Hi marek,
Thanks for the update.
I have Sangoma A104D and wanted to use ss7 signalling. I came accross
chan_ss7 but found sifira is not in active development. But is this
chan_ss7 stable and can be used in production server implementation.
We are going to have 2 to 3 boxes with ss7 signalling
T.38 will not work with the fxo card.
Zoa
Fernando Berretta wrote:
Dear All,
Are you telling me Asterisk 1.6.0b2/4 has support for t38 and rxfax
etc. and will be able to receive faxes and negotiate with voip CPE's
like ATA's to transmit faxes which comes from FXO cards to VoIP
Devices
Hi Clive,
Hi all,
Do some one experiencing running jabber applications (jabberstatus...) in
asterisk? I do experinced Asterisk 1.4.18 and wish to start it, however I
got such result.
IBM*CLI help jabber
No such command 'jabber'.
IBM*CLI help jabberstatus
No such command 'jabberstatus'.
I have spent some time this morning trying to add an Astribank to our
current Asterisk, but it failed, so I just removed the hardware,
restore the config files to the original setup and started asterisk.;
I could see that no Zap channels are started so I did load chan_zap.so:
pbx*CLI module load
I forgot to mentio asterisk log this 2 errors:
[Feb 26 08:38:01] ERROR[30245] chan_zap.c: Unable to get parameters
[Feb 26 08:38:01] ERROR[30245] chan_zap.c: Unable to register channel '1-15'
Any hint?
Thanks in advance.
Andres
On Tue, Feb 26, 2008 at 10:44 AM, Andres Jimenez [EMAIL PROTECTED]
Now it takes about 25 seconds after dialing number to make asterisk ready to
answer call with Unicall. I think that this stuck is because of timeout in
ANI request.
Here is my log.
[EMAIL PROTECTED] ~]# cat /var/log/asterisk/full | grep unicall
[Feb 26 11:18:40] WARNING[2968] chan_unicall.c:
On Tue, Feb 26, 2008 at 11:06:29AM +, Andres Jimenez wrote:
I forgot to mentio asterisk log this 2 errors:
[Feb 26 08:38:01] ERROR[30245] chan_zap.c: Unable to get parameters
[Feb 26 08:38:01] ERROR[30245] chan_zap.c: Unable to register channel '1-15'
Any hint?
Here's my guess:
You
Hello!
I just encountered a strange thing in my mysql cdr records. From a
certain date on Asterisk (1.4.6) stopped to populate the CLIR and SCR
flieds in the cdr table. As far as I know no changes happened to the
system on that date and until then CLIR are recorded properly.
The CLIR is still
Hi. Since I am stuck with kernel 2.6.24, is there any way to compile
zaptel 1.4.7.1 under kernel 2.6.24? I tried using
make KBUILD_NOPEDANTIC=1 -- however this does not compile. Any other
suggestions for this and can I still use the latest version of
asterisk if I do this successfully?
Thanks.
Note: [Urgent] is generally not a good way to escalate the issue on a
public mailing list. We're all here for the fun of it and demanding
prompt reply may actually serve the other way.
If you have paid to get support (e.g: by buying hardware), this may be a
good time to use it.
That said, your
On Tue, Feb 26, 2008 at 06:45:15AM -0500, John covici wrote:
Hi. Since I am stuck with kernel 2.6.24, is there any way to compile
zaptel 1.4.7.1 under kernel 2.6.24? I tried using
make KBUILD_NOPEDANTIC=1 -- however this does not compile. Any other
suggestions for this and can I still use
What's your output from 'ztcfg -vv'?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andres Jimenez
Sent: 26 February 2008 01:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [URGENT] Zap channels fail to
On Tue, Feb 26, 2008 at 12:21 PM, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
Note: [Urgent] is generally not a good way to escalate the issue on a
public mailing list. We're all here for the fun of it and demanding
prompt reply may actually serve the other way.
I am sorry about the scalating,
Comes from a previous message.
On Tue, Feb 26, 2008 at 12:25 PM, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
Here's my guess:
You built Asterisk vs. a newer Zaptel (that happened to have the
Astribank drivers).
Now you reverted to the old Zaptel drivers. And those are of a version
before
Benny Amorsen wrote:
Steve Underwood [EMAIL PROTECTED] writes:
Try reading the GPL and the FSF's interpretation of it. If things are
running in the same address space as my code, they need to be GPL
compatible, or I am likely to take action.
The GPL is not an EULA. You don't
Well, I don't have a 701 extension defined but I do have _XXX which is where
this call is jumping when I dial 701 to pickup.
I have the include = parkedcalls above the _XXX definition, so I assumed
that parked calls would be matched first. As well, since the 700 is
matching to parked calls, I
On Tue, Feb 26, 2008 at 12:05 PM, Louwrens Benadé [EMAIL PROTECTED] wrote:
What's your output from 'ztcfg -vv'?
pbx:~# ztcfg -vv
Zaptel Version: 1.4.8
Echo Canceller: MG2
Configuration
==
SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
Channel map:
Channel 01:
Tzafir,
I'm sorry, my question wasn't clear.
Apparently Asterisk 1.6.0b2 and b4 has support for t38 because of some
modifications on app_fax so the questions are:
1 - If I use Asterisk 1.6.0b2 o b4 and a fax is received from a FXO Card
and this FXO port is forwarded to other ATA/Gateway is
zoa wrote:
T.38 will not work with the fxo card.
Zoa
That statement is a bit vague. What has been put in add-ons so far is
only support for T.38 termination. Not T.38 gateway operation.
Steve
Fernando Berretta wrote:
Dear All,
Are you telling me Asterisk 1.6.0b2/4 has support for
i have 2 asterisk servers one on CentOS and one on Fedora , i configured IAX
trunking between the 2 servers so that i dial -say from a sip extension 2000 on
fedora server to a sip extension 3000 on CentOS server the call seems to be
established but hangup automatically after very short time
Thanks for clarify.. so Asterisk will be able to receive faxes which
comes from a Gateway using t38 but will not be able to relay faxes which
comes from PSTN through a FXO card to other Gateway using t38
can this version of app_fax be used with Asterisk 1.4x ?
Steve Underwood wrote:
zoa
Fernando Berretta wrote:
Tzafir,
I'm sorry, my question wasn't clear.
Apparently Asterisk 1.6.0b2 and b4 has support for t38 because of some
modifications on app_fax so the questions are:
1 - If I use Asterisk 1.6.0b2 o b4 and a fax is received from a FXO
Card and this FXO port is
I am getting this strange error:
make[1]: Entering directory `/usr/src/zaptel-1.4.7.1'
make -C /lib/modules/2.6.24-gentoo-r2/build SUBDIRS=/usr/src/zaptel-1.4.7.1
HOTPLUG_FIRMWARE=yes modules
make[2]: Entering directory `/usr/src/linux-2.6.24-gentoo-r2'
CC [M] /usr/src/zaptel-1.4.7.1/wcfxo.o
On Tue, Feb 26, 2008 at 07:50:09AM -0500, John covici wrote:
I am getting this strange error:
make[1]: Entering directory `/usr/src/zaptel-1.4.7.1'
make -C /lib/modules/2.6.24-gentoo-r2/build SUBDIRS=/usr/src/zaptel-1.4.7.1
HOTPLUG_FIRMWARE=yes modules
make[2]: Entering directory
thx guys , i think i discovered the problem , it seems that i had to put the
host=192.168.0.x in iax.conf and not host=dynamic ,,otherwise had to register
the clients
From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Tue, 26 Feb 2008 15:03:25
+0200Subject: [asterisk-users] iax trunking problem
Why does this look suspiciously like a T1 line? Are you sure this is a
fractional E1?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andres Jimenez
Sent: 26 February 2008 02:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I do have the newer one installed -- is this the problem?
And why this weird error anyway?
on Tuesday 02/26/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote
On Tue, Feb 26, 2008 at 07:50:09AM -0500, John covici wrote:
I am getting this strange error:
make[1]: Entering directory
Hi all,
I am having a strange problem. I am using my asterisk server AST1 to
register with another asterisk server AST2 using 2 accounts (2 register
commands in sip.conf). I have made 2 local users in AST1, and configured my
dialplan in such a way that both local accounts on AST1 use different
Slightly edited your message:
On Tue, Feb 26, 2008 at 08:48:13AM -0500, John covici wrote:
on Tuesday 02/26/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote
On Tue, Feb 26, 2008 at 07:50:09AM -0500, John covici wrote:
I am getting this strange error:
make[1]: Entering directory
On Tue, 2008-02-26 at 07:44 -0500, OCG Technical Support wrote:
Well, I don't have a 701 extension defined but I do have _XXX which is where
this call is jumping when I dial 701 to pickup.
I have the include = parkedcalls above the _XXX definition, so I assumed
that parked calls would be
Hi list,
I'm wondering if it's possible to transfer a call that is still ringing???
Actually, the problem is that my telco provider doesn't offer an uniform
method for answer/disconnection supervision, and by that I mean, some of
it's line (I think) offer a polarity reversal, but other lines (of
On Mon, 2008-02-25 at 21:03 -0500, Michelle Dupuis wrote:
I have 2 contexts in my extensions.conf: internal and external calls. I
have included the parkedcalls context in both.
Do I need to preface the include with a # symbol?
No, you do not. You simply need a like that says:
include =
OK that worked, but how can we resolve it without need
to type the command manually, as the destination might
change its IP address without our notice, so the
question is:
How can the host be updated periodically (like
externrefresh settings), but need it for host, any
help?
Regards
Bilal
It looks like I have a conflict! (See results of diaplan show below). How
can I force the parkedcalls context to be matched first? (I include
parkedcalls before I define the _X. priority).
pbx*CLI dialplan show [EMAIL PROTECTED]
[ Context 'entryocginternal' created by 'pbx_config' ]
'_X.' =
OK, here is the problem -- how do I compile 1.4.7.1 using kernel
2.6.24 -- when I try the make KBUILD_NOPEDANTIC=1 I get the no such
file or directory that I already mentioned -- when I take out that
KBUILD option, I get what I got before -- the error from the kernel
module build about the CFLAGS
Another clue...I repeated the dialplan show command for the 700 extension
and it too is listed AFTER the _X. match. However, forward a call to 700
works. Why would calling 701 not pickup the call? (Why is it matching the
_X. extension)
Thanks!
pbx*CLI dialplan show [EMAIL PROTECTED]
[ Context
Ian,
I'm having *THE SAME PROBLEM* and I've noticed that when a transfer fail
(only happens when receptionist dial an external number) the call is marker
as NO ANSWER in the CDR, even when the call *HAS BEEN ANSWERED* by the
other party (the callee). See my previous post below.
bilal ghayyad wrote:
OK that worked, but how can we resolve it without need
to type the command manually, as the destination might
change its IP address without our notice, so the
question is:
How can the host be updated periodically (like
externrefresh settings), but need it for host, any
On Tue, 2008-02-26 at 10:01 -0500, OCG Technical Support wrote:
It looks like I have a conflict! (See results of diaplan show below).
How
can I force the parkedcalls context to be matched first? (I include
parkedcalls before I define the _X. priority).
pbx*CLI dialplan show [EMAIL
On Tue, Feb 26, 2008 at 1:35 PM, Louwrens Benadé [EMAIL PROTECTED] wrote:
Why does this look suspiciously like a T1 line? Are you sure this is a
fractional E1?
My provider names the line a PRA, but this is understood anywhere as a
PRI (no fractional).
From the Asterisk configuration point of
checking wheather my mail goes to asterisk users mailling list or not
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On Tue, Feb 26, 2008 at 10:59 AM, Joel Solanki [EMAIL PROTECTED] wrote:
checking wheather my mail goes to asterisk users mailling list or not
ACK.
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asterisk-users mailing list
Hello,
I am trying to add a sip-trunk to my Asterisk 1.4.15/Elastix 0.9.2 server.
The system is in production with local extensions, a zap trunk and a
working sip trunk with sipgate.de.
My asterisk server is behind a NAT/Firewall, anyhow it registers and works
well with sipgate.de on incoming
- Original Message -
From: Jared Smith [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 19, 2008 11:53 PM
Subject: Re: [asterisk-users] Attatch monitor recording to a voicemail
On Tue, 2008-02-19 at
Hi List,
While I know that upping ulimit will fix the issue I am trying to understand
what will cause it. I have a few set ups that are almost exactly the same yet
some machines used to give this error often and others don't. I also noticed
the error a lot more on my boxes running 1.4.X.
TIA.
This smells of NAT issues. Since you said that you do have the server set up
as a DMZ did you set NAT=yes, externip= ? as well as the other NAT settings
?
- Original Message -
From: Sanjoy Rath [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, February 05, 2008 10:32
On Tue, 2008-02-26 at 19:48 +0200, Dovid B wrote:
Jared,
You mentioned that you have done it in the past.Can you post your code here
No, unfortunately this was done under NDA, but the general gist goes
like this:
Dialplan pieces:
A) Get automon working. Don't forget to set the
On Tue, 2008-02-26 at 19:53 +0200, Dovid B wrote:
While I know that upping ulimit will fix the issue I am trying to
understand what will cause it.
There are *lots* of things in Asterisk that open file handles, and to
try to track them down is probably a waste of time. Just configure your
On Tue, Feb 26, 2008 at 12:31 PM, Dirk Enrique Seiffert
[EMAIL PROTECTED] wrote:
Hello,
I am trying to add a sip-trunk to my Asterisk 1.4.15/Elastix 0.9.2 server.
The system is in production with local extensions, a zap trunk and a
working sip trunk with sipgate.de.
My asterisk server
Thanks,
Joel
On Tue, Feb 26, 2008 at 10:47 PM, Erik Anderson [EMAIL PROTECTED] wrote:
On Tue, Feb 26, 2008 at 10:59 AM, Joel Solanki [EMAIL PROTECTED]
wrote:
checking wheather my mail goes to asterisk users mailling list or not
ACK.
___
--
We have an Asterisk server with a small outgoing call center. We use
AMD and it usually works very well on Zap channels (E1 PRI). We added a
couple of SIP trunks to reduce long distance costs but now AMD gets
stuck when the call goes out through the SIP channels. Here is an
example call
- Original Message -
From: Jared Smith [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 26, 2008 8:11 PM
Subject: Re: [asterisk-users] Explain Cause of Error: manager.c:Accept
returned -1: Too many
- Original Message -
From: Jared Smith [EMAIL PROTECTED]
To: Dovid B [EMAIL PROTECTED]
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 26, 2008 8:08 PM
Subject: Re: [asterisk-users] Attatch monitor recording to a
Add an answer() and a playback of 1 second of silence or something else
to make sure the RTP is nailed up. AMD can/will hang if it has no media
to analyze.
Carlos Chavez wrote:
We have an Asterisk server with a small outgoing call center. We use
AMD and it usually works very well on
Greetings,
How can I call cheap to UK cell phones. I am located in Toronto, Canada, but
need to call UK cell phones both from Toronto and London.
--
Zeeshan A Zakaria
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Ours have been running fine since pointing the aastra.cfg to the LAN NTP.
Don't know what can be happening with yours.
On Tue, Feb 26, 2008 at 12:52 AM, Marius Muja [EMAIL PROTECTED] wrote:
There already is an ntp server on the LAN, but the phones still freeze.
On Mon, Feb 25, 2008 at 2:18
On Tue, Feb 26, 2008 at 1:19 PM, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
Greetings,
How can I call cheap to UK cell phones. I am located in Toronto, Canada, but
need to call UK cell phones both from Toronto and London.
I'd guess you could get an account with one of these providers:
Hi Steve,
Does it retransmit the invite six times and then hangup? When I have
seen this it was a firewall issue on the remote (provider) side.
Indeed it tries seven times. But I think this is the Asterisk default. The
same account configured in my Snom Phone works without problem, - from
Hi marek,
Thanks for the update.
I have Sangoma A104D and wanted to use ss7 signalling. I came accross
chan_ss7 but found sifira is not in active development. But is this
chan_ss7 stable and can be used in production server implementation.
We are going to have 2 to 3 boxes with ss7 signalling
Dovid B wrote:
Thanks. I like to know my errors and what cause them. Anyone available to
help me pick at their brain to see where its coming from or am I really
barking up the wrong tree ?
Dovid,
The number of concurrent calls on the server is tightly related to the
number of file
I've had it with Dell server garbage.They seem to change RAID
controllers as much as I change socks, and then the controllers don't work
with Linux, unless you load a new driver.They sell servers with a PCI-e
slot in them, but then you get it and find out the RAID controller is using
the
On Tue, Feb 26, 2008 at 3:10 PM, Matt [EMAIL PROTECTED] wrote:
I've had it with Dell server garbage.They seem to change RAID
controllers as much as I change socks, and then the controllers don't work
with Linux, unless you load a new driver.They sell servers with a PCI-e
slot in them,
I am running Trixbox 2.4 which has Asterisk 1.4.18-1
I have kind of followed:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38
I added to sip_general_custom.conf
;NEEDED!!!
t38pt_udptl = yes
I did not add this to the actual SIP extension, as I assumed this being
general it
On Tue, Feb 26, 2008 at 3:20 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Tue, Feb 26, 2008 at 3:10 PM, Matt [EMAIL PROTECTED] wrote:
I've had it with Dell server garbage.They seem to change RAID
controllers as much as I change socks, and then the controllers don't work
with Linux,
On Tue, Feb 26, 2008 at 2:10 PM, Matt [EMAIL PROTECTED] wrote:
I've had it with Dell server garbage.They seem to change RAID
controllers as much as I change socks, and then the controllers don't work
with Linux, unless you load a new driver.They sell servers with a PCI-e
slot in them,
I have gotten this to work and am glad to see an open source solution.
I have tested it as a gateway to our asterisk IVR.
I will need to have three to five skype instances running to keep away from a
potential busy signal.
Does anyone know of a good solution for this?
Does anyone know of a
2.6 CentOS 4
I can't speak to the PCIe issue, but I've never in my life had
compatibility issues with the Dell RAID controllers. What kernel are
you on?
Can anyone recommend an IBM or Gateway server that you have used with
Asterisk and are happy with, and which will support RAID-1 or
On Tue, 2008-02-26 at 12:31 -0500, Dirk Enrique Seiffert wrote:
I aquired an account with a reseller net-voz.com: I did some testing with
the account directly from a Snom300 phone - works without a problem,
(behind the nat) I spent hours testing and adjusting the trunk
configuration for
I had a server built for me by J and N Computer Services
http://www.jncs.com which is using a Super Micro c2sbe MB which I
think has what you need plus 4 PCI-32 slots! Its a nice MB and I have
an e8400 cpu in it.
on Tuesday 02/26/2008 Matt([EMAIL PROTECTED]) wrote
I've had it with Dell
Steve,
I have fielded several hundred Asterisk and related VoIP boxes.
I buy SuperMicro 1-U units mostly. I have also used their larger
units with RAID and a full load of ULTRA SCSI (for MySql application).
I like these because, after bad experience with DELL/COMPAQ/HP/IBM
compatibility issue,
Jared Smith wrote:
No, unfortunately this was done under NDA, but the general gist goes like
this:
As it happens, I deployed my solution to this on our live PBX today, which I
wrote with some help from another asterisk-users user. Here is what I
came up with:
Firstly, in features.conf I
I haven't bought from them recently, but I also have bought many servers
and desktop systems from J N. I have at least 3 servers they built that
are over 8 years old and still running in production. I've bought like 8
servers, and a half-dozen desktop systems from them since around 1996 or
I second Sun and supermicro.
Sun was really cool on the management facilities, the linux
compatibility and the speed was nice too.
Supermicro (opteron series) always amazes me how fast they are. They
really *feel* fast ;)
Only ever used support on supermicro and it was excellent. My box froze
Hi Jared,
Notice how the Contact Header and the SDP all have the IP address of
192.168.8.3? If your firewall isn't masquerading (rewriting) those
addresses as the SIP traffic goes through it, then the device on the
other end is going to try to contact 192.168.8.3, and I'm guessing it's
Posting the same question three times at 12 hour intervals will not get
you a faster reply.
Especially not to an original email that was written in November of last
year.
Joel Solanki wrote:
Hi marek,
Thanks for the update.
I have Sangoma A104D and wanted to use ss7 signalling. I came
On Feb 5, 2008, at 3:32 PM, Sanjoy Rath wrote:
I have a asterisk server. Two SIP Soft XLites are connected to the
server. I am able to make
calls from one SIP Phones to the other SIP Phones and landlines
successfully. The SIP Soft Phone on th eother side can hear my voice
but I cannot
Do you mean that an agent on the phone doesn't know if there are
callers in line behind the current one or not? I had that question as
well. Perhaps not the best way, but I solved it by taking an extra
phone and setting it up as a static agent in the closet. Then I made
a light on my
We currently using the x service servers as well, never had any problems with
them.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, 27 February 2008 7:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Feb 26, 2008, at 4:13 PM, [EMAIL PROTECTED]
wrote:
On Tue, Feb 26, 2008 at 3:10 PM, Matt [EMAIL PROTECTED] wrote:
I've had it with Dell server garbage.They seem to change RAID
controllers as much as I change socks, and then the controllers
don't work
with Linux, unless you load a
Particularly WRT T.38 fax.
Supposedly, when fax tones are detected, Asterisk is to do a reinvite
asking for T.38.
Here is what I am using in my dialplan:
[custom-fax1]
exten = s,1,Answer
exten = s,n,StopPlayTones
exten = s,n,Set(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten =
Okay, T1 card issue sorted out. New Lesson: Stay Away from TigerJet chips.
Next up, modem pool -- I wanted to know if the below config looked anywhere
near half-sane for defining in asterisk what is essentially a small pool of
four waiting modems that will handle faxes if another modem is
Just don't use T1 cards w/ TigerJet chipsets in them on DL385's (and very
likely, 380's as well). I just learned this the hard way.
--J
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Norman Franke
Sent: Tuesday, February 26, 2008 5:27 PM
To:
Is there something equivalent to SipT38SwitchOver in Asterisk (in
callweaver)...
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Hi,
I tried 3 times to send this message. It goes out but i dont recieve mail
sent on asterisk-users@lists.digium.com but when someone replies to that
email i recieve the email like you did.
I thought mails were not going to mailling list to tried 3 times. But it is
strange. As far as i know if
Tzafrir Cohen wrote:
On Sun, Feb 24, 2008 at 05:38:43PM -0700, James Finstrom wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Sean,
I believe the alarm is generated by the bits flipping .
In kewl is hangup so every time you hang-up you could
potentially alarm.
That is:
Sangoma cards work a treat in a HP DL380 or 320/260 for that matter.
I just like having two power supplies and hot swap RAID 5 plus a few
extra slots.
Thanks,
Steve Totaro
On Tue, Feb 26, 2008 at 5:51 PM, Joshua Kinard [EMAIL PROTECTED] wrote:
Just don't use T1 cards w/ TigerJet chipsets in
On 27/02/2008, Joel Solanki [EMAIL PROTECTED] wrote:
I tried 3 times to send this message. It goes out but i dont recieve mail
sent on asterisk-users@lists.digium.com but when someone replies to that
email i recieve the email like you did.
I thought mails were not going to mailling list to
On Tue, Feb 26, 2008 at 01:38:31PM -0600, Erik Anderson wrote:
On Tue, Feb 26, 2008 at 1:19 PM, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
Greetings,
How can I call cheap to UK cell phones. I am located in Toronto, Canada, but
need to call UK cell phones both from Toronto and London.
I'd
Alan [EMAIL PROTECTED] writes:
I have a small VPS server in www.eapps.com and im doing some research
in order to install Asterisk in that server..
Does anybody has installed Asterisk in a Virtuozzo VPS System??
I have done so, with success, for a SIP-only installation. Well, into
OpenVZ
If I am originate from Canada, how can I benefit from these cheap rates?
On Tue, Feb 26, 2008 at 7:20 PM, Steve Kennedy [EMAIL PROTECTED]
wrote:
On Tue, Feb 26, 2008 at 01:38:31PM -0600, Erik Anderson wrote:
On Tue, Feb 26, 2008 at 1:19 PM, Zeeshan Zakaria [EMAIL PROTECTED]
wrote:
You can interrogate the SIP information for some of this using the SIP debug
command on the CLI along with the udptl debug command. It's not perfect but
it works for what you're looking for.
On Tue, Feb 26, 2008 at 3:21 PM, Robert Moskowitz [EMAIL PROTECTED]
wrote:
I am running Trixbox 2.4
Just a heads up, the echo cancellation problem disappeared with Asterisk
1.4.15, zaptel 1.4.8, and libpri 1.4.3.
Still having other problems with the TE120P, but all OT from echo
cancellation.
On Mon, Feb 25, 2008 at 7:45 PM, arkda [EMAIL PROTECTED] wrote:
Sorry, 1.4. Keep forgetting 1.2 is
John Covici wrote:
Hi. I am using asterisk 1.4 (latest as of today) and zaptel 1.4
(latest as of today) and I cannot dial out using my 400P card with one
fxs module and one fxo module. I am using kernel 2.6.24 and get the
following log entries:
[Feb 25 17:28:13] VERBOSE[25071] logger.c:
It should look more like this:
exten = fax,1,Dial(IAX2/iaxmodem1/${NumberCalled}|20)
exten = fax,n,Dial(IAX2/iaxmodem2/${NumberCalled}|20)
exten = fax,n,Dial(IAX2/iaxmodem3/${NumberCalled}|20)
exten = fax,n,Dial(IAX2/iaxmodem4/${NumberCalled}|20)
exten = fax,n,Busy()
-Original Message-
I updated zaptel and I can dial out, but when someone calls in it
won't hangup unless my extension hangs up, which was not true before.
This is a better state than before, thanks much for fixing so far.
on Tuesday 02/26/2008 Shaun Ruffell([EMAIL PROTECTED]) wrote
John Covici wrote:
Hi. I am
- Original Message -
From: Atis Lezdins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 05, 2008 4:24 PM
Subject: Re: [asterisk-users] one CDR instead of multiple CDR
On 2/5/08, Arjan Kroon |
sean darcy wrote:
Tzafrir Cohen wrote:
On Sun, Feb 24, 2008 at 05:38:43PM -0700, James Finstrom wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Sean,
I believe the alarm is generated by the bits flipping .
In kewl is hangup so every time you hang-up you could
potentially
Well, on an E1 PRI config your D-channel is indeed assigned to channel 16,
the center channel. On a T1, your data channel is on channel 24, the last
channel.
Did you restore your zaptel config from samples or another source?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hello,
It has happened to me twice now that duplicated voicemail messages are
automatically created, every minute.
I have been unable to reliably repeat it (so far), but the basic flow
seems to be:
1. a call comes in via my TDM400P (PSTN line)
2. the call is not answered and goes to voicemail
Hello,
I'm trying to set up call recording for a queue. Right now the
recording appears to work correctly, but when I call and chatter for a
minute or so, at the end of the call I end up with a very small file
(less than 100 bytes), which contains about .06 seconds of silence.
If I talk for
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