Just a shot in the dark... but have you tried Answer() before
Playback()?
Josh McAllister
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom
Engleward
Sent: Monday, May 01, 2006 11:43 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
well, if you dont use/need a module, in modules.conf put noload =
app_intercom.so (for example).
i think you can choose whether to automatically load all then specifically
noload whichever you dont
want with a noload =, or with
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
that is a nice function
I use a cronjob to logout everyone each evening if anyone wants that script
I would love to provide it.
Please send the script to the list.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.:
Hi.
How does this work?
What if this SIP/account was a member (agent) of
a queue?
Ex: dial(SIP/account,20,tT). Would the dialstatus
be set as busy when one of the phones is actively talking, or will the other
phones continue to ring?
You may have seen my other submissions to this
Hi,
Joshua wrote:
Why don't you use something like the chan_local channel driver to send
the call into the dialplan where it will then execute the extension.
Joshua,sorry i don't understand chan_local channel driver to send
the call . Could you explain or link to appropriate documentation.Can i
Hi there, I am new to asterisk, I am trying to write a dialplan with 2 process running concurrently. Current dialplan only able to execute process with priority example exten =100,1,Answer()exten =100,2,Musiconhold()
exten =100,3,Hangupis it possible to have process musiconhold/background and dial
Hi.
The UPGRADE.txt of asterisk distribution contains the following snippet
under the MeetMe heading:
"MeetMe:
* The conference application now allows users to increase/decrease their
speaking volume and listening volume (independently of each other and
other users); the 'admin' and
On Sunday 30 April 2006 10:27, Boris Bakchiev wrote:
Opened pseudo zap interface, measuring accuracy...
This may be a stupid question but how did you do this?
--
Cheers
Wayne
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I downloaded the source and built it from that. SuSE10 comes with a
version of asterisk 1.0.X on the DVD.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yu Safin
Sent: 01 May 2006 16:31
To: Asterisk Users Mailing List - Non-Commercial
have to use 's' option in MeetMe app, to enable users to go to menu pressing * during conference. When in the menu... well, Allison explains what to do...
;-)
2006/5/2, Jan du Toit [EMAIL PROTECTED]:
Hi.The UPGRADE.txt of asterisk distribution contains the following snippet under the MeetMe
Hi
I want to submit a bug about auto-dial , but I
am not sure on which project the auto-dial comes, how
to know about which project , auto-dial comes
Thanks
Joseph
___
To
On 29 Apr 2006, at 03:10, Aryn Nakaoka wrote:
I have 2 Asterisk servers connected via IAX2 connections.
PBX1 is on the internet with a public IP Address
- with PRI
PBX 2 is behind a NAT router with IAX2 Ports forwarded
1-way audio is an issue with incoming and outgoing
Hi all,
i have an Asterisk box with an Eicon 4BRI with
chan_capi-cm and every thing works fine. We now plan
to install a new Asterisk using a Dialogic BRI/2VFD.
Is the Dialogic card supported and can i use
chan_capi-cm? Has anyone managed to install this card?
Unfortunately i was unable to find
I set up the Asterisk for my company which is a business center, I will assign a specific telephone number to my client that uses my serivces. All of their incoming calls will be first picked up by the receiptionist, can I disply the company name instead of the called number on my receptionist's
Armin Schindler wrote:
I use two possibilities.
a) when I want to connect the ISDN card directly with the device using a
short cable, I just cut the cable in the middle an reconnect them crossed
and add the resistors here. (Maybe this is not the correct way for
termination, but it
If the incoming call is from a PRI with DIDs, you could just simply map the caller ID name to the company name (company name being the name of the company being called, this is opposite of normal caller id name) based on the DID. Or, you could also play a sound file containing the company name.
On 2 May 2006, at 01:25, Tom Engleward wrote:
I have an asterisk system behind NAT, and need to
connect to public PSTN originators via SIP or IAX2,
but don't have the option of forwarding any ports
(4569, 5060, etc) to the asterisk system. However, the
NAT system does properly establish
Just a quick reminder - now is the last chance to register for
MeetAsterisk in
Brussels on Thursday and London on Friday.
We have updated the web site with location information and will
keep registration open until tomorrow lunch.
http://www.meetasterisk.com
See you!
/Olle
On a full cone NAT, I have never been able to get the ATA to register
without a stun.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Tuesday, May 02, 2006 1:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
Looking at my log fileI found the following
error:
May 2 12:00:45debian kernel: Zapata
Telephony Interface Registered on major 196May 2 12:00:45 debian
kernel: No ISA tormenta card found at dMay 2 12:00:45 debian
kernel: Zapata Telephony Interface UnloadedMay 2 12:00:45 debian
Hi,
just answering myself:
I am not allowed to send the leading 0 for my prefix with the callid, then it
works well.
Sebastian
Sebastian Reitenbach [EMAIL PROTECTED],Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com wrote:
Hi *,
now for a long time
On Tue, 2 May 2006, Klaus Darilion wrote:
Armin Schindler wrote:
I use two possibilities.
a) when I want to connect the ISDN card directly with the device using a
short cable, I just cut the cable in the middle an reconnect them
crossed
and add the resistors here. (Maybe this is not the
HiDinesh, thanks for help me.
To activate the transferences of calls in asterisk, I effected:
SIP.CONF in sip of the agent I qualified canreinvite=no, so that asterisk monitors this transference.
EXTENSIONS.CONF I qualified the parameters tT in the command Dial
FEATURES.CONF I qualified [
Found it!
It seems that Asterisk is looking at the date / time stamp of the call
file to process the call?? I was simply moving the call files hoping
it would just work (tm)
I guess that the call files created on the samba share I created carried
the time/date stamp of the local machine
Hi friends,Thank you for your response. I am using SuSe Linux 9.3 with kernel 2.6 version. I have installed Asterisk in my PC and "X-Lite" as softphone in my PC and client PC. Here my user name is "chandra" and client user name is "aarti". I have added these lines to configuration files at the
On Tue, 2006-05-02 at 14:52 +0200, Remco Barende wrote:
Found it!
It seems that Asterisk is looking at the date / time stamp of the call
file to process the call?? I was simply moving the call files hoping
it would just work (tm)
I guess that the call files created on the samba share
can u check what this command gives
iptables -L
or do iptables -F [ Not advisable , but for testing
OK ]
then try again
--- Crazy Boy [EMAIL PROTECTED] wrote:
Hi friends,
Thank you for your response. I am using SuSe Linux
9.3 with kernel 2.6 version. I have installed
Asterisk in
You are missing the dtmf mode, and most importantly
the codec to be used.
I would also add the nat=yes, that is probably why your
phone isnt registering.
See below for example config:
[chandra]
type=friend
username=chandra
secret=chandra
nat=yes
host=dynamic
dtmfmode=rfc2833
On 5/2/06, Wayne Gemmell [EMAIL PROTECTED] wrote:
Opened pseudo zap interface, measuring accuracy...
This may be a stupid question but how did you do this?
in your zaptel source dir (after making..): ./zttest -v
or search for zttest on voip-info.
cheers
Hi,
I'm wondering what I need to change to get the swedish type ring on a
SIP-trunk. When I make an inbound call i still have the US-type of
ring on my SIP trunks. I need help on changing this.
However I've successfully changed this on the Zap interface for all
inbound calls.
Thanks in advance!
--- Damon Estep [EMAIL PROTECTED] wrote:
Qualify=yes will send a SIP OPTIONS periodically and
keep the NAT open,
if you use 1 to 1 NAT (versus PAT where it is many
to one NAT) it will
work because port 5060 on the private address will
still be port 5060 on
the public address.
Tried that,
Armin Schindler wrote:
Thanks for verifying the crossover layout.
FYI: http://www.ksi.at/online-kataloge/kat8A/8A-109/8A-109.htm
I've ordered some of them - I do not like soldering :-)
Thanks for the link. But these devices do not seem to have a crossed
connection.
No, just termination.
Hello,
I have some problems with a new configuration:
I always have on my asterisk console the message:
chan_iax2.c:5886 update registry: restricting registration for peer '19'
to 60 secondes
I connect only two ip phone with iax protocol.
And when i want to call 19 phone, it's hangup. No
my guess is that you are trying to dial a sip channel to reach an iax peer.
Dial(SIP/19)
should be
Dial(IAX2/19)
Olivier Saulnier wrote:
Hello,
I have some problems with a new configuration:
I always have on my asterisk console the message:
chan_iax2.c:5886 update registry: restricting
--- Tim Panton [EMAIL PROTECTED] wrote:
Yes. That is the way that IAX2 likes to work.
Ok.
However, not all providers will allow it, some
require a fixed IPaddress
and port for them to send calls to.
Is this the reason for the recommendation I've seen in
various forums to have port 4569
Yes Sean,
I've just see that :-) I modify, and my communication is now OK.
But i always have the message on the console...
Bets regards,
OLS
Sean Cook a écrit :
my guess is that you are trying to dial a sip channel to reach an iax
peer.
Dial(SIP/19)
should be
Dial(IAX2/19)
Olivier
--- Josh McAllister [EMAIL PROTECTED] wrote:
Just a shot in the dark... but have you tried
Answer() before
Playback()?
http://www.voip-info.org/wiki/index.php?page=Asterisk+tips+answer-before-playback
says New versions of Asterisk have added Answer
capabilities to several functions like
I can either get the TE100P working or the 3 X100P clones working, but
never both. I have the TE100P connected to a channel bank, and X100P
clones to lines from the phone company.
This is my zaptel.conf
span=1,1,0,d4,ami
fxsks=1-24
loadzone=us
fxols=25-27
loadzone=us
I then do
[EMAIL
Giuseppe Parlato wrote:
Looking at my log file I found the following error:
May 2 12:00:45 debian kernel: Zapata Telephony Interface Registered on
major 196
May 2 12:00:45 debian kernel: No ISA tormenta card found at d
May 2 12:00:45 debian kernel: Zapata Telephony Interface Unloaded
Is this a problem? What is dnd anyway?Thanks,Jim.May 2 10:44:08 DEBUG[6277] db.c: Unable to find key 'SIP/201' in family 'dnd'
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On 2 May 2006, at 15:32, Tom Engleward wrote:
--- Tim Panton [EMAIL PROTECTED] wrote:
If you are using IAX2, you don't need to port
forward the ports.
Just have PBX2 register _often_ and that will keep a
mapping in your
router.
Where is this set? Is this the minregexpire and
maxregexpire
Are you seriously trying to run 4 cards in one system? The odds of getting
that working are about the odds of Angelina Jolie showing up on my doorstep
ready to whisk me off tobut I digress...you will have serious interrupt
issues trying to get 4 cardss working in one system. I am surprised
The last sip device to register gets the call. The way around this is to have your sip devices register under different accounts and create a ring group (dial(SIP/dev1SIP/dev2SIP/devN))AFAIK, there isn't a reliable method of determining if a sip device is busy other than calling it.
On 5/1/06,
On Tue, 2 May 2006, Wai Wu wrote:
[EMAIL PROTECTED] root]# modprobe zaptel
[EMAIL PROTECTED] root]# modprobe wcte11xp
ZT_CHANCONFIG failed on channel 25: No such device or address (6)
/lib/modules/2.4.20-8/misc/wcte11xp.o: post-install wcte11xp failed
Is this a problem? What is dnd anyway?
Not a problem, probably dialparties.agi checking if this extension as
DND enabled.
DND stand for Do Not Disturb
hth
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Hi Wai Wu, is seeming that you have two problems. In the case of wcte11xp, I find that its board TE110P is unplugged of slot PCI, removes the board and restarts the server. Later it again board the TE110P in slot correctly and restarts the server, must resolv this problem. In the case of wcfxo, I
You probably just don't have the original Tormenta ISA card. You can
ignore this message. Better yet, configure your system to not load the
torisa.o module.
first thanks.. I think so, I guess I don't have it. However I even don't
know what it is and why it was needed to be configured.
I'm
DND = do not disturb.
Sounds like you are running [EMAIL PROTECTED]
The error below just means that on an incoming call, it
looked to see if you set a flag for DND in the database. You didnt, so
it just went to the next number in the dial plan. That is what is supposed to
happen.
Hello,
When I compile zaptel application, i have this error log file:
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -lm gendigits.c -o gendigits
./gendigits
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
You probably just don't have the original Tormenta ISA card. You can
ignore this message. Better yet, configure your system to not load the
torisa.o module.
first thanks.. I think so, I guess I don't have it. However I even don't
know what it is and why it was needed to be configured.
I'm
I have recently upgraded to 1.2.7.1 from 1.2.4.
I can no longer use sip show inuse.
Below is the output... I know there are current calls:
redhat*CLI sip show inuse
* User name In use Limit
* Peer name In use Limit
Does anyone have an idea why this
Hi,
I've been using Asterisk for several months now with great success.
I'm working on a system that tries to deliver a recorded message to a user, as
follows:
1. a call file is placed in /var/lib/asterisk/outgoingcalls
2. This triggers a call to be placed
3. When answered, the caller hears
Hi,
I am running Asterisk v1.2.7.1 with a Digium TE110P. My dialplan is very
simple, when a call comes in on my analogue BT PSTN line, it rings the other
ZAP interface (my house phone). Slightly pointless (having a 1x1 switch) I
know, but I am planning on doing more with internal SIP extensions,
This is a system for our lab. I have no problem getting rid of X100P
clones. But I am just curious why can they work. Even the drivers are
not loading correctly.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kerry
Garrison
Sent: Tuesday, May 02, 2006
I have dial through application, that uses the wW options on the dial
command. However it's seems to be really hit or miss if asterisk picks up
the *1 and starts the recording. It can take 3 or 4 attempts before I can
see from the console that's it's started recording. A user just on the call
not
May I know if you could send me some coding on the *.conf, so that I
can follow the idea that you suggest??
ML
2006/5/2, Lacy Moore - Aspendora [EMAIL PROTECTED]:
If the incoming call is from a PRI with DIDs, you could just simply map the
caller ID name to the company name (company name being
Hmm, I don't have /etc/modprobe.conf, and wctdm is giving me problems.
Which device is it talking about?
[EMAIL PROTECTED] /]# modprobe wctdm
/lib/modules/2.4.20-8/misc/wctdm.o: init_module: No such device
Hint: insmod errors can be caused by incorrect module parameters,
including invalid IO or
I was about to post a bug, It hasn't worked for me since CVS 11/01/05!!!
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of William Piper
Sent: Tuesday, May 02, 2006 11:17 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Grab your fav. bottle of
${Insert_Your_Fav_Booz_bottle_brand_here} and get
working on it.
--- TV JOE [EMAIL PROTECTED] wrote:
I write perl applications for a living and have
developed code to
talk to all kinds of hardware. What I'd like to do
is pull a list of
phone numbers from sql
Are you seriously trying to run 4 cards in one system? The odds of
getting
that working are about the odds of Angelina Jolie showing up on my
doorstep
ready to whisk me off tobut I digress...you will have serious
interrupt
issues trying to get 4 cardss working in one system. I am
Not sure if this helps any, but I had no clue what sip show inuse would
show until I found out you had to put incominglimit (or call-limit) in
sip.conf/realtime for it to know that the phone was in use... Not sure if
this'll help.
On Tue, 2 May 2006, Alexander Lopez wrote:
I was about to
I seem to be getting a connection from telasip but instead of dialing my default extension, nothing happens. I listen to dead air.I have a fxo card configured and working on both inbound and outbound calls. Telasip is working outbound. I put in the recommended (by telasip) changes to the trunk for
richard Coco wrote:
Hi all,
i have an Asterisk box with an Eicon 4BRI with
chan_capi-cm and every thing works fine. We now plan
to install a new Asterisk using a Dialogic BRI/2VFD.
Is the Dialogic card supported and can i use
chan_capi-cm? Has anyone managed to install this card?
Unfortunately
But code quickly, as the quality produced is inversly related to the
amount of ${Insert_Your_Fav_Booz_bottle_brand_here} in your system.
Grab your fav. bottle of
${Insert_Your_Fav_Booz_bottle_brand_here} and get
working on it.
--- TV JOE [EMAIL PROTECTED] wrote:
Isn't a TE110P a PRI card? Are you sure that's the right model number for an
analogue interface card?
For our sites with BT lines, we have them configured as follows (I've
extracted the settings I think might be relevant):
usecallerid=no
hidecallerid=no
callwaiting=no
callwaitingcallerid=no
It does work now... thanks.
annoyed
I guess I need to add that line to all sip users if I want to monitor who is
on the phone and who isn't.
/annoyed
Thanks again,
bp
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel
Sent: Tuesday, May 02,
Actually, I take that back... It still isn't working. It does show the users
peers now, but they stay at 0.
I set this on our SIP carrier made both an incoming and outbound call...
it still showed 0 during the call.
Any other ideas?
Thanks,
bp
-Original Message-
From: William Piper
Not to mention the obvious, and this may not help your situation, but if
you were (or are) using templates it would be a one-line change.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William
Piper
Sent: Tuesday, May 02, 2006 12:49 PM
On Tue, May 02, 2006 at 05:39:53PM +0100, Chris Bagnall wrote:
[snip]
I think disabling asterisk from getting caller ID off an analogue line
improves its answering speed considerably. Of course, if you want CLID info
off your analogue line (and are paying BT for the privilege), you may not
Andrew Kohlsmith wrote:
On Thursday 13 April 2006 09:02, David Cook wrote:
My cell vm goes to asterisk, not the carrier. Apparently MWI is turned
on/off with specially formatted SMS messages. Anyone know how to do this
on a Treo 600? Having the phone light from Asterisk would be HUGE ...
not
How about AstRTB ? Asterisk Real Time Billing
--- Thameem Ansari [EMAIL PROTECTED] wrote:
Hello All,
I had the same question when I was writing my own
billing software in java.
Here is what I am doing to track multiple calls at a
time from the prepaid
account.
1. Keep on db table for
How about the Call Progress Analysis ?
Mike
- Original Message -
From: Dovid Bender [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, May 02, 2006 11:14 AM
Subject: RE: [Asterisk-Users] Asterisk as a phone survey
On 4/13/06, David Cook [EMAIL PROTECTED] wrote:
My cell vm goes to asterisk, not the carrier. Apparently MWI is turned
on/off with specially formatted SMS messages. Anyone know how to do this
on a Treo 600? Having the phone light from Asterisk would be HUGE ...
not to mention extremely cool.
So now I have a new question (besides my original,
about how to ensure that asterisk _always_ answers the
phone): why would enabling qualify cause an
immediate and consistent failure to ever answer
incoming external phone calls?
Because the firmware on your NAT router has an unconditional
On Tuesday 02 May 2006 13:10, Mark Johnson wrote:
I know this thread is probably a little aged, but I'm intrigued... How
are you forwarding cell vm to asterisk? When busy or unavailable, do
you forward to a DID set up to go directly to your asterisk voicemail?
heh... I cheat. I don't give
Mark Johnson wrote:
Andrew Kohlsmith wrote:
On Thursday 13 April 2006 09:02, David Cook wrote:
My cell vm goes to asterisk, not the carrier. Apparently MWI is turned
on/off with specially formatted SMS messages. Anyone know how to do
this
on a Treo 600? Having the phone light from Asterisk
On-line signup form are available on our website at www.cavp.ca in the
Membership section or please call 1-866-940-CAVP (2287) and select
option 3 (CAVP treasurer).
--
The CAVP is now accepting memberships applications.
This is a pivotal moment for the CAVP and we need your support.
In article [EMAIL PROTECTED],
Benoit Panizzon [EMAIL PROTECTED] wrote:
On Friday 28 April 2006 15:32, Eric ManxPower Wieling wrote:
What does the R option do?
Indicate 'Ringing' as soon as the called party indicates 'Ringing'.
The 'r' option indicates 'Ringing' as soon as the connection
On Apr 28, 2006, at 8:45 AM, Don Pobanz wrote:
Wai Wu wrote:
One question thought, does the hardware echo cancellation work much
better than software?
I bought a Digium TE411P hoping the hardware echo canceler would
improve my echo problems over the software echo canceler, but had no
Hi
all,
I've got an Asterisk
at home system running the new Free PBX front. It's solved all our small office
VOIP phone system which we are using as our only source of telephone
communications.
Anyway,I have
set up a few ring groups.
The first rings the
internal office extensions.
Dear guys:We are expanding our voip unit and currently looking for an Asterisk technician that can be part of our company here in Buenos Aires. If you know anyone who lives here and knows Linux and Asterisk, please contact me asap.
Best regards,Sergio Veltriwww.pointhorizon.comSuipacha 119 Primer
Hello
Everyone,
Here is the
scenario... I have a client who has two different companies in the same
officebut everyone works for both companies. Each person has a DID
for both companies. They only want to have one phone at their desk.
They have purchased the GXP-2000 ip phones for the
Hi,
I have a Sangoma 200A (I think that's the model #) analog 4 port card.
It works great... however almost everytime after someone hangs up a
call they were on.. the system rings the call back in, as though it
were a new call coming in. When they pickup no one is there.
Can anyone suggest why
Hey
everyone,
I have a TE110P card
hooked up to a PRI and about 40 Grandstream GXP 2000 phones using it.
Wheneverwe transfer an incoming call using the builtin GXP2000 transfer
button or using the Asterisk blind transfer, the caller is disconnectedif
theextension is busy.Is this how blind
First off, I agree w/ Gonzalo – softphones didn't work out for me
either. One thing that did work great tho was a combo.
We share SIP phones at the office in a 1:4 ratio. You're probably
asking – how do you know when a ringing phone is for you? Well,
everyone in our office gets an XLite
Mike Clark [EMAIL PROTECTED] wrote:
If you aren't going for the echo cancellation,
then I think either card will do fine.
We are now deploying only the A200 because we
never know if echo will be an issue or if it can
be tuned away
Thanks, Mike.
That's a good point in favour of the A200
John Novack [EMAIL PROTECTED] wrote:
Though many will probably disagree, you will be
LOTS better off with the Sangoma A200 It is MUCH
more forgiving regarding Motherboards and the
PCI 2.2 requirement,
That's one of my concerns. I'm working with
refurbished hardware, so don't have much freedom
Chris Bagnall [EMAIL PROTECTED] wrote:
The site in Northampton with 3 FXO has been an
absolute nightmare over the last 9 months the
system's been in place.
Once asterisk 1.2 was deployed, things improved
remarkably.
Do you think that was improved code in v1.2, or
the result of your
On Tuesday 02 May 2006 16:42, hugolivude wrote:
We share SIP phones at the office in a 1:4 ratio. You're probably
asking – how do you know when a ringing phone is for you? Well,
everyone in our office gets an XLite softphone, and I direct calls to
make BOTH the SIP phone AND the XLite ring.
I am not by any means recommending this to anyone but I wanted to
publish this for reference.
I have an Asterisk system connected to a provider via IAX trunks. There
are 32 phones on our network and we have about 400 calls per day to/from
our system. The hardware running this is a Pentium Pro
All,
I have this issue happening on 2 seperate asterisk boxes, it happen from version 1.2.4 i am currently running version 1.2.7.
What happens is i will be on a call, and all of a sudden I will hear a fast busy, the person that i was talking to can still hear me fine. It doesn't really matter
Maybe some kind of callwaiting/threewaycalling activated on that? The
system is identifying the hang up as a flash.
-Original Message-
From: Matt [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc:
Sent: Tue, 2
When I hit *1 in my system, I got a beep to let me know that the
recording started. Is this not happenning to you?
[]'s
MM
-Original Message-
From: Mark Ackroyd [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Cc:
I guess that would work to if I knew about any caller-id popup apps!
Wasn't that much overkill actually, we all had XLite installed for our
failed soft-phone trial. Besides some users travel and take the
XLites w/ them...
Anyway the idea's the same and that's what's important.
Howard
On
Setting the country=se in [general] context inside indications.conf
didn't work?
[]'s
MM
-Original Message-
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Cc:
Sent: Tue, 2 May 2006 15:32:24 +0200
Delivered: Tue, 02 May 2006 07:36:20
Subject:[Asterisk-Users]
I just tested this out and I am working again.
Thanks for the great advice.
Thanks Again -Jason
On Sun, 2006-04-30 at 19:27 +0200, Mathieu Chouquet-Stringer wrote:
[EMAIL PROTECTED] (Patrick) writes:
Looks like an SELinux issue. Try booting with selinux=0 or
I've had a heck of a time getting a SIP channel to work in Asterisk
1.2.7.1 (Redhat 9.0). I've done it successfully a number of times on
pre 1.2 versions of Asterisk so perhaps it's version related. Any
insights are welcome!
At first I wasn't able to dial out on the SIP channel _whenever_ I
I have a PSTN termination provider foo which will
accept standard U.S. calls in the form 110 digit
ph#.
I have an outbound route named foo, with dial
pattern 5|., with the only entry in trunk sequence
being IAX2/foo.
I have an X-lite local extension, on which I can
dial
5110 digit ph#,
Hi All,
I have a number of SPAX00X units (spa1001, 2002, etc) and about 30 odd
PAP2-NA units all hooked up to Asterisk. As you can imagine, setting
them up took a while, and changing settings on them also takes a
while. In order to prepare for future deployments, I'd like to use XML
provisioning
Remco Barende wrote:
Found it!
It seems that Asterisk is looking at the date / time stamp of the call
file to process the call?? I was simply moving the call files hoping
it would just work (tm)
I guess that the call files created on the samba share I created carried
the time/date stamp
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