Which version of Asterisk are you using?
According to http://www.voip-info.org/wiki/view/Asterisk%20T.38 unless you
are using Asterisk 10, there's quite some patching (or buying) you'll need
to be doing.
Alyed
2013/11/21 Bryant Zimmerman brya...@zktech.com
Can you funnel them through
Have you followed the instructions in:
https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway
and: http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway
??
If possible try with a different ATA since it seems not all of them work
fine with fax pass trough.
Alyed
2013/11/21 Damian
Think you only need to make sure you have in your sip.conf file these
configs:
[your-device-name]
.
.
disallow=all
allow=g729
.
.
Alyed
2013/11/20 Damian Gonzalez dgonza...@denwaip.com
Hello,
I have a problem with movistar in Mexico with a sip calls. Movistar send
to me
Please post one of your sip.conf phone configs, so we can have a look.
Alyed
2013/8/2 Carlos Chavez cur...@telecomabmex.com
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Hash: SHA1
On 8/1/13 9:17 PM, Michael L. Young wrote:
- Original Message -
From: Carlos Chavez cur...@telecomabmex.com
Thanks a lot for the link and the tip. Have been trying it these days and
think it wil work on my system.
Thanks again Shaun.
2012/11/8 Shaun Ruffell sruff...@digium.com
On Tue, Nov 06, 2012 at 06:49:09PM -0600, Alyed wrote:
Hello listers,
I'm trying to run DAHDI 1.4 on a 3.0 Debian
Hello listers,
I'm trying to run DAHDI 1.4 on a 3.0 Debian Kernel in an embedded system,
but have faced lots of problems mainly because it has lots of functions
looking for the PCI.
Have seen so many problems, I'm in fact thinking it cannot be possibly done
(at least not in a couple of weeks, by
Hi listers!
Have a problem with distortion in some analog lines. When some call comes in
from PSTN the sound is really distorte, nothing can be understanded, but
Internal calls work ok.
Funny thing is that when I start/stop asterisk,dahdi, and wanrouter services
eveything goes fine again. This
try it with _ in front of the *
exten = _**,1,.
Alyed
2010/6/4 Danny Nicholas da...@debsinc.com
Probably going to have to use read to detect this..
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com
to change te phone's IP, GW and mask parameters, have not yet a clue on how
to make it register with asterisk.
Has anyone out there got some experience dealing with something similar??
Thanks!
Alyed
--
_
-- Bandwidth
)
; long distance
I always use . and never had a problem.
Alyed
2010/5/27 Eddie Mikell ed...@rimmkaufman.com
All:
Yesterday I discovered something interesting. I dialed 1800ANCESTRY
from the asterisk system I am testing and got the number doesn't exist
message. I then dialed the same
Sorry for my mislead should have said I've never been able to with xlite
it's just with Sjphone it's straight forward.
Alyed
2010/4/19 bruce bruce bruceb...@gmail.com
That is not correct. It's possible by adding a display name and adding the
IP address of the pbx you are calling as the host
their
instructions for asterisk upgrade.
Alyed
2010/4/18 Tonty T ton...@gmail.com
You got him wrong.
He actually want to know the steps to upgrade to version 1.6.2 so he do can
a conference bridge using confbridge instead of of meetme because he does
not have dahdi installed.
He just want
You are right if going from 1.4.X to 1.6.2.X or similar that's the best, but
if not moving from revision, then I don't think you need to remake the
sample files.
Alyed
2010/4/19 Carlos Chavez cur...@telecomabmex.com
On Mon, 2010-04-19 at 11:19 -0500, Alyed wrote:
If that's the case what I
I guess what you meant, is you don't have a physical card to provide the
timing needed by Meetme. Then, if you are looking for dahdi to use kernel
timer, then you need not to upgrade Aterisk but Dahdi to 2.3.0
Alyed
2010/4/18 Thomas Perron thomas.per...@gmail.com
I read that I need to run
You can't do that with Xlite, try Sjphone instead.
Alyed
2010/4/17 bruce bruce bruceb...@gmail.com
Hi Guys,
Wondering if anyone has tried to make a direct SIP peer to peer call using
x-lite without any registrations of any sort. I can't seem to find the
setting.
Thanks,
bruce
list but maybe someone can tell if they
can help us here?
Alyed
2010/4/13 Randy R randulo2...@gmail.com
On Mon, Apr 12, 2010 at 7:17 PM, Darrick Hartman
dhart...@djhsolutions.com wrote:
That only addresses EC2 (and assumes that Amazon has any interest in
protecting their reputation). What
Have a look at:
http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication
It's about IAX but guess will give you some good hints on how to solve your
problem.
Alyed
2010/4/13 Mike Diehl mdi...@diehlnet.com
Hi all,
I'm trying to tighten things up a bit and I seem be be running
The context that I'm using for the local extensions is not [general].
Sorry quite didn't get what you mean. Nevertheless I I think it is a matter
of NAT/firewall management.
Alyed
2010/4/11 Daniel Bareiro daniel-lis...@gmx.net
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Hi, Alyed
know if it worked.
Alyed
2010/4/9 Ye Liu jaux...@gmail.com
Hello everyone,
I'm fairly new to asterisk and this list. Currently I'm working on IAX
trunks to send/receive calls between 2 asterisk boxes with asterisk
1.6.1.1+asterisk gui 2.0. After some work in the gui, two boxes can
send
[general] section.
Alyed
2010/4/10 Daniel Bareiro daniel-lis...@gmx.net
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Hi all!
I'm trying to test with a friend who has an Asterisk in his office with
the Asterisk which I have in my house. Then I have an extension that he
is trying
Sorry, the parameter should be.
srvlookup=yes
Alyed
2010/4/10 Alyed al...@vivoxie.com
Daniel, you are having a problem often seen in pre 1.4.14 versions.
Before this release srvlookup=no was the default for sip.conf and guess
the same for iax.conf . So if you are working with a previous
in the right place.
Pls look for them in the server you are actually having the problems with
cause I can't remember that sound file being on the official's asterisk
release.
Alyed
2010/3/30 Ott Rose sixfourimp...@hotmail.com
where are those sound files kept? i looked last night in
/var/lib
file (apparently
without errors) and then the next instruction is to hangup the call, hence
Asterisk hangs it up.
Just to be sure play this sound file independently.
Sorry but other than this there's little I can do, maybe someone else has
experience with this.
Alyed
2010/3/29 Ott Rose sixfourimp
!
Alyed
2010/3/28 Daniel Bareiro daniel-lis...@gmx.net
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Hi, Jim.
On Sun, 28 Mar 2010, Jim Dickenson wrote:
Make sure not to do make samples or you will overwrite your .conf
file
Yes I'm talking about Asterisk Now's GUI and yes, you can just install this
component.
google for Asterisk Gui 2.0 and you'll find plenty of info.
Regarding the DB I can't help you here, maybe someone else can.
Alyed
2010/3/28 Daniel Bareiro daniel-lis...@gmx.net
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/firewall won't just
close it.
try playing with qualifyfreq as well.
Let us know if it helped.
Alyed
2010/3/27 James Lamanna jlama...@gmail.com
Hi,
I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall.
After some period of time, asterisk says that some of them are
unreachable
much pressure on me or the listers :)
Alyed
2010/3/26 kamrun nahar bina bina...@gmail.com
Dear sir,
Thanks for your reply.
our memory size is 4GB.
concurrent calls no : 30.
Our memory condition is below :
Cpu(s): 0.3%us, 0.7%sy, 0.0%ni, 98.5%id, 0.0%wa, 0.1%hi, 0.3%si,
0.0%st
Just to check, have you set up
srvlookup=yes
under the general context in your sip.conf?
Alyed
2010/3/26 Luis Silva luis.si...@dreamware.pt
Hi again,
In other asterisk it happened the same... No internet, no justvoip
resolution, no sip...
Remove the trunk, sip up... I'm going to test
I guess to do what you want you need to dial directly between the phones.
Can't do it with xlite but you can with SJphones
Don't remember the exact syntax but guess it's something like
sip:usern...@the.phones.ip:5060
Alyed
2010/3/26 haloha haloha...@gmail.com
Hi all
my asterisk server, 2
everything possible to ensure that the DNS
lookups will not block for long periods of time.
Alyed
2010/3/26 Luis Silva luis.si...@dreamware.pt
Just to check, have you set up
srvlookup=yes
under the general context in your sip.conf?
Alyed
No, but I put it now but the result is the same
Seems like an Amportal configration problem not and Asterisk issue. Maybe
you should try in one of the FreePBX users list.
Alyed
2010/3/26 Ott Rose sixfourimp...@hotmail.com
i have posted this question couple of times and never really got any hits
i wasn't able to provide any debug info
(just restarting the service might not
work).
Alyed
2010/3/26 Ira i...@extrasensory.com
I get this when my brother in law tries to call in from his box to mine.
WARNING[4855]: chan_sip.c:12675 check_auth: username mismatch, have
100, digest has s
or after changing the register line
problems I suggest you
fix them instead of asking everyone to call using SIP uri.
Alyed
2010/3/26 haloha haloha...@gmail.com
Hi Alyed
xilte softphone work perfectly on other sip server(opensips server)
Don't remember the exact syntax but guess it's something like
sip:usern
it
will play a little with the SDP part of the SIP. Have a look at
http://www.voiptraversal.com/ice_methodology.htm to better understand what's
ICE about.
Alyed
2010/3/26 haloha haloha...@gmail.com
Hi Alyed
so the asterisk is in middle in all version, right? thank you for your
explanation
? 2 GB in RAM seems little against 600
registered agents.
Alyed
2010/3/25 kamrun nahar bina bina...@gmail.com
Dear sir,
We have been using asterisk for 4 years. Now we have got problems which
occurs during the attended transfer.
But we are not always getting this problem. Sometimes
Zap channels than your SIP
connection.
First try playing a sound in B when receiving the call, that way you can be
sure the connection is ok. If that one works then move to PSTN.
Alyed
2010/3/25 Aaron chen evane1...@gmail.com
i have a prablom here,
i want to send a call from A to B use sip
Guess it's not a matter of asterisk as it is of Linux scripting.
first check: http://linuxproblem.org/art_9.html
then try something like:
http://forums.digitalpoint.com/showthread.php?t=70926
but as Steve said, why you need to restart the asterisk service in the first
place
Fix what's wrong
If it is only a version move, say from asterisk 1.6.1 to 1.6.1.X it's
generally ok, but be careful since there are some changes that might hit you
from 1.6.0 to 1.6.2 need to have a read into the change log before changing
versions, same for the other packages you mention.
Alyed
2010/3/24 Ott
Don't be so hard in him/her we all make mistakes, let's just learn from them
and move on.
Alyed
2010/3/24 Ott Rose sixfourimp...@hotmail.com
thanks for hijacking my thread.
i have an idea don't help him/her so that people will help me!
now i am going to re-post this.
Date: Wed, 24
Try the same as in
http://lists.digium.com/pipermail/asterisk-users/2010-March/246316.html
just make sure to add this in the [channels] context ;)
Hope it helps.
Alyed
2010/3/23 Zhang Shukun bit...@gmail.com
hi, all
i use Queue() to call a Mobile phone, there is only one mobile phone
Make sure you have
busydetect=yes
busycount=3
somewhere below your [general] context in chan_dahdi.conf (or zapata.conf
depending on your asterisk version) and restart the the service.
This should be enoough to do the magic.
Alyed
2010/3/21 Daniel Bareiro daniel-lis...@gmx.net
-BEGIN
serious problem if you don't fix it. So don't
forget to include this parameters from now on. I have played with them and
found setting busycount=5 is not very efficent, so leave it to 3 or 4 at
most.
Good to hear your problem is solved.
Alyed
2010/3/22 Daniel Bareiro daniel-lis...@gmx.net
-BEGIN
The error lies here:
make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
make: *** arch/i386/boot: No such file or directory. Stop.
do you have the kernel-headers installed? (e.g.
glibc-kernheaders-2.4-9.1.87.i386.rpm for Fedora)
Alyed
promise NOT to use it for
Telemarketing, otherwise might the mighty spirits of (place the name of
some super natural power here) cast the most terrible spell on you
forever and ever.
so, if you still want it just contact me directly :)
Alyed
Return-Path
Had the same issue time ago, but Eric shed good light on it,
have a look at:
http://lists.digium.com/pipermail/asterisk-users/2006-November/172079.html
Summary: sorry, no nice work around.
Alyed
Return-Path: [EMAIL PROTECTED
of looking around I decided to re-compile
wanpipe (run ./Setup install where you untared the
wanpipe-X.X.tar.gz) and afterwards it worked fine.
if it doesn't try contacting me directly, maybe I can have an eye on your
system and help you out.
Alyed
Return
is happening: as soon as the telco makes a reset
of the trunk they say they start receving data as if the PBX would be
generating calls, then all channels go sealed except for 1.
Anyone having an idea on how to solve this?
Alyed
___
--Bandwidth
First look if you have the libidn.so.11 library. if you don't
then install it, otherwise you can simply copy-paste it into the /usr/lib
folder where Asterisk is looking for it or make a symbolic link to it.
Alyed
Return-Path: [EMAIL
as it
went
till I picked up).
Have tried kewlstart, loopstart, groundstart and even the
answeronpolarityswitch configs in zapata.conf but can't find the
solution.
Any one having solved this problem?Alyed
___
--Bandwidth and Colocation provided
as it went
till I picked up).
Have tried kewlstart, loopstart, groundstart and even the
answeronpolarityswitch configs in zapata.conf but can't find the
solution.
Any one having solved this problem?Alyed
___
--Bandwidth and Colocation provided
NATs. It is just because of how SIP and NAT work together.
Alyed
Return-Path: [EMAIL PROTECTED] Tue Nov 07 09:16:25 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; Tue, 7 Nov 2006 09:16:25 -0700Received
My advice is to first make some tests to see if a reload is enough for Asterisk to read any group definitions change in zapata.conf, otherwise no on-the-fly change will workAlyed
Return-Path: [EMAIL PROTECTED] Mon Oct 30 13:23:36 2006Received: from
Isn't your problem more about NAT traversal rather than the phones themselves?if so better use some iax softphone, have a look at: http://www.voip-info.org/wiki-VOIP+Phonesof course you can use SIP based hard/soft phones but using iax based ones is cheaper and faster.Alyed
Echo is generated by the analog end to where you place the call, not the IP side of it.
As far as I know the echo cancelation in the Asterisk can only be tweaked in the zapata.conf (since IP calls don't generate it)
I'm afraid there is little you can do to here.Alyed
you can also try using
busydetect=yes
busycount=4
in your zapata.conf
Hopefuly you won't start getting sudden hang ups, due to false positives and it will be helpful enough.
Alyed
Return-Path: [EMAIL PROTECTED] Tue Oct 17 14:30:11 2006Received: from digium-69-16-138-164
calling search space, and then give them access to
each other you can do that.In the attachment, I circled the calling search space field I see on my Add NEW SIP TRUNK PAGE.Hope this helps. -- Original message --From: "Alyed Tzompa" Many thanks for
anyone had a similar problem??? I'm not a Cisco expert so
dun't know if I need to "enable" SIP messageing/reception in the Cisco.
Regards,
Alyed
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users ma
risk trunk on the Call manager is in the same "calling
Search Space" as the phones are in, or make sure there is access
between the "calling search spaces"-Eric -- Original message --From: "Alyed Tzompa" Hi! I'm trying to communicate a C
What I want is to transfer some calls to a Cisco extension, so think I don't need to do the upgrade to CM5.I'm I right?AlyedOn Tue, Oct 10, 2006 at 01:21:28PM -0500, Lacy Moore - Aspendora wrote: I'm begining to think this is more of a Cisco config problem than Asterisk, has anyone had a
Be careful when using heavily ChanSpy. We
did couple of weeks ago and the result was having Asterisk crashing
almost once every day. How heavy? around 4 people using it 8 hours a
day, each one using ChanSpy every 3-5 mins.
we were not able to find the exact reason, so just stop using
I'm experiencing the same problems, but
unfortunatelly haven't been able to associate them with any number
since they appear to be random. But maybe we can do a little research
about it, and hopefully find teh solution for both:
are your PSTN lines POTS or E1/T1? can you make a couple of
I'm curious... why will this work??
busydetect will just cut the line if there are 4 tones (les or more
depending the busycount param), and call progress will in fact try not
to cut the call due to false hangups.Alyed
Return-Path: [EMAIL PROTECTED] Wed Sep 27 16:12:13
appearance, end-user feedback, any infowill be appreciated.thnx!Alyed
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quality, visual appearance, end-user feedback, any info
will be appreciated.
thnx!
Alyed
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Sorry but I've ran out of ideas...Anyone else out there with a successful Polycom g729 pass through-only experience?Alyed
Return-Path: [EMAIL PROTECTED] Thu Sep 21 11:27:21 2006Received: from nz-out-0102.google.com [64.233.162.206] by maila11.webcontrolcenter.com with SMTP
ed to have the proper transcoder,
as it does not, then the error arises... at least that's what I think :)
set "canreinvite=yes" (or just comment it since that's the default) on both parties and try again.
Let me know if it works.
Alyed
Return-Path: [EMAIL PROTECTED]
Since the phone is the one behind a NAT,
and the registration is done only with SIP packages, setting or not the
"nat" is not an issue (ONLY for registration purposes). You can see
this since Asterisk is receiving the registration. Why is it denying
it?... wel, that's something that will
it.
Alyed
Return-Path: [EMAIL PROTECTED] Wed Sep 20 18:27:45 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; Wed, 20 Sep 2006 18:27:45 -0700
Good Day,I
am new to Asterisk and I need help in configuring
Make sure the codec used by the Polycom
will be only g729 via the phone's web interface, as far as I remember
Polycom will try always to use ulaw or alaw first unless it is
configured to use only or as first choice the g729 codec.Alyed
Return-Path: [EMAIL PROTECTED] Tue
concurrent calls constantly
Any ideas?
Many thanx!Alyed
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your phones set with nat=yes and qualify =yes, will not
affect the behaviour of your phones if your network is not really full,
but will be a bad and dirty way to do it :)Alyed
Return-Path: [EMAIL PROTECTED] Tue Aug 01 08:35:05 2006Received: from digium-69-16-138-164.phx1
Could you please explain what the network configuration you want to try? it would be really helpful.
you can be as simple as: SIPphone-- internet -- NAT-- asterisk
or whatever your particular scenario is.Alyed
Return-Path: [EMAIL PROTECTED] Mon Jul 31 11:43:16
it will. Anyway you can ask this directly to their tech support. Alyed
Return-Path: [EMAIL PROTECTED] Mon Jul 24 10:09:29 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Mon, 24 Jul 2006 10:09:29 -0700Received: from digium
) with festival.confFeel like you are in the right track?try dealing with any ".c" file, recompile asterisk and make it behave just the way you always dream of (btw if it works you might want to share your new feature with all of us :) )Alyed
Return-Path: [EMAIL PROTECTED] Tue
As of Asterisk 1.0.X a "#" was recognized as a pattern not as a digit, hence in order to use it at the begining of an extension you should use "_" before it. I guess this is still valid in 1.2.X versions.i.e: use _#31#0046011 in your extensions.confAlyed
Return-Path:
like if some one speaks loud, he would get a low volume.I'm sorry, but this goes far beyond Asterisk (at least for the moment) :)Anyway you can still play with rxgain and txgain in zapata.conf, but this will increase/reduce the overall volume gains and can also affect echo perception.Alyed
You have a little confusion:
friend = can GENERATE and RECEIVE calls
peer = can only GENERATE calls
user = can only RECEIVE callsAlyed
Return-Path: [EMAIL PROTECTED] Fri Jul 07 09:27:13 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by
Hi there!
I'm setting up an E1 with a new Telco and they are asking me to add the
extension number (CallerID)into an "Additional calling party number". Guess it
refeers to a part of the E1 trace they are getting. I've been
playing around with the callerid and in zapata.conf and sip.conf
2 things might worth having a look:
a) set up in your zapata.conf:
musiconhold=default
b) You say the asterisk version is 1.1, but 1.1 is developement
version, maybe was just a typo, but you should be using either a 1.0.X
or 1.2.X versionAlyed
Return-Path: [EMAIL
Hi there!
I'm setting up an E1 with a new Telco and they are asking me to add the
extension number into an "Additional calling party number". Guess it
refeers to a part of the E1 trace they are getting. I've been
playing around with the callerid and in zapata.conf and sip.conf but
have
Have you tryed phoning a fixed line instead of a cell phone?is this giving the same result?I assume your outgoing call to a the cellphone goes through a Zap channel. Try another one (e.g. Zap channel 2), and let us know the result.Alyed
Return-Path: [EMAIL PROTECTED] Sat Apr 01
I used g729 couple of times in the past and got the warning messages ONLY when I was trying to use more channels than the total amount of licenses I'd got.If you are sure you are using only one device that needs the license, I would suggest to check out how it is communicating with Asterisk.
That was a bug fixed in Asterisk version 1.2.3 recently version 1.2.6 was released, so don't worry you can try the latest one without timing fears :DAlyed
Return-Path: [EMAIL PROTECTED] Sat Apr 01 15:42:39 2006Received: from digium-69-16-138-164.phx1.puregig.net
I use Portaone's PortaSIP for everything related to LCRAlyed
Return-Path: [EMAIL PROTECTED] Wed Mar 29 16:48:54 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Wed, 29 Mar 2006 16:48:54 -0700Received: from
I may add a very nice configuration:
- Use two (or more) Asterisks to create your own VoIP network
Very useful if you have broadband and several facilities spread out in distant geographical locations.Alyed
Return-Path: [EMAIL PROTECTED] Wed Mar 29 16:32:16 2006Received:
,Hangup
exten = 200,1,Dial(Zap/g1/${EXTEN},20)
exten = 200,1,Hangup
Then you'll end up with 2 extensions using the same FXS channel (of course not at the same time).
Hope this is what you are looking for.
Alyed
Return-Path: [EMAIL PROTECTED] Wed Mar 29 15:42:30
Think a zaptel recompile is just what you need.Alyed
Return-Path: [EMAIL PROTECTED] Thu Mar 23 17:05:27 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Thu, 23 Mar 2006 17:05:27 -0700
i've got a
Polycom's can work in one of two ways:
a) using self configuration
b) downloading it from a ftp server
To make your Polycoms work with Asterisk you actually don't need the
phone to download any configuration, with the one embeded is ok. In any
case, when turned on, the phone searches for
o use
them as
Voicemail(u100) and
Voicemail(b100) respectivelly in your extensions.conf
Alyed
Return-Path: [EMAIL PROTECTED] Thu Mar 23 16:34:27 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMT
Have a customer running some 25-28 concurrents calls (with about 35 agents logged in)without problems with a P4 2.X Ghz, 1GB RAM,I'm doing no transcoding btw.Alyed Return-Path: [EMAIL PROTECTED] Sat Feb 04 16:59:29 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by
Also get the book (again I dont have the URL if some one does please post it). http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 Alyed If you are new I would reccomend using [EMAIL PROTECTED]http://asteriskathome.soundforge.net . It is a greatresource for beginers. Also get
I would be useful if you could post your config files and the pri debug as well. check your zapata.conf or paste it here so we can take a look.AlyedReturn-Path: [EMAIL PROTECTED] Thu Jan 12 10:04:28 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by
SIP -- NAT -- Internet -- Nat -- Asterisk call them I'm afraid you would need to use a SIP/RTP router. Alyed Return-Path: [EMAIL PROTECTED] Thu Jan 12 09:29:42 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;Please note that recent IOS h
was "stable") So my advice here is to check the driver version you are using if not the very last one, then update it. Try looking at the /var/log/messages file for any extra info, you might find something interesting. Alyed Return-Path: [EMAIL PROTECTED] Thu Jan 12 06:16:56 2006Rece
can't you ask the users to dial a prefix? that can solve your problem. btw, which provider are you using for your calls to the USA? Alyed Return-Path: [EMAIL PROTECTED] Wed Jan 11 09:47:29 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com
Don't know if you can actually adjust the volume in any of them, but you can try from the asterisk with rxgain / txgain in your zapata.confAlyed Return-Path: [EMAIL PROTECTED] Mon Jan 09 16:27:24 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by
here if you continue experiencing problems. Alyed Hi,I'm just in the process of replacing a crappy Siemens PBX with a new andshiny Asterisk system. To connect Legacy equipment I hooked up a smallISDN PBX (DeTeWe OpenCom 36) to one port on a Junghanns.net quadBRIcard. That port is configured for NT
the verbose option . The file will be saved wherever is defined in the asterisk.conf (the default is /var/log/asterisk) after editing the file you'll need to do either an Asterisk restart or input CLI logger rotate at the Asterisk console. i.e. ;logger.conf [logfiles] mylogfile = verbose Alyed I'd
Then stop looking for easy solutions and get your hands dirty changing your c files Alyed Well, I want the output that the NoOp's generate. I want to be able to manually log lines to a file through some mechanism. I just wish I could do it without all the extra NoOp stuff at the front. I just
. -Original Message- From: Alyed Tzompa [mailto:[EMAIL PROTECTED] Sent: Thursday, January 05, 2006 11:59 AM To: Douglas Garstang; asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Asterisk Debugging Then stop looking for easy solutions and get your hands dirty changing your c
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