 4000) -- no answer -- sends to voicemailbox 1000
I want to do this to be able to have one extension number (and
Asterisk has been working fine for me for several weeks using MGCP to a Adit600 for intra office calling. I have recently loaded h323 and the following errors occurs when starting asterisk.
[chan_h323.so]Aug 10 09:09:18 WARNING[7824]: loader.c:258 ast_load_resource:
Hello everyone,
I have just noticed a fairly obvious feature that it looks like many
people have been looking for...
If you have a queue defined with strategy=ringall, members of the queue
will still get incoming calls when they are already on a call (call
waiting). The only solution
yes, I know, in my extensions.conf is writen correctly.
Thanks
Joao
Bryce Chidester wrote:
On Wed, 2005-08-10 at 15:51 +0100, Joao Pereira wrote:
exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r)
Just an observation that you have an invalid address there; you have
1193 instead of
does libpt_linux_x86_r.so.1.5.2 exist on your machine?
maybe try running ldconfig or if that file is in a non-standard
location, maybe add that path to ld.so.conf and then run ldconfig again
On Wed, 2005-08-10 at 08:09, kurt turner wrote:
Asterisk has been working fine for me for
Another solution could be to assign each person a POTS line. When the
POTS line rings route it directly to their desk.
Mark
Niklas Larsson wrote:
On Wed, 10 Aug 2005 10:54:20 +0200, O-Zone\ wrote:
Hi all,
i'm using Asterisk with several extensions with 7 PSTN lines. Is
possible, for a
For ZAP cards, you can tell Asterisk to answer calls immediately across
trunks. Does CAPI have the same type of setting? I am not familiar with
Asterisk and CAPI so I am not sure of the options.
In Zapata.conf, setting immediate=yes will make the call drop into the 's'
extension of the
Chris,
The problem is that your compiler can't find a library called
libcrypt.so.0.9.7. This library is apparently needed by
libssl.so. These are both runtime, shared libraries. The
result is that you end up with undefined symbols (probably
variables used in services the libraries provide). You
On Wed, 2005-08-10 at 00:36 -0500, Dave Redmore wrote:
Thanks to Jean-Michel for the info. re: getting 10 calls over a 1024/256
ADSL using g729... Just the sort of info. I was looking for... Anyone
else?
I hate GSM - sounds horrible to me... but, iLBC sounds pretty good and
I think
yes.. i have the following
IPD:/usr/local/lib# ls firmware libpt_linux_x86_d.so.1.5 libpt_linux_x86_r.so.1 python2.3libpt_linux_x86_d.so libpt_linux_x86_d.so.1.5.2 libpt_linux_x86_r.so.1.5libpt_linux_x86_d.so.1 libpt_linux_x86_r.so libpt_linux_x86_r.so.1.5.2
I found ldconfig under root
I have looked at the wiki and the list archives and I haven't found
anything specifically addressing this problem, so here goes:
We are using asterisk voicemail to page users (on-call personnel) when a
message is received. However, we only have numeric pagers, not
alpha-numeric. If I send an
On 08:45, Wed 10 Aug 05, Jean-Michel Hiver wrote:
1) your provider is voluntarily screwing up VoIP traffic
2) some idiot purposingly fills up your pipe with UDP traffic
If they fill the pipe with TCP traffic, UDP will be dead as
well. Protocols don't matter, bandwidth does.
--
Michiel van
Ondrej Valousek wrote:
Hello all asterisk users!
Question: Does anybody know about any good USB handset that would
understand SIP and Asterisk and will run with Linux?
USB Phones don't understand anything. They are effectively four components:
a) Microphone
b) Speaker
Title: RE: Info / recommendation on either Audiocodes or Vegastream gateways
I am looking for how to information / references on use of either Audiocodes MP104 or 108, or Vega 50 Gateways for interconecting Asterisk to the PSTN via FX0 interfaces.
Any info of references / personnal
Joseph [EMAIL PROTECTED] wrote:
On Wed, 2005-08-10 at 08:10 +0400, Jean-Michel Hiver wrote:
I do up to 10 g.729 channels over 1024 / 256 DSL without noticeable
difference in call statistics (i.e. avg length of calls). If you are
using ADSL, the maximum bandwith you'll be able to use is your
That should not be a problem. My users conference
using a voip line from an ITSP so at any time there may be 4-8 calls passing
over the firewall and terminating in the MeetMe conference. It works
great. I would recommend Pix BTW. Linksys would be my next
rec. But hey, they are both Cisco
Joseph [EMAIL PROTECTED] wrote:
Don't forget to experiment with nice to increase priority of for
Asterisk.
By default asterisk run with priority 0 same as apache and any other
applications.
We run a web-server on the same machine as asterisk and increasing
nice for Asterisk to -15 helped a
Everything seems to be working. The admin seems to works, the express
talk software on my pc works but I can't get my ip phone to make calls.
They can accept calls but can't make them. my log file only has this error.
Aug 10 12:10:01 DEBUG[4159]: Setting NAT on RTP to 0
Aug 10 12:10:01
just my .02
2 or 3 shabby DSL connections and use asterisk to monitor the quality of each
connection
then route calls according to the best option at any given time
I really think this is going to be a false economy would compared to the cost
of an SDSL line with
an SLA. Also network issues
or you could have a tel (or DDI) number for each internal extension giving each
1 a real tel number.
__
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we got this installation :
WinSip(demo version) - ser(radius accounting) - asterisk(from sip to
h323 channel) - gsm gateway(with 32 sims in it)
we configured winsip to make 28 calls like from 28 different sip
accounts, to 28 different cellular phones numbers
after the first ten :
--
On Wednesday 10 August 2005 04:39 am, Ondrej Valousek wrote:
Hello all asterisk users!
Question: Does anybody know about any good USB handset that would
understand SIP and Asterisk and will run with Linux?
I have found tons of them, but they are mainly only supported in Windows
environment.
Storm D. J. Petersen wrote:
Any network device (ie: switch, router, firewall) will add a small amount of
latency. To test the latency your firewall adds, you could simply try to do
a ping www.google.com, directly in front and behind the firewall, and look
at the ms response times.
Cheers,
S.
Elwin Andriol wrote:
Although I'm not 100% sure, I believe the queue app never calls a busy
agent.
The problem you might have is that your phone(s) have their ability to
receive multiple calls (on snom's it's called Call Waiting Indication)
enabled. If this feature is enabled, from the
Hi
I am trying to find a possible cause of 'audio fading in and out' which effects
about 0.1% to
0.15% of calls placed to a voip provider for termination. (I am using SIP
alaw throughout in
this case)
I don't believe this to be something that is network / jitter etc related (?)
and looking
ok, they let me know I'm an idiot, maybe
outboundMax=10
has something to do with it
after the first ten :
-- Executing Dial(SIP/5060-081925b0,
OH323/[EMAIL PROTECTED]) in new stack
-- H.323 call to [EMAIL PROTECTED] with codec(s) alaw
-- Called [EMAIL PROTECTED]
we get :
Ok, I m getting to the point,
This route:
exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r)
Isn't working because the dialed number isnt maching _74XXX
I putted Asterisk in capi debug mode and when I dial 74118 he says:
gnugk*CLI capi debug
CAPI Debugging Enabled
-- CONNECT_IND ID=001
That's a good question. I have no idea. I'm pretty new at this, so I'm
just combining bits and pieces of what I find together. If anyone
could help, it'd be greatly appreciated.
On 8/10/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Tuesday 09 August 2005 18:56, Min Hwan Chang wrote:
I'm
At 9:49 AM -0400 on 8/10/05, Michael Lunsford wrote:
Does anyone know when Asterisk is supposed to support prioritization of
the SRV records returned. I think it's accepted that right now the
Asterisk always just uses the first record returned regardless of
priority.
I also noticed, on register,
Original Message
Subject: RE: [Asterisk-Users] will a firewall slow down asterisk?
From: Wiley Siler [EMAIL PROTECTED]
Date: Wed, August 10, 2005 11:04 am
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
That should not
I've got a Cisco 7960 + 7914 working with one registration and one line.
I want to use the 7914 as a busy lamp field for the rest of my
extensions which are all SIP. Is this possible?
Craig Bruenderman
Network Advocates, Inc.
300 Envoy Circle
Suite 300
Louisville, KY 40299
Main: 502-412-1050
Someone can help me
in the configurationbetween Panasonic TDA200 and Asterisk viaTE110P
and Panasonic E1-PRI Card???
I get the
connection, but the problem is : From Panasonic extensions , cannot make calls
to SIP Phones???.
From SIPI can
make calls to Panasonic extensions ,and PSTN .
OK, I have a Asterisk @ Home 1.0.7 server with two Digium TDM400 cards, one 4
port FXO and one 4 port FXS. When I plug an analog cordless phone into the
TDM40B card and setup a ZAP extension, the phone rings in and you can answer
just fine. The weird part is when you try to get dialtone from
On Wednesday 10 August 2005 12:48, Min Hwan Chang wrote:
That's a good question. I have no idea. I'm pretty new at this, so I'm
just combining bits and pieces of what I find together. If anyone
could help, it'd be greatly appreciated.
Have you read through the Asterisk Handbook draft?
On Wed, 2005-08-10 at 12:57 -0400, Craig Bruenderman wrote:
I've got a Cisco 7960 + 7914 working with one registration and one line.
I want to use the 7914 as a busy lamp field for the rest of my
extensions which are all SIP. Is this possible?
Yes, you can do this.
You have to have your hints
I have a really baffling problem.
A couple of months ago I purchased a pair of GrandStream GSX-2000 phones for
use with Asterisk.
At first all was well. But recently I've noticed terrible sound quality
problems. Basically the sound will glitch or stutter randomly from time to
time.
Now, what is
When running ldconfig -v, did you see it find the files under the
directory /usr/local/lib?
If not, edit /etc/ld.so.conf with your favorite editor and add
/usr/local/lib in a new line.
Then rerun ldconfig -v.
Check that the libpt* files were seen.
kurt turner wrote:
yes.. i have the
Greetings all!
I bought some Polycom 501s and got them installed, configured
and running. I'm using bootrom 2.6.2 and sip 1.5.2. My problem is that
when I press the Do Not Disturb button, the phone stops responding and
reboots by itself eventually. Has anyone seen this problem before?
Jesus
I have a TE410P with two real Telco T1's and the
other 2 portsterminate into an in-house voice mail/IVR system. Calls
arrive from Telco are routed to the appropriate in-house system based on the DID
Digits. This part works perfectly.
Now I what to allow the in-house VMS to dial though
our asterisk is configured to retrieve sippeers and iaxpeers via odbc
from a mysql database. after each call show processlist; within the
mysql console shows 2 more persistent connections which are showing no
further activity and will not go away even after restaring asterisk.
is anybody else
How did you set up the zaptel cards originally? Have you run
genzaptelconf? Did you remember to plug in a power adapter to the TDM
card?
Also, to clarify, if you plug a cordless phone into port1 of the FXS
card and pick up the phone to dial, all of the other ports on the FXS
card ring? This is
Hi, Everybody
I had installed Asterisk successfully with Beronet BRI
cards, if any body needs help in installing then you
can contact me. I feels good of helping you in its
installation.
Lokesh
Portugal
mail at [EMAIL PROTECTED]
I've got it working, but I'm having random echo issues with the TNT.
What TAOS are you running on the TNT ?
Which ethernet card are you using?
When you changed from the default 323 signalling to sip (assuming you
did) did you reboot the TNT ?
W. Kevin Hunt
CCIE #11841
MCSE, Linux+ SME
Is asterisk 2.0 real? Running in c#? I see references to
it but cannot find it anywhere.
---
Justin
Selleck
NetworkEngineer
Smooth Fusion, Inc.
http://www.smoothfusion.com
t.
806.771.3873 x230
f. 806.771.2862
Is there a way to see what priorities are other processes using?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Kevin Walsh
|Sent: Miércoles, 10 de Agosto de 2005 11:08 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject:
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