I tried the same, and my Telco company told me
(although sometimes it's hard to trust them,
you never know what kind of guy from the call center is answering
your call) that p2p lines already have l1 permanent.
Nonetheless it goes down sometimes for quite long periods.
I'm starting wondering
If you have divas4linux package installed (from Eicon), you
can use the Config textual gui utility, it always reports which cards
(model and revision)
are found in your system.
Klaus Darilion ha scritto:
Hi (Armin)!
Does someone knows how to identify the type of the card? The delivery
note
We are testing Asterisk
Realtime configuration with ODBC/MySql. This is our extconfig.xonf
extensions = odbc,asterisk,extensions In the extension.conf
we have inserted this line: switch = Realtime/[EMAIL PROTECTED]
where mainmenu is the context. and in the table 'extensions' we
have this
This is an update on the issues of CDR inaccuracies, i hope this will help someone in need.
In order to remove the Authenticate() function and still be able to perform call accounting and authentication, we pass the authetication process through an AGI. the reason beig that the Authenticate()
Hi Scott,
so it seems that are polycom phones not working well...
have you tried with other IP phones or only with polycom?
Giorgio Incantalupo
Scott Scecina wrote:
Giorgio,
I'll answer in reverse order:
I've not had reports of noise from my users. However, when I went down to
get the
Hi,
I'm testing Asterisk with MySql and I would want to
insert sip users in a table "sip_users". After I modified extconfig.conf with
"sipusers = odbc,asterisk" and I create the table sipusers, which changes
must I make to sip.conf?
Thank's
Maury
P.S.: C'è qualche utente italiano nella
Hello ppl,
Can someone explain to me the meaning and use of the variables
accountcode and amaflags in sip.conf,etc.
Googled, voip-infoed, wikied, etc for it. Couldnt get much of it. I
know, they are billing related, but not much beyond that.
Any ideas?
cheerz
- Ben.
If I put versionStamp in cnf.xml file, how do I check it on the phone?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
We're running 2 TE412P's in a Dell 1850 just fine, been running like
this for well around 6 months to a year now without any problems.
They're not exactly 212P's but I imagine it won't be much different.
On Wed, 2006-10-18 at 10:54
Hi all,
What does 'Got reject for frame...' message really mean, what could be
causing it, and how should one start troubleshooting it?
Thanks in advance,
Alex
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Hi,
As I am a newbie, I am going to ask a newbie question ;-)
I saw Digium has TE212P (with DSP) and TE210P cards (no DSP), and
Sangoma has A102 T1/E1 AFT card (no DSP)...
What is the best choice of T1/E1 card (2ports) for an installation on
Centos 4 (or simply what is the best choice of
Hi Jarkko,
I had the same problem..It worked with an old version of misdn-install
(taken from beronet site) but not with actual mqueue-misdn-install. I
tried to put it in every misdn.conf section I have without success. The
updated beronet install manual doesn't mention that parameter anymore
Hi,
I have a sangoma PRI card on an Asterisk PBX. I have problem with
outgoing caller ID: when I make an outbound call, the called party gets
x1 instead of x240 where x is the my company prefix and 240
is the phone extensions I call from.
I read something about usecallingpres on
Giorgio Incantalupo wrote:
Hi,
I have a sangoma PRI card on an Asterisk PBX. I have problem with
outgoing caller ID: when I make an outbound call, the called party
gets x1 instead of x240 where x is the my company prefix
and 240 is the phone extensions I call from.
I read
Hi,
After upgrading to new version of Asterisk and Zaptel, my Tormenta2 card has stopped working. Zaptel doesn't detect it. What should I do to make it work again. It is enabled in /etc/sysconfig/zaptel. Where else I have to enable it so zaptel can detect it and make it work.
zaptel.conf and
Hi list,
I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP)
registered to Asterisk. One call the other-one, is it possible to order
Asterisk to force answering the call ? i.e. Xlite call Idefisk, Idefisk is
ringing, I send a command to Asterisk which force answer, so Idefisk answer the
Thank you all very much: it solved my problem.
exten = 554,3,GotoIfTime(08:30-14:30|mon-wed|*|*?novm,567,1)
...
[novm]
exten = _X.,1,Macro(exten-vm,novm,${EXTEN})
...
The remaining dialplan is not my specific,
is the standard dial plan provided by FreePbx, which I integrate in
_custom.conf
It's not possible.
The idefisk however has a button to auto answer.
Zoa
Gregory Duchatelet wrote:
Hi list,
I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to
Asterisk. One call the other-one, is it possible to order Asterisk to
force answering the call ? i.e. Xlite call
Hi Maurizio,http://www.voip-info.org/wiki/view/Asterisk+RealTime+SipOften you can find what you seek just spending a minute with google
Cheers,Giovanni (Italiano)2006/10/19, Maurizio Pederneschi [EMAIL PROTECTED]:
Hi,
I'm testing Asterisk with MySql and I would want to
insert sip users in
Hi Greg,Idefisk support Auto-answer only in a biz versionI suppose you got free version..You will find more details http://www.asteriskguru.com/idefisk/free/
Cheers,Giovanni2006/10/19, Gregory Duchatelet [EMAIL PROTECTED]:
Hi list,
I have 2 softphones, 1 Idefisk (IAX), 1 Xlite
I've narrowed it down to 2 configs,
It wil either be a dual:
Intel xeon dual core 3.73Ghz 1066 2x2MB cache
or a dual:
AMD Opteron DUAL CORE 285 (2.6GHz 32/64bit)
So effectively there will be 4 real CPU cores to handle processes/transcoding.
* Are there any numbers on how many (SIP-SIP) Alaw to
Erick,
It looks like the 2.5 laptop drive requires 5 watts to spin up. Adding
that to the 15 watts for the Digium card, leaves about 40 watts
available for the MB. It's unlikely that the system will be producing
ring voltages when the drive is spinning up. It depends on how
conservative you may
Message: 12
Date: Tue, 17 Oct 2006
18:07:04 -0700 (PDT)
From: sdgesa gaeharth
[EMAIL PROTECTED]
Subject: Re:
[asterisk-users] Extremely choppy sound on some of
ourPOTSnetwork calls; goes away with mute
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
amaflags : Categorization for CDR records. Choices are default, omit, billing, documentation and choices are defaul, omit, billing, documentationaccountcode : string : Users may be associated with an accountcode (billing purpose)
Cheers,Giovanni2006/10/19, Benjamin Jacob [EMAIL PROTECTED]:
Hello
In cdr_mysql.conf add userfield=1 under the globals setting.
bp
On 10/18/06, unplug [EMAIL PROTECTED] wrote:
I want to set some custom data in the field of userfield in table CDRas following.exten = s,19,Set(CDR(userfield)=1234)
exten = s,20,Dial(SIP/1234)However, the userfield doesn't get update
Giovanni,
Appreciate your lines mate.
But, Ive already read those, all over the net.
my qs inline :
amaflags : Categorization for CDR records. Choices are default, omit,
billing, documentation and choices are defaul, omit, billing,
documentation
wot r these categories??wot decides these
Hi!
lspci -nv reports:
:0a:03.0 0280: 1133:e013 (rev 01)
Subsystem: 1133:e013
Thus, I suspect I really got a 4BRI-8M V2
Also divactrl reports a 4BRI:
bbgast01:~# /usr/lib/divas/divactrl ctrl -c 1 -CardName
Diva Server 4BRI-8M 2.0 PCI
Let's test faxing :-)
thanks
klaus
Armin
Hi,
For a significant time now (since about 0.2.0-rc8n) the qozap driver
has become very verbose if an ISDN line is not connected... I get the
messages below every couple of seconds in the asterisk logs.
The flaw in the messages is the Alarm cleared message - The alarm
cannot possibly be
Hi,
I have an Asterisk box connected via an anaogue lines(ZAP/1-1) to a Siemens PBX. I take calls off
the PBX and put send it to a premicell connected via ZAP/7-1. Calls orginate from the PBX, hit
Asterisk, then get sent to the premicell.
Can anyone tell me why there is multiple bridge
Hi,
sometimes on my Asterisk 1.2.10 box I get these errors, there are about 50
active SIP channels so I
dont know if calls are getting dropped or not. Should I be worried?
2006-10-18 09:33:59 WARNING[4375]: channel.c:787 channel_find_locked: Avoided
deadlock for
'0xb7341470', 10 retries!
Hi Tomislav,
I use Dell hardware for desktops only. Each time I tried to use a Dell
pc with telephony cards I get problems. It works only with a TDM400 but
if u plan to add something more it is a real nightmare!
Giorgio Incantalupo
Tomislav Parčina wrote:
In article [EMAIL PROTECTED],
Hi Giovanni,
I follow step by step the documentthat you
suggest. The connection between asterisk and MySql works fine, but sip users
can't resgister.
If I query my sipuser tablethe command
realtime load family column value I have not any
result...
What can I check in my configuration?
Hello list, I am trying to include a new message after I receive a Register in
chan_sip, at the beginning I would like to forward the same message to a fixed
IP address, I have seen that fileds like p-sa.sin_addr and p-sin.sin_addr
have to be with the IP address, but I am not sure about how
Like Aaron, our asterisk systems are on Dell servers and even some Dell optiplex systems for small offices. However we use Sangoma cards to skirt compatibility issues.On 10/19/06,
Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi Tomislav,I use Dell hardware for desktops only. Each time I tried to
Robert La Ferla wrote:
I have been experiencing a problem where after someone calls me from an
analog line, the phone call is terminated after a period of time
(anywhere from 15 seconds to 15 minutes) The phone that I use to answer
the call is an Aastra 9133i SIP phone. There are several
Guillermo Salas M. a écrit :
On Wed, 2006-10-18 at 20:08 +0200, Francesco Peeters (Asterisk) wrote:
On Wed, October 18, 2006 19:03, Paul Gaffney wrote:
Hi, can anyone recommend a good IAX phone for use with Asterisk? I'm
looking for a NAT-friendly solution and my SIP phones are good
Is it an NTFS Share?On 10/19/06, Paul Gaffney [EMAIL PROTECTED] wrote:
Message: 12
Date: Tue, 17 Oct 2006
18:07:04 -0700 (PDT)
From: sdgesa gaeharth
[EMAIL PROTECTED]
Subject: Re:
[asterisk-users] Extremely choppy sound on some of
ourPOTSnetwork calls; goes away with
On Thu, Oct 19, 2006 at 10:15:32AM +0530, K Y Iyer wrote:
Hello Again
I ran the genzaptelconf - but I cannot see any channels in asterisk CLI.
I did make and install zaptel after I put in the card. What am I
missing?
zaptel.conf is now configured (hopefully with the correct signalling,
Hello, if somebody using this scenario in production successfully,
please send me info, which ISDN card for asterisk server is usefull for
me (Digium, Sangoma)?
my crucial requirement is caller id name transfer/display between ISDN
(Siemens PBX) and IP phone connected to asterisk
I'm using PRI
I just built 1.4.0 beta 3 on OS X 10.4.8, and it went pretty
smoothly. I didn't need to install wget.
Asterisk starts and runs with 0% CPU. The CLI also works, but hangs
if I try to tab-complete commands. However, that might be because I
don't have any working config files and/or have some
Did you ever get an answer to this problem
?
I too am seeing this and its driving me mad !!!
Jim
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We are having an occasional one w-way audio problem that occurs about
every 25 - 30 calls on a system configured as follows:
Asterisk 1.2.12.1
Sangoma A101 w/wanpipe beta9
Polycom 500s w 1.5.3
This happens only on inbound calls from the PRI. The external caller can
hear our customer answer
I'm a Certified Apple Sys Admin - lots of experience with Macs and
Mac servers. However, when setting up an asterisk server, I'm still
thinking a Dell box with linux is the best direction - to get the
full reliability and full support of this group. Am I mistaken? Or
is using a Mac box
Hi Mike,
Sounds like you're having about the same problem Giorgio and I are having.
I'd be really surprised if you don't start having the same problem from
SIP-SIP calls to. I also have a Sangoma card, and originally thought it was
only on calls coming from a PRI. But as time has moved forward,
Hi List:
Please someone help me to point out where I can get the idea to
configure
Asterisk for mysql server running on different Linux.
Many thanks,
Tielin
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We use almost all Polycoms, several hundred
had one way audio with 1.6.4 or 5, forget which
1.66 and 2.01 seem to be ok
We did have a few phones (2-3) that had random one way for a long
time, replaced everything feeding them and it still happend. A month
ago I replaced the phones and have
When using Asterisk 1.4b3, everytime I make a call I get continuously
(around 20 times a second) error messages like the following upon the
call connecting (or getting to ring):
[Oct 19 17:23:22] WARNING[28682]: channel.c:767 ast_queue_frame: Unable
to write to alert pipe on IAX2/gradwell-7,
On Thu, Oct 19, 2006 at 05:38:46AM -0400, Zeeshan Zakaria wrote:
Hi,
After upgrading to new version of Asterisk and Zaptel, my Tormenta2 card has
stopped working. Zaptel doesn't detect it. What should I do to make it work
again. It is enabled in /etc/sysconfig/zaptel. Where else I have to
I don't have access to the sip code right now, but from past network code
I've writen you could try this:
inet_aton(192.168.1.10, p-sin.sin_addr);
If they are just wrapping the struct sockaddr_in as sin in p. Worth
a try...
Ryan
Hello list, I am trying to include a new message after I
Right -
I get the error on the console - I just can't tell how many
transcodes are occuring at any given point in time...
On 10/18/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
Mr. Jones wrote:
Is there some way I can tell?
On 10/16/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
Mr. Jones wrote:
The latest X-lite version has autoanswer button on the front.. marked AA
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
Hi Greg,
Idefisk support Auto-answer only in a biz version
I suppose you got free version..
You will find more details
Our company's PBX is running Asterisk 1.2 under OS X Server and it's
been pretty reliable for the 50+ extensions that we have. The system
has an uptime of 245 days, and no one has ever reported dropped calls
or any other disturbing behavior. The reason we're using OS X instead
of Linux is
Is plainvoip down?
I've tried to contact them via email and their 800-956-3285; nobody is
answering or replying to emails
--
#Joseph
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Joseph wrote:
Is plainvoip down?
I've tried to contact them via email and their 800-956-3285; nobody is
answering or replying to emails
I can get there just fine. Your routes might be toasted
[EMAIL PROTECTED] ~]# ping -c 10 plainvoip.com
PING plainvoip.com (66.199.240.2) 56(84) bytes of
If you are using MySQL for storing CDR's, this is what I use (slightly
modified, of course):
cat /etc/asterisk/cdr_mysql.conf
;
[global]
dbname =
hostname=
password=
port
Joseph wrote:
Is plainvoip down?
I've tried to contact them via email and their 800-956-3285; nobody is
answering or replying to emails
Since when is the asterisk-users list second level support for VoIP
providers?
If they are down, I am sure they are well aware of it.
Jeremy McNamara
Im looking at getting a T1 into a location in
Melbourne, Australia and was wondering if anyone has a good source and pricing
for this.
Cheers,
Steve
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On Thursday 19 October 2006 14:43, Forum wrote:
I'm looking at getting a T1 into a location in Melbourne, Australia and was
wondering if anyone has a good source and pricing for this.
I think your looking for a E1 Australia follows the European standard last i
looked. I never did any voice
I caught a thread the other day concerning Astricon and users embedding
Asterisk on a Linksys or Netgear broadband router. I lost track of the
email thread, if anyone is presently working with this scenario please shoot
me an email.
Thanks
Cory Andrews
++
VoIPSupply.com
Cory Andrews wrote:
I caught a thread the other day concerning Astricon and users embedding
Asterisk on a Linksys or Netgear broadband router. I lost track of the
email thread, if anyone is presently working with this scenario please shoot
me an email.
Thanks
Cory Andrews
++
I am now running 1.4 beta3
I have an ongoing issue that it does not recognize my DTMF key press. I
will call and press as many numbers and the background message still plays.
I am also having an issue with transfers
NOTICE[30930]: chan_sip.c:13289 handle_request_invite: Unable to
create/find SIP
CanI now 5th it ? All this makes me wonder
why Digium dosent work harder. I have mainly only seen others praise Sangoma
over Digium.
- Original Message -
From:
Tom Vile
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, October 18, 2006 4:22
I have a backup of a working version on my server
some where. When i find it I will post it.
Dovid
- Original Message -
From:
Eric
Jacksch
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 18, 2006 4:20
PM
Subject: [asterisk-users] Findme
Ooops. I read the email wrong. The macro I created
called one number. If the person didnt accept the call or if they didnt pick up
then it tried the second person. Let me know if you still want it.
Dovid
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Hi,
we have tested the Digium-Cards, they work fine, but don't expect to much!
Only segmentation 1 in Ecma (it is not a digium-problem)
The Name ist displayed, but only in Hex-Code (this is due to the
Libpri/Zaptel Drivers but I didn't fint a way to display it in *)
There is also very less
On Thu, 2006-10-19 at 22:39 +0200, Dovid B wrote:
Can I now 5th it ? All this makes me wonder why Digium dosent work
harder. I have mainly only seen others praise Sangoma over Digium.
I strongly suspect digium is painfully aware of the problems with some
combinations of mboards and their
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark
Sent: Thursday, October 19, 2006 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Occasional one-way audio - Sangoma A101
We are having an
Thanks Dennis,
I am an Australian living in Canada and like yourself have done no voice
work
back home
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dennis Gilmore
Sent: Thursday, October 19, 2006 1:06 PM
To: Asterisk Users Mailing List -
Bristuff has been updated;
http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1u.tar.gz
--
Vidar
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If I set the logging.conf to log DTMF it only seems to log dtmf messages
that are bridged through the * server. If the call goes into a menu the
DTMF dont get logged. Is the intended behavior?
Scott England
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On 23:04, Thu 19 Oct 06, Vidar wrote:
Bristuff has been updated;
http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1u.tar.gz
Thanks for the information.
It's a shame we need to read this here and not see it on
their website.
--
Michiel van Baak
[EMAIL PROTECTED]
Mike,
I'm not able to download the newer polycom software(s) 'cause I'm not
certified in anything.
I just saw the announcement on the Asterisk updates today. I'll get them
installed tonight...
- Scott
Scott Scecina wrote:
Hi Mike,
Sounds like you're having about the same problem
I think the recent Digium and Sangoma cards are quite similar. (and
about the same price)
I didn't try sangoma so far, never had any issues with the digium cards,
I have no clue how the digium helpdesk is, i never needed to call them.
(well not really correct i did call them once, years ago
We are looking at migrating our office from a Samsung PBX to an Asterisk
PBX. I am looking at ordering a PRI with 12 Channels for now (we currently
have 8 analog lines) and need to know what PRI card you guys would recommend
that we use. I have seen some with Echo Cancellation and so on, but
Further, I should have mentioned my resolution. As Bob mentioned,
upgrade the power supply now to give you options and peace of mind in
the future. I am using a picoPSU-120 in the aforementioned itx box, for
example,
http://www.mini-box.com/s.nl/it.A/id.417/.f?category=13
and have been
Joseph wrote:
Is plainvoip down?
I've tried to contact them via email and their 800-956-3285; nobody is
answering or replying to emails
This is starting to sound like a rerun of Livevoip. Remember that company?
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On Thu, 19 Oct 2006 16:10:23 -0400, Cory Andrews wrote:
I caught a thread the other day concerning Astricon and users embedding
Asterisk on a Linksys or Netgear broadband router. I lost track of the
email thread, if anyone is presently working with this scenario please shoot
me an email.
Cory,
Conrad Wood wrote:
On Thu, 2006-10-19 at 22:39 +0200, Dovid B wrote:
Can I now 5th it ? All this makes me wonder why Digium dosent work
harder. I have mainly only seen others praise Sangoma over Digium.
I strongly suspect digium is painfully aware of the problems with some
On Thu, Oct 19, 2006 at 11:27:07PM +0200, Michiel van Baak wrote:
On 23:04, Thu 19 Oct 06, Vidar wrote:
Bristuff has been updated;
http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1u.tar.gz
Thanks for the information.
It's a shame we need to read this here and not see it on
Hi folks,
I have a Sangoma A200 10 port FXO card for sale.
US$500 secures plus shipping.
Thanks
Mark
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Thanks!!
Just one more question. Can I do the same add fieldname=1 if I add
a field fieldname in the cdr table to perform the same action?
On 10/19/06, William Piper [EMAIL PROTECTED] wrote:
In cdr_mysql.conf add userfield=1 under the globals setting.
bp
On 10/18/06, unplug [EMAIL
E1 is readily available in Australia, and very easy to set up.
later,
PaulH
On Thu, 2006-10-19 at 15:05 -0500, Dennis Gilmore wrote:
On Thursday 19 October 2006 14:43, Forum wrote:
I'm looking at getting a T1 into a location in Melbourne, Australia and was
wondering if anyone has a good
On Fri, 2006-10-13 at 01:55 +1000, Matt wrote:
Avi Miller wrote:
On 04/10/2006, at 1:55 AM, Matt wrote:
How can I make * aware of the other ext on the remote box so the DID
caller can access them like he can with the local box?
On each box, define the other range:
Box A:
Dennis, I work for a company that can provide E1's in a secure data
center in Melbourne CBD.
Contact me off list for a quote etc.
Paul Hales wrote:
E1 is readily available in Australia, and very easy to set up.
later,
PaulH
On Thu, 2006-10-19 at 15:05 -0500, Dennis Gilmore wrote:
On 2006-10-19 08:51:01 -0700, Dustin Wenz [EMAIL PROTECTED] said:
I just built 1.4.0 beta 3 on OS X 10.4.8, and it went pretty smoothly.
I didn't need to install wget.
Asterisk starts and runs with 0% CPU. The CLI also works, but hangs if
I try to tab-complete commands. However, that might
On 2006-10-19 09:30:14 -0700, Todd- Asterisk
[EMAIL PROTECTED] said:
I'm a Certified Apple Sys Admin - lots of experience with Macs and Mac
servers. However, when setting up an asterisk server, I'm still
thinking a Dell box with linux is the best direction - to get the full
reliability
I have installed the libtiff(3.5.7),spandsp-0.0.2pre24,app_txfax and
app_rxfax,asterisk-1.2.12.1 on the CentOS 4.2.
I set sip.conf like this:
[sip_local]
host=192.168.2.111
type=friend
dtmfmode=rfc2833
canreinvite=no
insecure=very
I am having the same issue as below. Has this issue been solved or does anyone know an
answer? This error recently began and we
have multiple phones out of commission.
PLEASE HELP!!
http://lists.digium.com/pipermail/asterisk-users/2006-August/162841.html
How did you find out about
On 2006-10-18 12:34:43 -0700, [EMAIL PROTECTED] said:
Hello, well, I need to configure two asterisk box like SIP trunks to send sip
calls from one asterisk to the other and visceversa. So How I setup confi g
files to get this working?.Thanks.
You can do it via IAX2, there was a recipe posted
Okay, I have Asterisk up and running on Fedora Core 5 with a TDM400
board with one FXO and FXS module. Zap is up and running and * is
functioning with the modules. Oh yeah, and I have some soft phones
configured and have them working as well.
Now I'm ready to begin playing with dial plans
On Thursday 19 October 2006 14:10, Cory Andrews wrote:
I caught a thread the other day concerning Astricon and users embedding
Asterisk on a Linksys or Netgear broadband router. I lost track of the
email thread, if anyone is presently working with this scenario please
shoot me an email.
I
* is having permission problems accessing /dev/zap/channel. When I
look, these devices (everything in /dev/zap) shows root.root for uid and
gid. If I start Asterisk from the command line, it runs fine (running
as Root). When I start it as a service, I get
Oct 19 23:02:55 WARNING[10587]
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