Den 19-01-2011 00:19, Nick Ustinov skrev:
Hello!
I have just upgraded to asterisk 1.8.2.1 and see some weird messages
in log when client tries to register:
[2011-01-19 00:52:47] WARNING[25624] chan_sip.c: Failed to parse contact info
[2011-01-19 00:52:50] NOTICE[25624] chan_sip.c: Peer
Den 02-03-2011 16:12, Jeremy Kister skrev:
On 3/2/2011 9:46 AM, Leif Neland wrote:
Some of the phones are being disconnected with Asterisk saying no reply
to critical packet
What kind of phones are they? I might have nothing to do with your
network configuration; try adding to sip.conf
I'm running asterisk on a Freebsd with 2 Nic's.
Inside NIC is 192.168.5.x where the phones are.
Outside NIC used to be a public IP with the ISP's device set to
bridging, but the new WiMAX router only offers me the public ip
94.18.x.x on the outside,
and forwarding everything to 192.168.1.50
I'm looking for a way for linux to query a pc if user X is on, and has
used the pc recently or the screensaver is not active.
If so, I'll route a call for user X to the phone near that PC.
Ideas, anyone?
Leif
--
_
--
Den 28-01-2010 20:15, Danny Nicholas skrev:
Here's one solution:
- exten = _911,1,Set(IMAT=EXTEN)
- exten = _911,2,Set(IMAT=CUT(IMAT|\/|2)
- exten = _911,3,Dial(DAHDI/1,w911)
- exten = _911,4,Background(emergencyin${IMAT})
Where you would record /var/lib/asterisk/sound/emergencyin100
Den 29-01-2010 19:38, Danny Nicholas skrev:
This might help
- exten = _911,1,Set(IMAT=EXTEN)
- exten = _911,2,Set(IMAT=CUT(IMAT|\/|2)
- exten = _911,3,Dial(DAHDI/1,w911)
- exten = _911,4(keepup),Background(emergencyin${IMAT})
- exten = _911,5,wait(10)
- exten =
Jeff LaCoursiere skrev:
On Fri, 15 Jan 2010, Hans Witvliet wrote:
If you connect your pc with GB-lan card to an dual-ported ip-phone, you
and up with an 100Mbps lan connection to your pc.
Only way to avoid that, is to insert a cheap second lan-card in your pc,
and connect your phone to
Hans Witvliet skrev:
During my last blackout i found out that all but my switches were on the
UPS... bummer!
Coincidentially, in danish, oops is spelled ups.
It also gives funny images when your packages are delivered by a company
called Oops...
Leif
--
hbk skrev:
Hi,
Is it possible to regret blind transfer while its ringing (not answered)?
Call pickup. If the phone is in your pickup-group.
Leif
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Tilghman Lesher skrev:
On Saturday 16 January 2010 11:02:52 Deep D wrote:
On Sat, Jan 16, 2010 at 9:21 PM, Tilghman Lesher tles...@digium.com wrote:
On Saturday 16 January 2010 06:04:01 Deep D wrote:
When I create a sip peer in users.conf then a hint is automatically
- Original Message -
From: randall
To: asterisk-users@lists.digium.com
Sent: Friday, January 15, 2010 7:54 AM
Subject: [asterisk-users] 10/100 voip phones and gigabit connection
hi all,
just subscribed to the list and first mail, nice to be here.
Hopefully i'm in
- Original Message -
From: Zhang Shukun
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, January 15, 2010 11:48 AM
Subject: [asterisk-users] Realtime queue not work
hi, all
i try to confiture realtime queue, but not work, details as below:
- Original Message -
From: randall
To: asterisk-users@lists.digium.com
Sent: Friday, January 15, 2010 2:11 PM
Subject: Re: [asterisk-users] 10/100 voip phones and gigabit connection
On 01/15/2010 02:00 PM, Leif Neland wrote:
- Original Message
Zhang Shukun wrote:
Thank you! it's very helpful.
now i have another question:
in asterisk, each agent should login first and then can response to
the caller. but i don't want to the login action.
i need agent shold response directly without login first. how should i do ?
can users in
- Original Message -
From: jonas kellens
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, January 11, 2010 2:30 PM
Subject: Re: [asterisk-users] Some minor configuration issues with queues
To answer my own question :
I had the following in my
If xlite subscribes on a hint, and the phone is offline, xlite says so
(not online)
If SPA942 does the same, the led is green for available. The other
hints work: blink red for ringing and red for busy.
I seem to remember the led once showed amber for subscribed phone offline.
The SPA extended
It seems dahdi is needed for meetme, but not available under FreeBSD.
So what do I do then?
Asterisk has only SIP-channels.
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Taylor, Jonn skrev:
Leif Neland wrote:
I can't believe anyone would use RJ-11 any more. You can multi-purpose
RJ-45 jacks to work with POTS lines. Run everything down to a central
panel and send pots over the jacks that you need to. That way if you
decide you need/want to go IP
Tim Nelson skrev:
- Leif Neland le...@neland.dk wrote:
I want some cheap ip-phones with auto-answer, to work as paging system
at dinnertime.
Options, please.
Leif
I've had great luck using the BT201 phones from Grandstream for this purpose.
In fact, this is the only
Gordon Henderson skrev:
On Wed, 30 Dec 2009, Leif Neland wrote:
Tim Nelson skrev:
- Leif Neland le...@neland.dk wrote:
I want some cheap ip-phones with auto-answer, to work as paging system
at dinnertime.
Options, please.
Leif
I've had great luck using
I want some cheap ip-phones with auto-answer, to work as paging system
at dinnertime.
Options, please.
Leif
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I can't believe anyone would use RJ-11 any more. You can multi-purpose
RJ-45 jacks to work with POTS lines. Run everything down to a central
panel and send pots over the jacks that you need to. That way if you
decide you need/want to go IP in the future, you're all set.
Darrick
I'd like to put a phone in a special context, where a test is made on its
business hours, then if so, proceed to the normal context to do whatever it
does with outgoing and local calls.
I've tried, just to go from one context to the next:
[specialoutgoing]
exten = _X.,1,noop(This is a special
- Original Message -
From: Tim Nelson
To: asterisk-users@lists.digium.com
Sent: Wednesday, December 02, 2009 12:06 AM
Subject: [asterisk-users] Slightly OT - Oreka Call Recording
Greetings all-
I'd like to install Oreka on a Centos 5.x server for monitoring my Asterisk
I made a patch to app_dial.c to make Dial(ext1ext2ext3,tumeout,B)
return busy when just one extension is busy.
http://www.neland.dk/app_dial.c.diff
It works, but...
I can't figure out setting/reading an option.
It looks fairly easy, but the flag is always set.
*** app_dial.c.org 2009-11-04
Norbert Zawodsky skrev:
What you're suggesting, though, violates the ENUM standard... and should
not be allowed.
N.
Sorry N. !
But - at least here in Austria - it is definitely *no* assumption that
my number with some extra digits can not be issued to someone else.
You
Rob Hillis wrote:
Leif Neland wrote:
I made a patch to app_dial.c to make Dial(ext1ext2ext3,tumeout,B)
return busy when just one extension is busy.
Forgive me for the question, but /why/ do you want this behaviour?
Isn't the whole point of dialling multiple extensions so that a call
it to ring
on my cell phone.
On Tue, 01 Dec 2009 21:35:11 +1100, Rob Hillis r...@hillis.dyndns.org
wrote:
Leif Neland wrote:
I made a patch to app_dial.c to make Dial(ext1ext2ext3,tumeout,B)
return busy when just one extension is busy.
Forgive me for the question
Philipp Kempgen wrote:
Leif Neland schrieb:
Norbert Zawodsky skrev:
The number +43-1-3207978 is my telephone number. I own it as long as I
pay for it. And with extra digits behind it I can do whatever I like. I
can create any extension - physical or virtual. I can attach a phone
mtha...@gmail.com wrote:
later i figured out the following.
my sip.conf was 2.2Mega Bytes size when populated with 50k users. That
means 2.2 x 1024 = 2.2 GB of memory. which is definitely not an option
with my small amazon system. I tried with 20k users, hola..
everything works fine.
Leif Madsen wrote:
Leif Neland wrote:
I made a patch to app_dial.c to make Dial(ext1ext2ext3,tumeout,B)
return busy when just one extension is busy.
In order to have your patch considered at all, you will need to file an issue
in
the issue tracker and attach your file
Leif Neland wrote:
But my problem comes when I speak on 0317998985 and someone calls on
985, the call
get to my celluar phone and ofc the other way around.
Is there a way to check if any extension is busy and in that case
jump to VoiceMail(0317998...@inputinterior.se,b)?
If both phones were
In a (futile?) attempt to get rid of warnings, I have this:
[Nov 30 10:39:49] NOTICE[68467]: loader.c:937 load_modules: 149 modules
will be loaded.
[Nov 30 10:39:49] WARNING[68467]: utils.c:1427 __ast_string_field_init:
trying to reset empty pool
(5 times more)
SIP channel loading...
(5 lines of
Tilghman Lesher wrote:
On Sunday 29 November 2009 17:03:04 Leif Neland wrote:
mtha...@gmail.com skrev:
Anyone know how many users i can record in sip.conf. (NO..NO i am not
discussing the simultaneous sip calls).
I tried with 50k users in sip.conf, but the sip module didn't reload
Leif Neland wrote:
I think a modification should be done around here to return busy if
just one channel was busy (only enabled if an option on dial is set)
in asterisk-1.6.0.15/apps/app_dial.c, line 610
Is somebody willing to try?
while (*to !peer) {
struct chanlist *o;
int
Leif Neland wrote:
#define OPT_PEER_H ((uint64_t)1 34)
#define OPT_SINGLE_BUSY ((uint64_t)1 35)
but all these constants have the value zero!
I'm compiling on FreeBSD, asterisk seems to work anyway...
Whats going on?
doh... 64 bits doesn't fit in %d
%llu works better.
Leif
mtha...@gmail.com skrev:
Anyone know how many users i can record in sip.conf. (NO..NO i am not
discussing the simultaneous sip calls).
I tried with 50k users in sip.conf, but the sip module didn't reload.
tried with few hundred of users and it works. any idea what is the
limit in
But then you create phonenumbers in enum, which doesn't exist as
pstn-numbers.
Not the idea behind enum.
On the other hand, if you owned 10 or 100 pstn-numbers in series, you
could get the last one or two digits delegated to your dns-server.
Why do I create numbers in enum which doesn't
- Original Message -
From: Norbert Zawodsky
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, November 26, 2009 10:46 AM
Subject: Re: [asterisk-users] Please some enlightment on ENUM !!
Leif Neland schrieb:
But if a pstn or cell call
- Original Message -
From: Norbert Zawodsky
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, November 26, 2009 10:46 AM
Subject: Re: [asterisk-users] Please some enlightment on ENUM !!
Leif Neland schrieb:
But if a pstn or cell call
Norbert Zawodsky skrev:
SIP schrieb:
Yes... you would have to register (and possibly pay for, dependent on
the ENUM registrar) each individual number. The idea behind ENUM is that
it's an E164 number that is already yours that maps to whatever you want
it to map to (email, SIP, etc).
- Original Message -
From: Norbert Zawodsky
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, November 23, 2009 3:15 PM
Subject: [asterisk-users] Please some enlightment on ENUM !!
Hello all you Gurus out there!
Please could you explain
Norbert Zawodsky wrote:
Leif Neland schrieb:
- Original Message -
*From:* Norbert Zawodsky mailto:norb...@zawodsky.at
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com
*Sent:* Monday, November 23, 2009 3
Magnus Benngård skrev:
Hi!
Part of extensions.conf:
exten = 985,1,Dial(SIP/0317998985H323/00702221...@avaya,20)
exten = 985,2,Goto(985-${DIALSTATUS},1)
exten = 985-BUSY,1,VoiceMail(0317998...@inputinterior.se,b)
exten = 985-BUSY,2,PlayBack(vm-goodbye)
exten = 985-BUSY,3,HangUp()
exten =
Philipp Kempgen skrev:
Leif Neland schrieb:
Philipp Kempgen skrev:
Leif Neland schrieb:
Mostly to debug/test BLF, is there a softphone or another app. which can
subscribe to hints on Asterisk?
X-Lite
It does not
subscribe to hints on Asterisk
Philipp Kempgen skrev:
Leif Neland schrieb:
Mostly to debug/test BLF, is there a softphone or another app. which can
subscribe to hints on Asterisk?
X-Lite?
http://www.counterpath.com/x-lite.html
Philipp Kempgen
It does not
subscribe to hints on Asterisk.
Leif
Ira skrev:
At 07:06 AM 11/18/2009, you wrote:
I know that I'm not looking for Dial(SIP/xSIP/y) - as documented, this
handles nothing like what I'm looking for.
It's not the answer you're looking for, but that feature is built
into a Aastra 480i-CT and I think a 57i-CT.
Do you
Leif Neland wrote:
- Original Message -
*From:* Ex Vito mailto:ex.vitor...@gmail.com
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com
*Sent:* Thursday, November 12, 2009 3:59 PM
*Subject:* Re: [asterisk
I've got a SPA942 subscribing to hints to a local asterisk 1.6; this works.
But when I try to subscribe to a remote asterisk 1.4, it doesn't work;
the BLF is flashing yellow.
I see this in the log: Received SIP subscribe for unknown event
package: call-info
The SPA942 extended function for the
Mostly to debug/test BLF, is there a softphone or another app. which can
subscribe to hints on Asterisk?
Heck, it doesn't even need to be able to do calls :-)
Leif
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- Original Message -
From: Ex Vito
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, November 12, 2009 3:59 PM
Subject: Re: [asterisk-users] BLF with SPA941?
Although I've never tested such feature on those devices, I know
that it was only
I think just renaming the [default] to [public] or [unautorized], and a comment
saying
Don't put outgoing calls in this context, as unauthorized users, even from
outside, are routed here by default.
would be enough.
I'm not sure if local phones should automatically be routed to a [local]
- Original Message -
From: aster...@opensourcesolution.in
To: asterisk-users@lists.digium.com
Sent: Friday, November 13, 2009 9:47 AM
Subject: [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE
hi all,
i had installed and configured asterisk on centos 5.3, i had
Appearently SPA941 is less than a SPA942 without two ports, poe and backlight.
There is less features too, it doesn't support BLF.
Is it possible to hack 942-software into 941, or is there another workaround?
Leif
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- Original Message -
From: Dan Journo
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, November 12, 2009 1:24 PM
Subject: [asterisk-users] Incoming Call Ring
Hello,
I have Asterisk set up with 6 extensions. When a call comes in, I use a
- Original Message -
From: B.Masoud @ SH
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Sent: Tuesday, October 06, 2009 1:14 AM
Subject: [asterisk-users] Networking Concept
Hello,
I would like to know how Asterisk deal in this case:
Assume I
In relation to our CRM-system I'd like to send a query to asterisk who
is extension xxx talking to.
When the operator enters the page with customer data, the crm should
send a query to asterisk, to get the cli of the call the operator is having.
If the number is matching the customers number in
I have 3 phones, SIP/3, SIP/6 and SIP/9
SIP/3 subscribes on hint on SIP/9
Phone 6 calls phone 9, blf on phone 3 flashes until 9 picks up, then it
is steady red. That's correct.
But when 9 hangs up the hint goes to InUseRinging, the light on 3 is
still flashing.
It keeps flashing until somebody
I have a SPA742, which can autoanswer a call
In the dialplan, I have this:
exten = 28,1,SIPAddHeader(Call-Info:\;answer-after=0)
exten = 28,2,dial(SIP/36)
Now I want some external event initiate a call to that phone and play a
message.
I have been thinking of dialfiles, but I believe there is
Ex Vito skrev:
2009/9/27 Leif Neland le...@neland.dk:
Can I, via a callfile, or command-line parameters to Asterisk start a
dialplan-script?
eg asterisk -someflag execute callalert
then in dialplan
[callalert]
exten = s,1,SIPAddHeader(Call-Info:\;answer-after=0)
exten = s,2,dial(SIP
(catching up while my adsl is offline)
David L. West wrote:
I want callers to go into the queue(s) and just hear ringing instead
of MOH. Is this possible?
If everything else fails, you can generate a file with ringing tones, and
use that for moh.
Leif
(While my adsl is down, I'm reading old posts.)
Tom Lanyon wrote:
Hi list,
Does anyone have any advice on the following:
Incoming calls to our office come in on a SIP trunk. Since all our
offices/desks are in close proximity, we would like just a single
phone to ring when a call comes in
Steve Totaro wrote:
Stephen Bosch wrote:
Olivier wrote:
I'm really after 1U-2U silent servers as I've got the feeling most
of them are too noisy for offices and most of our clients don't
have server rooms.
Try this:
http://www.tomshardware.com/2006/01/09/strip_out_the_fans/
-s
The
When phone A registers, I want phone B to ring, when picked up, it should
call phone A and connect the phones.
Translated: When GF in Mexico powers up laptop where soft iax-phone
registers automatically, I want to talk to her asap :-)
How to?
Leif
J. Oquendo wrote:
Anyone experience ring oddities with extensions.conf rollovers? Let me
summarize...
One of my extensions.conf file is built to ring during the day, ring/go
to voicemail after a certain time:
[main-aa]
exten = s,1,GotoIfTime(17:00-8:30|mon-fri|*|*|*?main-night-aa,s,1)
exten =
Erick Perez wrote:
Hi there, im looking for another place that provides manuals and
firmware updates for the ATCOM AT 468 and their configuration with
asterisk.
the site www.atcom.com.cn has non functional download links.
I suppose you mean the AG 468
If you can find somebody who still uses
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
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
Is it possible to pickup a caller, who is in the menus somewhere, for
instance he may be lost in the telemarketer torture script?
Just like it is possible to pick up a call on a ringing phone.
Leif
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I have some siemens wireless ip-phones.
There is no problem entering ** which I have configured in features.conf
to be transfer. But then it is difficult to enter the extension, because
one have to wait the right amount of time before entering the extension.
Because we only have few
Nick Ellson wrote:
How might you identify a mobile #? (assuming you refer to cellular
phones) Now that phone companies are allowing you to transfer your
land line to a mobile, it's no longer practical to use prefix
blocking.
If a land line is transfered to mobile, does it cost more to call
Rich Adamson wrote:
Dovid Bender wrote:
Good Morning List,
When setting up a pbx and you want to test your 911 settings do you
call 911 and tell them its a test call or do you relly that you set
it up properly and hope for the best when some one call's 911 ?
I believe most 911 centers would
Original Message
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, January 29, 2006 1:29 AM
Subject: [Asterisk-Users] Multiple Subscriptions to SIP accounts at
SameDomain
Sorry not to have observed etiquet and lurked here for a bit before
wading in with a
Original Message
From: Jean-Michel Hiver [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com Sent: Sunday, January 15, 2006 1:16 PM
Subject: [Asterisk-Users] Detecting Long PDD
Hi List,
I've had some issues with some VoIP
Original Message
From: Jean-Michel Hiver [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com Sent: Tuesday, January 10, 2006 2:17
PM Subject: Re: [Asterisk-Users] Recommendations on a WiFi phone for *?
Rupert Gregory a écrit :
Original Message
From: Andrew Nowrot [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com Sent: Wednesday, January 04, 2006
8:53 PM Subject: [Asterisk-Users] Email2fax big problemo
Hi,
Few days ago I installed Email2fax
Original Message
From: Andreas Koch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com Sent: Monday, January 02, 2006 1:03 PM
Subject: [Asterisk-Users] connect more the one phone to ONE sip Acoount
Hello,
how is it possible to
Original Message
From: Ross C [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com Sent: Friday, December 30, 2005 7:18
AM Subject: RE: [Asterisk-Users] Semi-OT: porting numbers away
Thanks, but I'm looking for information on
An interesting wrinkle I'm running against is that you cannot port
numbers from a cellular carrier to a landline. i.e. I can't port my
cell # to a DID on my PRI. I am not sure if this is just a line of
bullshit fed to me from Bell Mobility (Canadian CDMA carrier) but
I've not had the time to
Original Message
From: Eck [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com
Sent: Saturday, December 31, 2005 8:26 PM
Subject: RE: [Asterisk-Users] voicemail/privacy system
If you dont want to get too stuck into the guts of Asterisk yet, the
[EMAIL PROTECTED]
Original Message
From: Peter Bowyer [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, December 31, 2005 11:34 AM
Subject: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP
Hi all
Slightly OT but I know a lot of GS experts hang out here - I just
upgraded a GXP-2000 to
Original Message
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com Sent: Wednesday, December 21, 2005
9:14 AM Subject: Re: [Asterisk-Users] Asterisk - Skype
anywhere/anyhow?
On Tue, 20 Dec 2005, AR Tarzi wrote:
could
From: Jan Saell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 30, 2005 9:32 AM
Subject: Re: [Asterisk-Users] Blind transfer question
I did a quick check on the blindxfer config parameter and i cant find
I find the transfer functions a little lacking.
Examples:
I get a call
I do an attended transfer, but the called extension never answers/I get
impatient/I discover I have dialed the wrong extension.
I can not get the call back.
If I hangup, the caller is also hung up. I'd prefer the caller to
- Original Message -
From: Patrick [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, December 01, 2005 4:00 PM
Subject: Re: [Asterisk-Users] Better transfer
On Thu, 2005-12-01 at 15:50 +0100, Leif Neland
Original Message
From: Innocent Evil [EMAIL PROTECTED]
you can use 'w' option with 'Dial' on 1.2.x
I don't think w do anything like 'wait', If I am wrong, correct me
someone please According to app_dial.c
w- Allow the called party to enable recording of the call by
On 08:48, Tue 29 Nov 05, [EMAIL PROTECTED] wrote:
From memory (at a previous installation) you will need a newer version
of
Asterisk than 1.09 for the lights to work.
on 1.0.9 the lights work.
In this way:
person is on the phone: light is on
Person is not on the phone: light is off
I still continue to reboot my asterisk box everyday.
I posted a message on November 22, but it was on another thread and
no one answered me, so I try again here,
where a lot of people told be I was a bad administrator (Like a
Windows administrator and I don'0t want to resolve my problem)
Original Message
From: Esteban Maestre [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, November 25, 2005 11:22 AM
Subject: [Asterisk-Users] sound problem, please help!
Hi all!
I have a strange problem when using asterisk. I have configured
asterisk to receive
From: Bartosz Wegrzyn - asterisk [EMAIL PROTECTED]
Hi,
After I compile asterisk v.1.2 is tells me that last thing to do is to
make install. Unfortunately it goes it to loop after I type make
install
this is the loop:
else \
mv include/asterisk/version.h.tmp
Adrian A wrote:
Does anyone know what exactly the option
transmit_silence_during_record in asterisk.conf does? Is this useful
for voicemail recording?
Could the option be named any more explicitly? It does _exactly_ what
it says it does.
Some providers terminate the connection if nothing is
I need a hint:
From pbxmanager/doc/INSTALL
2. Install a database adaptor via rubygems. Postgresql, Mysql, and Sqlite3
are all supported and tested to work.
Eh... How to install?
Leif
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--Bandwidth and Colocation sponsored by Easynews.com
Original Message
From: Chris Cahill [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, November 17, 2005 1:15 PM
Subject: [Asterisk-Users] /spool/outgoing delays
Hi,
I have a rather interesting problem with my Asterisk setup at the
moment, and was wondering if
Original Message
From: Louis-David Mitterrand [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, November 18, 2005 10:10 AM
Subject: [Asterisk-Users] gpx-2000 early dial support
The gxp-2000's lack of a dialplan (or did I miss it?) led me to
activate its early dial
Original Message
From: John Heng [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, November 18, 2005 1:56 AM
Subject: [Asterisk-Users] Hung Zap channels
Hi all,
Once in a while, I've found
that the zap channel will get stuck (or blocked) even after the call
has
Original Message
From: Joseph Rothstein [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, November 15, 2005 10:42 AM
Subject: [Asterisk-Users] voicemial maxmsg
Has anyone tested the maxmsg parameter in the voicemail.conf file? I
am trying to restrict the number of
Original Message
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com Sent: Tuesday, November 15, 2005 2:27
PM Subject: Re: [Asterisk-Users] Multiple Outbound SIP Trunks
On Tue, 15 Nov 2005, Pikoro wrote:
There will be
Is there some way my uplink can tell my * the price of a call, either per
timeunit in the conversation at start of the call, or the total cost at the
end of the call?
I'd like to pass the bill on to the extensions.
Leif
___
--Bandwidth and
Original Message
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com Sent: Monday, November 14, 2005 4:50
PM Subject: [Asterisk-Users] Problem with Cisco local conference and
hangup
Cisco 7960 gets a call from zap/1,
Cisco 7960 gets a call from zap/1, hits conf to call out on zap/2,
then hits join, after a while cisco hangsup, at which point zap/1
and zap/2 can still talk, shouldn't asterisk hangup on all three?
That is the way I would prefer it to work.
Like an attended transfer.
I cannot understand
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