On Wed, 24 Jan 2007, Yuan LIU wrote:
I have a really dumb question. It appears that Yahoo, MSN, AIM, you name
them, they don't have a NAT problem, and some use SIP. I don't think they
all stay in voice path, either. What takes?
Their SIP servers aren't behind NAT firewalls, so the problem
On Wed, 2007-01-24 at 11:26 -0800, Serge Blazhievsky wrote:
Hello everybody,
I was wondering if anybody knows how to make channel IDs different if
all call are coming from the same host:
core show channels
Channel Location State Application(Data)
My experience has been to be consistant. The only time I have had
problems with DTMF is when I am not using the same DTMF encoding
technique on all hardware. Your choices are: INFO, RFC2833 or
INBAND. Some equipment also has an AUTO option but I would not
recomend it. Stick with INFO or
True the Panasonic will need to be told to trunk a new extension range out
over the ISDN for the gateway to pickup but this seems a lots less hassle
and everything remains SIP.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: 24 January 2007
Hi, how do I get Asterisk to start in very verbose mode every time it boots?
Neil
safeharbour IT Ltd
Your IT Department
fax: 0845 867 2891
mob: 07812 114784
voip: [EMAIL PROTECTED]
email: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED]
web: http://www.safeharbourit.co.uk/ www.safeharbourit.co.uk
asterisk.conf
[options]
verbose = 3 ; Verbosity level for
logging (-v)
Neil Tancock wrote:
Hi, how do I get Asterisk to start in very verbose mode every time it
boots?
Neil
___
--Bandwidth and Colocation
-Original Message-
Gordon Henderson
Sent: 25 January 2007 08:17
On Wed, 24 Jan 2007, Yuan LIU wrote:
I have a really dumb question. It appears that Yahoo, MSN, AIM, you
name them, they don't have a NAT problem, and some use SIP. I don't
think they all stay in voice
That's interesting, as I've still not managed to completely resolve the
problem.
I've managed to reduce it by upgrading to Zaptel-1.4 and rerunning fxotune, but
there is
still a distinct crackle, which is more noticeable on calls to mobiles.
I am yet to try removing the hardware echo module,
Hello, everyone.
I'd like to ask how does asterisk 1.4 with GUI register itself at the
provider's end (when I mark a checkbox 'register' while creating a
Service Provider). Before I used to write something like:
register = 924980111:[EMAIL PROTECTED]/924980111
in sip.conf. Having that line,
Has anyone seen this issue with asterisk running like a dog when the
internet is down ? Internal calls, incoming ISDN calls etc all seem to be
affected. There is a local DNS server that is always available so I’m not
sure why asterisk is so unresponsive.
I’ve seen this on two different
Peter Mitchell wrote:
Has anyone seen this issue with asterisk running like a dog when the
internet is down ? Internal calls, incoming ISDN calls etc all seem to
be affected. There is a local DNS server that is always available so
I’m not sure why asterisk is so unresponsive.
I’ve seen
http://www.youtube.com/watch?v=ONOxNJquatk
On 1/23/07, Dovid B [EMAIL PROTECTED] wrote:
Link please ?
Ooops!, sorry
--
Damián D. Fossi Salas
¡Software Libre hasta el 2 mil siempre!
Uso:
Debian Etch Kernel 2.6.18-3-686
Ubuntu Edgy Eft Kernel 2.6.15-27-amd64
Ulanix 0.4-14 Kernel
I've got a question regarding Cisco IP Phones and licencing.
When using a third party PBX like asterisk is a licence required for the
Cisco phones ? Has anyone got anything in writing from Cisco to clarify this
?
Eg can I just use CP-7961G or do I need CP-7961G-CH1 even though I'm not
using
Hi,
I have an Asterisk systems setup with a Channel bank to serve a number of
analog telephone handsets, aside the IP phones and ATAs that associate with
the asterisk.
A queuing group with a global number for a group of extension numbers is
configured[Global number = 9000; Extensions in
Hi
I've got an Asterisk box connected to a Siemens Hicom 300 using a Digium
TDM400. The Hicom provides the calling extension as DTMF at the beginning of
the call followed by two *, as in 3425** when 3425 calls my extension, I can
hear all 6 tones if I have a handset connected but using Asterisk's
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Try safe_asterisk , for an easy way to start asterisk in background,
a plain 'asterisk' is even better and safer.
asterisk -U asterisk . is better.
/etc/init.d/asterisk start
is similar.
Why is this better than safe_asterisk?
Andrew Joakimsen ha scritto:
Actually I noticed just three days ago there is a new release, and the
releae notes seem to address
Disable TrMail and Pickup keys
Disable call progress indication
___
but it does not address poor guys' troubles with
On 1/25/07, George C. Attopany [EMAIL PROTECTED] wrote:
Hi,
description of problem cut out for brevity
member = Zap/9-1
member = Zap/10-1
member = Zap/11-1
member = Zap/12-1
member = Zap/13-1
member = Zap/14-1
member = Zap/15-1
member = Zap/16-1
I don't think you want the -1 on the
I think, ci$co phones can not be even purchased without licence...
btw, what is your reason, to buy ci$co phones, when known issues exist
with this phones, if working with anything other than callmanager? :-\
PJ
Peter Mitchell wrote:
I've got a question regarding Cisco IP Phones and
I second that request
On 1/25/07, Kenneth Padgett [EMAIL PROTECTED] wrote:
I ran into this problem with an early batch of IP650s. Polycom's
firmware
version 2.0.3b made this issue go away.
Speaking of Polycom firmware, anyone have an up to date source for the
stuff? The site I ordered from
Quoting Nick Whitaker [EMAIL PROTECTED]:
The problem I'm
having
is the only PCI slot shares an IRQ with the SATA controller. Any
altering of one device's IRQ takes the other device's IRQ with it in
lockstep.
Nick, the word from Dell is that SC stands for Simplified
Configuration and there is
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi,
i can see AOC messages on the asterisk console. Can i sendtext() them to the
caller or put them in cdr?
Regards, Andreas.
I'm also interested in this. If you find solution, please mail it to the list.
--
Tomislav Parcina
I have read that 1.4 has shared line appearances, which I assume will
work with Polycom phones. Has anyone configured this and verified it
working? I was going to start playing around with it but wanted to see
if anyone else has tackled it yet.
Bill
I think it can be done, but not with a GrandStream HandyTone ATA because
the manual says this:
What it CANNOT do:
- Terminate a VoIP call into the PSTN port
- Allow a call from PSTN to route other VoIP devices (different from the
FXS phone) over the IP network
- Automatically route calls made
I'm planning a new * system which will utilize 48 stations (Polycom
Soundpoint 501s mostly) and a dual span PRI card and I have some questions.
The system will host MeetMe conferences of 10-15 users on a regular basis
and see fairly high usage as it is going into a medical setting.
1. I haven't
Hi all,
I'm currently on an NTL PSTN line, using Asterisk 1.2 and a X101P cheapo
card.
The problem lies with detecting when the far end has hung up. It fails
to detect it, and will only cleardown when the silence timeout has been
reached. Now, I've seen the thread at
I posed the same question to both our Cisco partner and direct to our Cisco
rep. Neither one could tell me what I would not be able to do with a
non-licenses IP phone. As you probably know the phone will work with out a
license, but that may not be acceptable in Cisco's eyes.
In the end we went
There was talk last week that SLA in 1.4 was not working correctly and was
being rewritten for a 1.4.1 release.
On 1/25/07, Bill Gibbs [EMAIL PROTECTED] wrote:
I have read that 1.4 has shared line appearances, which I assume will
work with Polycom phones. Has anyone configured this and
On Fri, 12 Jan 2007 07:14:08 -0800 Kevin P. Fleming wrote:
I won't waste your time, because the current SLA implementation is
broken. We expect to have replaced it when Asterisk 1.4.1 is released,
and there will be better documentation at that point as well.
Technically, Cisco requires you to purchase both a Smartnet (To obtain a
CCO login for access to firmware), as well as a SIP/MGCP license token,
to utilize their phones with SIP firmware, regardless of platform.
The CH1 nomenclature applies to Callmanager, the CCME nomenclature
applies to
Hi!,
this issue makes me crazy. I read a lot of docs, also * mailling list
and I try a lot of things without success.
Any help will be appreciated. Here is the info:
Hardware:
Supermicro Server with motherboard X7DB8, chipset Intel 5000P, Xeon
Hi David,
I removed the 1s as suggested, but it did not work.
Well, I have halted the beep/bleak/single-ring somehow, by taking down the
inter-tie line between my asterisk and a PANASONIC TDA200.
What I have in place are:
ASTERISK SIDE: ASTERISK PBX 1.2 + Adit600 Channel bank(which gives
Hi,
I had the same crackle problem with the same hardware.
Actually for me was a shared IRQ problem.
Now that I fixed it the situation is much better (maybe not perfect but
anyway good).
Maybe this IRQ are affecting more the Echo Module than the card?
Ciao
On Thu, Jan 25, 2007 at 01:37:50PM +0100, Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Try safe_asterisk , for an easy way to start asterisk in background,
a plain 'asterisk' is even better and safer.
asterisk -U asterisk . is better.
Thanks, I had a notebook crash and must have missed that. Appreciate the
replies! I will be patient.
Bill
-Original Message-
From: [EMAIL PROTECTED] on behalf of Steve Langstaff
Sent: Thu 1/25/2007 9:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
(IAX2/4427-1, record-enable|4427|OUT) in new stack
-- Executing GotoIf(IAX2/4427-1, 0 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(IAX2/4427-1,
recordingcheck|20070125-102531|1169738731.2435) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin
Hi,
I'm analyzing freepbx extensions. When creating ivr with freepbx, it
writes like this:
exten = ,1,Answer
exten = ,n,GotoIf($[${CONTEXT}=from-internal]?USERCID:SETCID)
exten = ,n(USERCID),Macro(user-callerid,)
exten = ,n(SETCID),Set(CALLERID(name)=${CALLERIDNAME})
exten =
I know of the call pickup issues but what asterisk issue and what BLF issue?
On 1/25/07, Alberto Pastore [EMAIL PROTECTED] wrote:
Andrew Joakimsen ha scritto:
Actually I noticed just three days ago there is a new release, and the
releae notes seem to address
Disable TrMail and Pickup keys
Hi...
In my dialplan, I have the following:
exten = s,1,Background(${RECORDING}|m)
exten = s,n,Voicemail(${DID_NO})
exten = 0,1,Voicemail(${DID_NO})
exten = a,1,VoiceMailMain(${DID_NO})
exten = h,1,Hangup
In version 1.2, when I hit 0 during the playback, I will be directed
to voicemail. But
On 1/25/07, Peter Mitchell [EMAIL PROTECTED] wrote:
Has anyone seen this issue with asterisk running like a dog when the
internet is down ? Internal calls, incoming ISDN calls etc all seem to be
affected. There is a local DNS server that is always available so I'm not
sure why asterisk is so
Hi,
when I build zaptel-1.2 and asterisk-1.2 I can modprobe ztdummy and
start asterisk to be able to use MeetMe().
When I build zaptel-1.4 and asterisk-1.4 I can modprobe ztdummy and
start asterisk but I am not able to use MeetMe().
What do I miss?
Stefan
--
amooma GmbH - Bachstr.
On Wed, Jan 24, 2007 at 11:09:21PM -0800, Yuan LIU wrote:
From: Brad Templeton [EMAIL PROTECTED]
On Mon, Jan 22, 2007 at 09:59:06AM +, Tim Panton wrote:
In the meanwhile, use IAX, which understands about NAT pretty well.
If you have multiple SIP phones on a LAN behind a NATing router,
exten = ,n,Queue(|t|||300)
exten = *,1,Macro(agent-add,,)
exten = **,1,Macro(agent-del,,)
So my question is , what means these one/two asteriks (*,**
).Maybe it is like priority.?
It means that to login as an agent on the queue you have to dial
* and
I'm having various issues that may or may not be related to each other (I'm
pretty sure they are). We've had this system for a year now (quad T1 card,
right now we have 1 T1 coming in, 2 going out to channel banks) and we've
had intermittent ghost calls--it appears that what is happening is a
Hi,
We have a very basic setup of Asterisk 1.2 with a 4 inbound line Digium card. The phones are
Grandstream GXP-2000 with the latest stable firmware.
When we get calls and put them on hold and then get a new external call coming in, it drops the
person on hold. They just get disconnected.
Hi,
does anyone have some informations on when the SVN repository of
digium.com will be synchronized again? Since few days we are sticked
with trunk #51363.
--
Daniel
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
- Original Message -
From: Paul [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 24, 2007 4:23 PM
Subject: Re: [asterisk-users] Digium Forums
Dovid B wrote:
Hi List,
Does anyone know where I can
On Wed, 2007-01-24 at 12:20 -0800, Michael Collins wrote:
__
From:[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Halliday
Sent: Wednesday, January 24, 2007 11:56
Here's how it's currently working:
1) Call comes in
2) Operator parks call (700)
3) Operator picks up call on another phone (701)
4) Operator tries to transfer to a different phone (we use #0) but the
transfer doesn't work.
We can transfer initial callers all we want and it works fine. Once
Hi all,
I am using asterisk 1.2.14 release on a 3GIG box/1 GIG RAM with a
TDM2400E card.
For the most part my echo problems are gone (I have not noticed any issue).
The problem I have is SOMETIMES I get really low audio.
This typically happens when a call is coming in to the TDM2400E card
and
Jay,
there is a bug in Mantis regarding this, a change was made to allow native
bridging of parked calls. The change has been reverted in a more recent SVN
version of 1.2. See http://bugs.digium.com/view.php?id=8804
On 1/25/07, Jay Moore [EMAIL PROTECTED] wrote:
Here's how it's currently
Hello
I have a working * server with a TDM card and 4 FXO ports.
We have 4 lines now and need to add 2 more lines (and possibly two more
later).
I'm wondering the best upgrade path for this situation.
The simplest I can invision is adding another TDM400 card with
4 FXO ports, and use 2 now and
I am doing some testing with 1.4 and the imap storage and a exchange 2003
server. I have not had any positive results so far using the notes on the
wiki or the docs in the release. My current settings are
imapserver=server
imapport=143
imapfolder=Voicemail
;imapflags=novalidate-cert
Bruce,
I'm running 1.2.14. I am not willing to switch to 1.4 yet due to the
stability issue. From what I read on the page you linked, I could not
find what version had the supposed fix. I also can't seem to find a
later 1.2 version of Asterisk (if one exists).
Any suggestions?
Thanks,
Jim,
I have 2 TDM400s in my * box (as well as a T1 card). I use all 8 ports,
and aside from some minor echoing during peak periods, it's running
smooth as ice.
Jay
Jim Freeze wrote:
Hello
I have a working * server with a TDM card and 4 FXO ports.
We have 4 lines now and need to add 2
On Thursday 25 January 2007 3:27 pm, Bruce Reeves wrote:
I am doing some testing with 1.4 and the imap storage and a exchange 2003
server. I have not had any positive results so far using the notes on the
wiki or the docs in the release. My current settings are
I've done some work with the
There was a recent driver fix that *might* help you. It's not in an
official 1.x.x release yet, but if you check out 1.2 from svn, you
should get the latest version of the driver with the fix.
Matthew Fredrickson
On Jan 25, 2007, at 9:15 AM, Marc Patino Gómez wrote:
Hi!,
this issue
Jay,
The proble is in both 1.2.14 and 1.4 the fix mentioned in the bug was added
to the svn revisions of both versions. If you are not wanting to switch from
1.2.14 to 1.2 svn the you can edit the features.c file and add the lines
mentioned in the notes back to the file, then make and make
I'm having the same crackle/static issue. Seems more noticeable on
outbound calls over inbound ones. I'm running a TDM2400 with 7 FXO
lines currently in use. Card is on it's own IRQ, Athlon 3200 processor,
Nvidia chipset. It's somewhat intermittent - much like my zttest
results - I'm not sure
Where can I get the latest copy of this file. I thought google found
ithere, but it doesn't compile correctly on 1.2.14. And the copy on
voip-info.org that I found initially appears to be old. It's not in
the 1.2tree.
On 1/25/07, Asterisk [EMAIL PROTECTED] wrote:
On Wed, 2007-01-24 at
Jim Freeze wrote:
Hello
I have a working * server with a TDM card and 4 FXO ports.
We have 4 lines now and need to add 2 more lines (and possibly two
more later).
I'm wondering the best upgrade path for this situation.
The simplest I can invision is adding another TDM400 card with
4 FXO
Ah, I misread. I'll probably do that and hopefully it'll fix the issue.
Thanks!
Jay
Bruce Reeves wrote:
Jay,
The proble is in both 1.2.14 and 1.4 the fix mentioned in the bug was added
to the svn revisions of both versions. If you are not wanting to switch
from
1.2.14 to 1.2 svn the you
On 1/25/07, Jay Moore [EMAIL PROTECTED] wrote:
Jim,
I have 2 TDM400s in my * box (as well as a T1 card). I use all 8 ports,
and aside from some minor echoing during peak periods, it's running
smooth as ice.
Hi Jay. Thanks for the info. Digium logged onto my box early on and
fixed some
Hi Leif
On 1/25/07, Leif Neland [EMAIL PROTECTED] wrote:
I have no experience with the TDM cards, but costwise it is not the best
solution, in my opinion.
A TDM04B (4FXO) cost around $378 at voiplink.com, while a Grandstream
GXW-4104 (also 4FXO) cost $250 at voxilla.com., and the 8FXO
Since you've done some work with Courier and Asterisk's IMAP voicemail, is
there a place you documented your findings? I'm interested in merging the
two. Is there any way to do it without having to ask all of my users for
their passwords?
On 1/25/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
I tried testing with mtest in the imap toolkit, I think that is what you
meant and it connects. The connection string
{server:143/imap/user=breeves}INBOX prompts for a password then
connects. I will keep digging the answer has to be around.
On 1/25/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Thursday 25 January 2007 4:48 pm, David Gomillion wrote:
Since you've done some work with Courier and Asterisk's IMAP voicemail, is
there a place you documented your findings? I'm interested in merging the
two. Is there any way to do it without having to ask all of my users for
their
On Thursday 25 January 2007 5:01 pm, Bruce Reeves wrote:
I tried testing with mtest in the imap toolkit, I think that is what you
meant and it connects. The connection string
{server:143/imap/user=breeves}INBOX prompts for a password then
connects. I will keep digging the answer has to be
Andrew,
I tried both, did you set the server authentication with authuser=user and
authpassword=password in the general section?
On 1/25/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Thursday 25 January 2007 5:01 pm, Bruce Reeves wrote:
I tried testing with mtest in the imap toolkit, I
Jim Freeze wrote:
Hi Leif
On 1/25/07, Leif Neland [EMAIL PROTECTED] wrote:
I have no experience with the TDM cards, but costwise it is not the
best solution, in my opinion.
A TDM04B (4FXO) cost around $378 at voiplink.com, while a
Grandstream GXW-4104 (also 4FXO) cost $250 at voxilla.com.,
On 25 Jan 2007, at 16:48, Patrick W. Foster wrote:
I've a call center using IAX softphones provided by a third party.
We've observed problems where the IAX phones seem unable to use our
PRI trunks. A sample anonymized call is provided below with the
PRI debug calls embedded. Any
From the VICIDIAL SCRATCH_INSTALL doc:
- cd asterisk-1.2.14/apps
- wget http://www.eflo.net/files/app_amd2.c
- mv app_amd2.c app_amd.c
- vi Makefile
replace this line(line 32):
app_mixmonitor.so app_stack.so
with this line:
app_mixmonitor.so app_stack.so app_amd.so
- wget
On Jan 25, 2007, at 5:38 PM, Leif Neland wrote:
A Grandstream GXW-4108 with 8FXO cost $400, almost the same as a
TDM404B fully populated 4FXO card.
I'm currently testing a GXW-4108... my verdict is still out. I've had
some problems, some minor, some major.
In the minor department, it
That's the same code as I have. It's identical. Are you using it over a
SIP channel?
Peter
On 1/25/07, Matt Florell [EMAIL PROTECTED] wrote:
From the VICIDIAL SCRATCH_INSTALL doc:
- cd asterisk-1.2.14/apps
- wget http://www.eflo.net/files/app_amd2.c
- mv app_amd2.c app_amd.c
- vi Makefile
On 1/25/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Thursday 25 January 2007 4:48 pm, David Gomillion wrote:
Since you've done some work with Courier and Asterisk's IMAP voicemail,
is
there a place you documented your findings? I'm interested in merging
the
two. Is there any way to do
Cory,
I know the 7940 and 7960 had a SIP licence you could buy. It was simple,
buy the phone and then buy the SIP licence if you want to use it for
asterisk.
79X1 phones now come bundled with licences - and I can't find a separate SIP
licence like the old 79x0 models.
Whats the non callmanager
I've got a brand new Eicon Diva Server BRI card and I want to configure
it with Asterisk. I managed to get asterisk and zaptel to compile and
install, I've compiled and installed the drivers for the Diva card and
now I need to compile and install the chan_driver for chan_capi.
Unfortunately
Kyle Gordon wrote:
Hi all,
I'm currently on an NTL PSTN line, using Asterisk 1.2 and a X101P
cheapo card.
The problem lies with detecting when the far end has hung up. It fails
to detect it, and will only cleardown when the silence timeout has
been reached. Now, I've seen the thread at
As per my (numerous) prior statements on this subject Asterisk WILL
NOT properly work with the spa-3000 DTMF in rfc2833. Use INBAND when
dealing with Asterisk on both the FXO/FXS ports of the spa3k if you are
dealing with Asterisk. This is a setting in BOTH sip.conf and spa3k pstn
and line 1
Hi,
someone has made me realize that a more detailed description is needed
for those who don't know about DeStar, so:
DeStar is a Web-based management and configuration tool for the Asterisk
PBX.
DeStar's main features include:
* Hosted PBX and virtual PBX features, which allow you to have
Heya everybody.
I have been peering into the code for zaptel for a while now, I am
keenly interested in the dacs support, being able to apparently
redirect certain spans to other spans. Not sure if this has to be on
the same T1 interface or can be used between T1 interfaces on the same
board or
I tested it over a SIP channel and an IAX channel and it did work, but
I have not used it in production that way. I only use Zap channels(T1
PRI) In prodution at the locations that I use AMD at.
MATT---
On 1/25/07, Peter Halliday [EMAIL PROTECTED] wrote:
That's the same code as I have. It's
I just saw the TC400B transcoder card at the IT Expo in Fort
Lauderdale. The Digium representative confirmed it was shipping. Does
anybody have one of this and can give us some feedback?
Thanks,
--
Andres
Technical Support
http://www.telesip.net
On Thursday 25 January 2007 6:30 pm, David Gomillion wrote:
I mean that I would like to have a system in place so that Asterisk, as a
privileged service, can gain access to Courier's IMAP storage. Having to
keep track of all of our users' passwords in the Asterisk configuration is
going to
I have same problem and no mailing list response. I suggest we go for
reporting bug.
- Original Message -
From: Jack Wei [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, January 26, 2007 1:16 AM
Subject: [asterisk-users] background() with m option
Hi...
In my
David,
According to the imap docs there should be away to set a single user and
password that Asterisk will use for IMAP connections, all that has to be
done then on the IMAP server is give that account full access to each
mailbox. That is according to the docs, I have not got my account to
I already put this in there, but this is the context for the call. I got it
right out of voip-info.org's article. This is correct right?
[outboundmsg1]
exten = s,1,NoCDR
exten = s,n,AMD
exten = s,n,GotoIf($[${AMDSTATUS}=AMD_PERSON]?humn:mach)
exten = s,n(mach),WaitForSilence(2500)
exten =
From: Brad Templeton [EMAIL PROTECTED]
I have a really dumb question. It appears that Yahoo, MSN, AIM, you
name
them, they don't have a NAT problem, and some use SIP. I don't think
they
all stay in voice path, either. What takes?
When you control both ends of the path, you can eliminate
Is it possible to have different families refer to different databases for
the same database driver? The examples I have seen specify the same host,
database etc. For example is this possible:
extconfig.conf
sipusers = mysql,asterisk,asterisk_sip
voicemail = mysql,mail,voicemail
If it is
I have a situation where the two Zap channels on a TDM400 are staying
offhook after a random period of time; it is not (I believe) related to the
FXO side not hanging up. Actually I suspect the server is overheating but I
need to do more analysis.
Anyway, my question is, how do I get the
From: C F [EMAIL PROTECTED]
Cory, it's called dialplan magic it realy depends what PBX it is, not
all of them allow dial plan magic. But it is possible on most pbxes.
CF: What exactly is diaplan magic? I googled but found little info.
The basic use case in Cory's posting does not seem to
Hi
is there a way to barge calls and record them at the same time ?
i use trixbox 2 with hudlite
adi___
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24 jan 2007 kl. 18.10 skrev Eric ManxPower Wieling:
James Fromm wrote:
The behavior we see is that the SIP interface in the queue will
sometimes not release from the in-use state. Connecting to the
interface from another SIP device and immediately hanging up will
clear the state.
The
Seems like a bug to me.
File a bug report in the bug tracker, bugs.digium.com. Upload
backtrace and all information you have.
Thank you!
/O
24 jan 2007 kl. 21.20 skrev Bruce Reeves:
I have one system that is crashing everytime a call is parked and I
have tried recompiling the asterisk,
25 jan 2007 kl. 08.26 skrev Darryl Dunkin:
There is an SVN branch with this feature:
http://svn.digium.com/view/asterisk/team/oej/multiparking/
I had hope this would be a feature added to Asterisk 1.4, but fail to
see it on the changelog.
It wasn't approved due to some architecture issues.
This gets fixed using the chan_capi.HEAD, not chan_capi.0.7.1; chan_capi
compiled right out of the tar.gz, no changes required (the defaults in
the Makefile are ok)
Cosmin Prund wrote:
I've got a brand new Eicon Diva Server BRI card and I want to
configure it with Asterisk. I managed to get
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