Good news!
I did an SVN update to my 1.4 branch again today, and 1.4-r46154 seems
to have resolved the asterisk hogging the whole CPU issue.
I still can't use the regular console though (asterisk -r) as that is
unresponsive.
Using asterisk -c to start it , works and gives me a color CLI
I would like to put srvlookup=no in my SIP conf, so that I don't get DNS issues
(Asterisk stops responding). I use VoIP Buster and in sip.conf I use
sip1.voipbuster.com. When I do sip show peers in CLI I get
voipbuster/tomo 194.221.62.207 5060 OK (27 ms)
And when I ping
Steve Underwood wrote:
If someone wants to take my code and make it work with Asterisk under
GPL conditions, that's fine. The GPL gives you that right. Please make
sure you stick to GPL conditions, though. You can't use G.729, for
example, in an Asterisk that's using spandsp.
I do not see
2006/10/24, Alberto Pastore [EMAIL PROTECTED]: (there are a couple of serious issues using bristuff
and we've been looking for alternate drivers).Hi,Which issues do you have ?Regards
___
--Bandwidth and Colocation provided by Easynews.com --
On 2006-10-24 17:25:37 -0700, Steve Underwood [EMAIL PROTECTED] said:
The development of Asterisk has now degraded to the point where I will
no longer contribute anything to it.
I am not interested in a flame war, but would love to here a more
explicit explanation for what is occurring
I think this is now the Eicon HMP platform. It looks like Eicon bought
this when the fools paid good money for Dialogic.
Its amazing how many companies have got on the HMP bandwagon since we
started the Zapata work in 1999. If you do a Google search you can find
something like 10 companies
On 2006-10-24 13:04:02 -0700, Matt [EMAIL PROTECTED] said:
Hi,
I have a customer who experiences, once in a while, one-way audio...
That is... they can hear the person they called, but the person can
not hear them.
The customer is connected via IAX2 to our softswitch.
On the customer's end I
On 2006-10-24 06:44:01 -0700, Wildheart
[EMAIL PROTECTED] said:
Hi,
Does anyone know a what to use a different codec for calls which are
handset to handset (eg, G711) then when we have calls to the out side
world (via an asterisk server) to use a different codec(eg, G729)?
The idea
In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
Kristian,
I don't have any zap hardware What do I put in zaptel.conf if I don't
have any hardware?
On some other systems we have, with chan_zap not loaded, and no zaptel.conf
(running
1.2.9.1), meetme runs fine.
Hi all!
We've released VoiceOne 0.4.0, a web-based and open source solution
which allows to fully manage an Asterisk service hosted on a LAMP server.
We focused on an charming and overall user-friendly interface. Thanks to
the authentication based on roles, once configured by a super user,
Thank you.
Indeed, this is what I want to know. When somebody wants to make a call
(using a standard telephone, connected to a media gateway), he doesn't
know what user is in my Asterisk conf. He only knows that he wants to
call John, who has the number 102 for example. He dials 102 from his
Hi,I have installed Asterisk, Zaptel, Libpri, Addons, Sounds in my Linux system. I got registered with sipgate.co.uk and got the UK phone number i.e., 0207100. I configured my Asterisk server with 0207100. When I made a call to this number from outside phone, my XLite extension is not
Hello,
you have to install package with kernel sources or at least with
kernel headers to compile zaptel sources...
Sincerely
Jan Marek
On Sat, Oct 21, 2006 at 08:18:32PM +0530, ram wrote:
Hi
iam installing zaptel 1.2.10 on my FC5
when i make iam getting following error
any one suggest
Hello
What soundfile format, is the one that uses least transcoding during playback?
As I can see, I can choose wav or gsm. What sucks least cpu power, during
playback to example a Zap channel? I would guess wav, but is this correct?
Kind Regards
Jon Leren Schøpzinsky
--
No virus found in
On Wed, 2006-10-25 at 11:24 +0200, Jon Schøpzinsky wrote:
Hello
What soundfile format, is the one that uses least transcoding during playback?
As I can see, I can choose wav or gsm. What sucks least cpu power, during
playback to example a Zap channel? I would guess wav, but is this correct?
Hi All ...
I'm running Asterisk 1.2.13-BRIstuffed-0.3.0-PRE-1v which has a
Junghanns QuadBri card in it (lspci reports Cologne Chip Designs GmbH
ISDN network Controller [HFC-4S] (rev 01))
I have a regular KPN ISDN2 Line connected which works fine with the
the zapata.conf
below.
However, I
On 10/25/06, Giray Devlet [EMAIL PROTECTED] wrote:
/etc/asterisk/zapata.conf
switchtype = euroisdn
; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode)
signalling = bri_cpe_ptmp
Have you tried signalling = bri_cpe if you have a group of ISDN
channels, they are more often in
Don Wisdom wrote:
Hi All,
Im trying to erase the config in a addit that I got off of ebay. I know
Try no password. Just hit enter. If that doesn't work, you'll have to
contact Carrie Access technical support. They'll charge you an arm and
a leg. Nobody has reported any successes
Xue Liangliang wrote:
Hi, all i receive these logs quite often, and all the calls hangup
after receiving these .
Oct 25 11:17:44 NOTICE[5121]: chan_zap.c:8176 pri_dchannel: PRI got
event: Alarm (4) on Primary D-channel of span 1
Oct 25 11:17:44 WARNING[5121]: chan_zap.c:2289 pri_find_dchan:
dtmf = inband
Best regards,
Al Bochter
Bochter Services
(Voip PBX) Toll Free: 866-638-1254 EXT: 250
(Voip PBX) Free World DialUp: 780217 EXT: 250
(Voip) Cellular: 712-432-5401
http://www.BochterServices.com/?t=Email
BUY and sell Coins, Silver and Gold
Hi Marty,
By the outside world, I mean the PSTN connection. I am still interested
in how you would set this up. Can you paste in a sample config?
With thanks,
Tim
On 2006-10-24 06:44:01 -0700, Wildheart
[EMAIL PROTECTED] said:
Hi,
Does anyone know a what to use a different
On Wed, Oct 25, 2006 at 11:11:00AM +0300, Paul Ianas wrote:
Indeed, this is what I want to know. When somebody wants to make a call
(using a standard telephone, connected to a media gateway), he doesn't
know what user is in my Asterisk conf. He only knows that he wants to
call John, who has
So no one has any solution to this, huh? We can't be the only two
people having this problem.
On 10/24/06, Matt [EMAIL PROTECTED] wrote:
Just as a follow up.. on the OTHER server that is connected I'm seeing:
chan_iax2.c: Received VNAK: resending outstanding frames
On 10/24/06, Matt
Marty,
Thanks for the suggestion... unfortunately it is not a case of the
bandwidth being hammered. The only things on this connection is the
voice.My thought is there is something wrong, possibly, with the
cable provider's node. Still.. Asterisk shouldn't just barf with
one-way-audio.
On
Pavel,
What version of asterisk are you connecting back to? Is it also 1.4. branch?
On 10/25/06, Pavel Jezek [EMAIL PROTECTED] wrote:
I have same problem, but only with 1.4 branch and when some bigger
jitter occur (1.2 is working fine, even in case with big jitter),
I dump packets with tcpdump
Hi,
Im posting here as I have found an issue in
1.4, and hoping someone might be able to help.
I have setup a call queue in asterisk, a call comes
into the queue, asterisk calls the agents, an agent answers the call fine, but
if they try and transfer the call, asterisk drops out with
Is anyone having problems and Broadvoice with incoming DTMF not being
recognized from a caller originating on the PSTN connection to Broadvoice?
This is the reason why I left them two months after I signed up with them.
Broadvoice tech support confirmed this issue as a result of their carrier
Henry.L.Coleman wrote:
Yep, just swop the two wires. Sometimes the Tip and Ring get reversed
and most loop start interfaces don't really care (they work either way).
It's worth a try since if the disconnect is a reverse polarity flash then
the card may see not see this condition as it is
Hi Steve,
THX!!! This works ... couldn't really find anywhere what other
options I could use as values for signalling ... thx!
gd
From: Steve Davies [EMAIL PROTECTED]
On 10/25/06, Giray Devlet [EMAIL PROTECTED] wrote:
/etc/asterisk/zapata.conf
switchtype = euroisdn
; p2mp TE mode (for
Are you behind NAT. Any firewall's ?
- Original Message -
From:
Crazy
Boy
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 25, 2006 10:54
AM
Subject: [asterisk-users] Call is not
coming through sipgate.co.uk+Asterisk
Hi,I have installed
[EMAIL PROTECTED] wrote:
We have a problem where callerid works 50% of the time on both lines. What
we are seeing in the logs is:
Hi Phil,
Unfortunately your configuration looks OK to me.
Here's mine, which works 100% with CID (but not dratted hangup
detection!). There are some
Marlin Unruh wrote:
Hi,
I have a Polycom 501 that is currently unusable because I started a
firmware and sip upgrade that I can't complete.
The Ubuntu box address is set static at: 192.168.1.101.
The phone address is set static at 192.168.1.51.
The phone settings for the server menu are:
Hello,
I try configuring Quintum DX gateway as link to PSTN for *. Now, I can dial
number which is connect to Quintum, and call is diverted to *. I don't know
what I should set, if I want call from SIP_phone registred in Asterisk to PSTN
via Quitnum. I set in sip.conf account for Quintum
Marlin Unruh wrote:
Glad to say I got it working. Sad to say I had to go to Windows to
accomplish it. I used tftpd32 and it worked perfect.
I would like to use tftp under Linux. May I will try again later.
Why not use just standard FTP? I use ProFTP and setup a Polycom user.
Works great.
Marlin Unruh[EMAIL PROTECTED] Wrote on: 10/25/2006 8:12 AM:
Marlin Unruh wrote:
Hi,
I have a Polycom 501 that is currently unusable because I started a
firmware and sip upgrade that I can't complete.
The Ubuntu box address is set static at: 192.168.1.101.
The phone address is set static
I have a setup with a polycom 601 and an act p160s. All on local segment, no
NAT.
Can call the act p160s, from the polycom, rings, connects, and a conversation
can take place. The reverse is not true, Dialing from the act to the polycom
does not work. SIP debug shows, at the end, Incoming
Thanks Andrew
I have no plans to VoIP my Faxes to a VoIP provider
I just would like to send them from my desktop (which is windows) to my
PBX (which is AstLinux inside a net 4801)
The PBX connects to PSTN lines via a FXO Gateway (CG-410 in my case)
So really it's trying to get Windows to
Hi,
Is it possible to define maximum talk time in a queue? ie any one who
joins a queue should not be able to talk more than say 5 minutes to
the agent.
raj
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
I am attempting to implement Nerdvittle's Call Reminders on my *
1.2.12.1 PBX. It has 8 Zaptel trunks provided by 2 Digium TDM400P
cards.
If I use the call reminders internally, it works flawlessly. The
problem happens when I set the call-back number to an external number so
that the call goes
ram wrote:
Hi all
I have installed 1.2.12.1 http://1.2.12.1 in FC5 with libpri.1.2.4
when i start
iam getting the following error and it quits
== Registered channel type 'Local' (Local Proxy Channel Driver)
[chan_zap.so]Oct 23 16:16:07 WARNING[11084]: loader.c:325
__load_resource:
IMHO, FTP really is the way to go - you get the ability to have the
phones detect config file changes and automatically reboot, and you
get the ability to upload logs, custom configs and directories from
the phones.
We use vsftpd, with the default user and password for the phone.
CP
On
John McCollough wrote:
I was wondering if anyone had any suggestions on how to work around this
problem. The only thing I can think of that is within my ability is to
The common work around for analog lines it to loop a message asking the
caller to press 1 to accept the call. Loop it
That too.
I never used Broadvoice but from what users have told me high priced
poor service.
There are better with no connect fees
Best regards,
Al Bochter
Bochter Services
(Voip PBX) Toll Free: 866-638-1254 EXT: 250
(Voip PBX) Free World DialUp: 780217 EXT: 250
(Voip) Cellular:
So a PRI line resoves this issue as well? That's good. I believe there
are plans for upgrading to one.
Thank you
John
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Wednesday, October 25, 2006 9:43 AM
To: Asterisk Users Mailing
On Wednesday 25 October 2006 09:20, John McCollough wrote:
What appears to be happening is that the reminder script simply waits
for a connected call, then starts playing it's message, but * reports a
connected call when it connects to the trunk, not when the other party
picks up. The result
On Wed, Oct 25, 2006 at 01:54:43AM -0700, Crazy Boy wrote:
My sip.conf file contents:
...
[250]
type=friend
username=250
secret=danny
callerid=Danny
host=dynamic
context=demo
register = 100:[EMAIL PROTECTED]/100
...
My Extensions.conf file contents:
You are welcome. Please let me know if this makes any difference.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
Henry.L.Coleman wrote:
Yep, just swop the two wires. Sometimes the Tip and Ring get reversed
and most loop start interfaces don't really care (they work
I have installed 1.2.12.1 http://1.2.12.1 in FC5 with libpri.1.2.4
when i start
iam getting the following error and it quits
== Registered channel type 'Local' (Local Proxy Channel Driver)
[chan_zap.so]Oct 23 16:16:07 WARNING[11084]: loader.c:325
__load_resource: libpri.so.1.0:
Hi Edwin -
rename bootrom.ld to something else like bootrom.ld-disabled.
did that. it hung on sip.ld, rename sip.ld, it hung on
phone1.cfg. seems like if the file is bigger than say 1k.
it'll hang.
I like ProFTPd - it's my ftp daemon of choice for configuring Polycom
phones (including
Hello Users,
Is Without Zaptel interface Installed, conference Bridge is worked
or not.
Why it need, For SIP conferences through OpenSER
Please Help me
For me its Giving Some Errors and warnings.
== Parsing '/etc/asterisk/meetme.conf': Found
Oct 25 18:16:13 WARNING[12281]:
Hi Bill -
I would like to setup Asterisk for voicemail with CallManager 3.3(5). I
would like to know what would be the best Distro of Linux to use and
version, what version of Asterisk works best to interact with CallManager,
and what H323 ChannelType works. As you probably read in another
Hi list,
I ran into some trouble trying to get asterisk (1.4beta2) to compile with
misdn support. (I need to run a hfc card in NT mode)
when I run ./configure --with-misdn=/usr
it results into the following error:
checking for mISDN_open in -lmISDN... yes
checking
Hi Matt -
I have a customer who experiences, once in a while, one-way audio...
That is... they can hear the person they called, but the person can
not hear them.
On the customer's end I have the following config in iax.conf:
trunk=no
(I have also tried trunk=yes and nothing for trunk=)
What's the native soundfile format for SIP? Any idea which soundfile
takes the least CPU for mixing together in conferences?
How about whether the CPU load for conferencing native data is
greater/less than the CPU load for transcoding non-native data that is
CPU lighter in the
On Wed, 2006-10-25 at 19:52 +0530, sunkara wrote:
Hello Users,
Is Without Zaptel interface Installed, conference Bridge is worked
or not.
Why it need, For SIP conferences through OpenSER
Please Help me
For me its Giving Some Errors and warnings.
You need to
-Original Message-
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 25, 2006 1:26 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Meetme... No channel type registered for
'zap'
In article
[EMAIL PROTECTED],
Douglas Garstang [EMAIL
Is Without Zaptel interface Installed, conference Bridge is worked
or not.
Why it need, For SIP conferences through OpenSER Zaptel interfaces provide timing that is necessary for meetme conferences. When you start a conference, on the cli you can see that asterisk opens a ZAP/pseudo channel.
On 2006-10-25 08:14:43 -0700, Noah Miller [EMAIL PROTECTED] said:
Hi Matt -
I have a customer who experiences, once in a while, one-way audio...
That is... they can hear the person they called, but the person can
not hear them.
On the customer's end I have the following config in iax.conf:
Hello,
i hev a subscription to a international voip provider and I want
all calls for numbers _001xx to go through my voip provider. I
tried many settings in sip.conf, extensions.conf and iax.conf. Please
give me some simple example for how can i transfer the specified calls
to my external
Hello Users,
Good Morning,
I'm doing on Conference Bridge with Asterisk + OpenSER with CBMySql
modules.
And I'm not Using the Zapptel Cards.
9001 -- dial 19001(conference Users)---openSER
- Asterisk
In Extension.conf
[from-sip]
exten =
What soundfile format, is the one that uses least transcoding
during playback?
As I can see, I can choose wav or gsm. What sucks least cpu power, during
playback to example a Zap channel? I would guess wav, but is this correct?
When you get down to it, the asterisk native format is slinear.
On Wed, Oct 25, 2006 at 10:06:02AM -0600, Douglas Garstang wrote:
-Original Message-
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 25, 2006 1:26 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Meetme... No channel type registered
Dear Friends and Supporters!
I try to write a php application to monitor the
asterisk, but when I open the .php to access to asterisk to retrieve the
information about the queues status, sip show peers, realtime mysql status
etc... However, It just return to me "Unable to connect to remote
Hi Raj,
if you use Local channels for agents (or callback agents), you can easily
do this in the Dial() command after the Local channel is called. Of course
your clients may get a bit angry at being disconnected, it is usually
better to jave each agent stay aware od the call length and
On 2006-10-24 06:44:01 -0700, Wildheart
[EMAIL PROTECTED] said:
Hi,
Does anyone know a what to use a different codec for calls which a
re
handset to handset (eg, G711) then when we have calls to the out side
world (via an asterisk server) to use a different codec(eg, G729)?
snip
I
That Wiki covers CCM4 and my company doesnt have the cash to upgrade to that
yet. I have to stick with H323. I actually started from scratch and went to
the 1.2 version of Asterisk.
-Original Message-
From: [EMAIL PROTECTED] on behalf of Noah Miller
Sent: Wed 10/25/2006 9:54 AM
To:
I don't have any calling search spaces defined.
-Original Message-
From: [EMAIL PROTECTED] on behalf of Pavel Jezek
Sent: Wed 10/25/2006 3:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323
Did
PJ Wrote:
Did you apply correct calling search space on callmanager
gateway configuration page for incomming calls from asterisk
to callmanager? imho, oh323 is obsolete/unmaintained, I'm using
original chan_h323 with callmanager 4.1 and it working fine
(including dtmf), ooh323 is probably
On Wed, 2006-10-25 at 12:15 -0400, Noah Miller wrote:
What soundfile format, is the one that uses least transcoding
during playback?
As I can see, I can choose wav or gsm. What sucks least cpu power, during
playback to example a Zap channel? I would guess wav, but is this correct?
When
On 2006-10-25 08:14:56 -0700, Matthew Rubenstein [EMAIL PROTECTED] said:
What's the native soundfile format for SIP?
??? I think you might need to do some research (the above is a nonsense
question I think).
Any idea which soundfile
takes the least CPU for mixing together in
-Original Message-
From: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 25, 2006 10:18 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Re: Meetme... No channel type registered
for'zap'
On Wed, Oct 25, 2006 at 10:06:02AM -0600, Douglas
On Wed, Oct 25, 2006 at 07:14:23PM +0300, Jonson Player wrote:
i hev a subscription to a international voip provider and I want all
calls for numbers _001xx to go through my voip provider. I
tried many settings in sip.conf , extensions.conf and iax.conf. Please
give me some
On 24 Oct 2006, at 01:05, Carla Schroder wrote:
What does option '11. Module Embedding' do in Asterisk 1.4? The
default is
none of them are selected:
[ ] 1. apps
[ ] 2. cdr
[ ] 3. channels
[ ] 4. codecs
[ ] 5. formats
[ ] 6. funcs
[ ] 7. pbx
[
In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
Tony Mountifield [mailto:[EMAIL PROTECTED] said:
Doug, it sounds to me like you don't have the /dev/zap device files.
Do you have the file /etc/udev/permissions.d/zaptel.permissions and
Asterisk seems to have a bug which is not letting me set the caller id
to another peer's caller id.
http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg23230.html
I've sent this to the asterisk-users mailing list, hopefully I get a
response soon if there is a workaround.
I'm going
Asterisk is current running with the a file in /var/run/asterisk.ctl for the user asterisk. I have set asterisk to be the owner of the folder /var/run, and start asterisk with user is asterisk. HTTPD is run under asterisk user. My
manager.conf has an entry.
Check to make sure the file is
Same here with Brazilian Portuguese. :)
Nicolas S. wrote:
Hi,
I can help in French translation if needed.
Drop me the procedure to do it.
Regards
Le mercredi 25 octobre 2006 à 09:51 +0200, Alex a écrit :
Hi all!
We've released VoiceOne 0.4.0, a web-based and open source solution
Well I seem to have removed my call pattern too many times and now CCM isn't
routing it anymore.
-Original Message-
From: [EMAIL PROTECTED] on behalf of Dan Austin
Sent: Wed 10/25/2006 11:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users]
If the audio is dropping out completely, then I suspect the whole
jitter buffer thing is a red herring (waste of time).
Perhaps it's a nat issue? What kind of router if any is involved? I
am reaching here... Also, please do tell us which version of asterisk
you are running...
I apologize.. I
Dear Friends and Supporters!
I try to write a php application to monitor the
asterisk, but when I open the .php to access to asterisk to retrieve the
information about the queues status, sip show peers, realtime mysql status
etc... However, It just return to me "Unable to connect to remote
I have a colleague who had an IP PBX solution put in by a reseller and
they are having an issue with their ZAP channels becoming unresponsive.
They are using a Digium TDM2400 Series, all inbound and outbound through
the FXO ports, VOIP is internal only.
Anyone aware of any known issues with
We are looking at porting asterisk onto a embedded platform based on IXP or ARM. I would like to know the experiences of anybody who has already ported to these platforms. I am also particularly interested in issues related to performance and scaling on these platforms.Also, is anybody aware
We are looking at porting asterisk onto a embedded platform based on IXP or ARM. I would like to know the experiences of anybody who has already ported to these platforms. I am also particularly interested in issues related to performance and scaling on these platforms.Also, is anybody aware
-Original Message-
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 25, 2006 11:10 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Meetme... No channel type registered for
'zap'
In article
[EMAIL PROTECTED],
Douglas Garstang
Hello List,
Question: Has anyone been able to create multiple queue_log files in
/var/log/asterisk for multiple queues?
We are designing a multi-tenant system and separating the log files
would be useful, instead of dropping all queue actions into one file.
Is it possible this is a user
Hi
I am looking for a good wirless headset to use with the Polycom Soundpoint 501
phone. I would greatly appreciate hearing from anyone with good experiences
with a specific device.
Thanks
--
Jim Freeze
___
--Bandwidth and Colocation provided by
On 09:51, Wed 25 Oct 06, Alex wrote:
snip
/snip
Any plans to support multiple virtual pbx-en on one asterisk
instance ?
That's something almost no webbased tool implements. It's
all one asterisk, one pbx while asterisk is very capable
of virtualhosting PBX-en on one instance.
Would be a great
Hi List:
I have been testing Xlite 2.0 and 3.0. The Xlite 2.0 is slow on
initiate time, but I can add second sip proxy account, which is very
critical to my testing. I installed Xlite 3.0, which I could not add
second account on SIP account settings. After I add the first one, the
Add button is
On 10/25/06, Christopher Aloi [EMAIL PROTECTED] wrote:
Hello List,
Question: Has anyone been able to create multiple queue_log files in
/var/log/asterisk for multiple queues?
We are designing a multi-tenant system and separating the log files
would be useful, instead of dropping all queue
In over three years of using Asterisk in the lab and also in real-world deployments and supporting other Asterisk users, the single most common problem I have encountered and seen others encounter is the message No Authority Found and the inability to call between machines when using IAX. This is
On 13:12, Wed 25 Oct 06, Maps wrote:
Dear Friends and Supporters!
I try to write a php application to monitor the asterisk, but when I open the
.php to access to asterisk to retrieve the information about the queues
status, sip show peers, realtime mysql status etc... However, It just
I've used the Plantronics ones, similar to these: http://www.plantronics.com/north_america/en_US/products/cat29880043/cat29880058/prod5510016
and they work very well with the headset lifter, The range is pretty good too.However there are more elegant and complete solutions, with those headsets
Dan, can you supply your ooh323.conf for me? I would like resolve my
issue with not recognizing dtmf by ooh323 from callmanager
my ooh323 is quite simple, also on callmanager config page for gateway
to asterisk is nothing special, no faststart, no mtp; ccm v4.1.3sr3a
[general]
disallow=all
On Wed, Oct 25, 2006 at 11:37:35AM -0700, Tielin Xu wrote:
I have been testing Xlite 2.0 and 3.0. The Xlite 2.0 is slow on
initiate time, but I can add second sip proxy account, which is very
critical to my testing. I installed Xlite 3.0, which I could not add
second account on SIP account
On 14:29, Wed 25 Oct 06, Christopher Aloi wrote:
Hello List,
Question: Has anyone been able to create multiple queue_log files in
/var/log/asterisk for multiple queues?
We are designing a multi-tenant system and separating the log files
would be useful, instead of dropping all queue
On Wed, 2006-10-25 at 13:31 -0500, Jim Freeze wrote:
Hi
I am looking for a good wirless headset to use with the Polycom Soundpoint 501
phone. I would greatly appreciate hearing from anyone with good experiences
with a specific device.
Thanks
We've used the Plantronics CS50 wireless
I like the Plantronics CS55/HL10, it's a DECT Wireless boom headset with
a lifter kit for the phone, works like a charm, great range.
-Cory
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Freeze
Sent: Wednesday, October 25, 2006 2:31 PM
To: Asterisk
In your configuration files, for the providers, put:disallow=allallow=g729For the phones leave them as it is, they might use G711 between the phones and the server, but if its a local lan it really wont matter unless its not well designed and managed.
On 10/24/06, Wildheart [EMAIL PROTECTED]
On 10/25/06, Andrew Joakimsen [EMAIL PROTECTED] wrote:
I've used the Plantronics ones, similar to these:
http://www.plantronics.com/north_america/en_US/products/cat29880043/cat29880058/prod5510016
and they work very well with the headset lifter, The range is pretty good
too.
However there are
PJ Wrote:
Dan, can you supply your ooh323.conf for me? I would like resolve my
issue with not recognizing dtmf by ooh323 from callmanager
my ooh323 is quite simple, also on callmanager config page for gateway
to asterisk is nothing special, no faststart, no mtp; ccm v4.1.3sr3a
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