Announced today, Linksys SPA9000 IP Telephony Key System
http://voipspeak.net/index.php?/content/view/60/2/
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
Steven wrote:
Just do:
exten = _12xx,2,Dial(${TRUNK}/0${EXTEN}|30,r) ; adding zero
exten = _012xx,2,Dial(${TRUNK}/${EXTEN}|30,r) ; not adding zero
The zero is added before ${EXTEN}.
I have only ever used the stable versions and have always done it this way.
Never trust anyone that tells you
On Wed, Jan 04, 2006 at 11:34:44AM -0800, Mike Fedyk wrote:
I presume you mean 2.4 and 2.6.
Six months ago the Stable release of Debian couldn't run 2.6 kernels
without installing a few updated packages from their backports.org
repository. There has been a release since then that includes
Dear All,
Now I Have Asterisk and wow ... it's worked. I Have a simple question. How if we have a IVR for our departement.
Say if someone dialed 204 the IVR will appear and tell the caller to dial
204 - Me [ The IVR Ext ]
205 - MyFriend
Somebody help me please ...
Thanks
have you tried to parse the traffic what phone is requesting from your
tftp-server ?
maybe you get a hint where
[EMAIL PROTECTED] wrote on 05.01.2006 03:21:07:
I am working on adding three older Cisco phones to *, two 12SPs and one
30VIP. One of the 12SPs
(griffin) and the 30VIP (scott) is
Jason D. Wolfe wrote:
Hello,
If I use an IAX termination service to connect outgoing VoIP calls to a PSTN
will I have answer supervision so that my script won't initiate too early?
Correct. (At least it should be correct as any decent service provider
will be using PRIs)
Hello,
I bind asterisk sip to all interfaces/addresses by bind=0.0.0.0
(ie. a.b.c.d, a.b.c.e IPs assigned on PC), I experienced problem when UA
registers to a.b.c.e, asterisk sends register response from a.b.c.d and
client ignores reply, because a.b.c.e != a.b.c.d. Is it bug, feature or
I have installed several call centers in the netherlands with the
eyebeam softphone (from the counterpath guys)
It is not free, but very stable, and pretty easy to use.
It works great with asterisk (specially the presence option, so agents
can see whether somebody is actually ready to take a
I don't know if it's possible, but I use a workaround to simulate the
external dialtone:
I use '0' to access external lines
exten - _0,1,ChanIsAvail(Zap/g1)
exten - _0,2,playtones(dial)
exten - _0,3,goto(external_tone|et)
...extensions if some dialed without waiting for dialtone
Have a look at our idefisk softphone. (available for windows, mac and
linux).
freely downloadable from http://www.asteriskguru.com/tools/
We also have a callcenter version, contact me offlist if you want more info.
Greetings
Zoa
Andrey Loginov wrote:
Chris Bagnall wrote:
I've been
Zoa ha scritto:
Something is using up way too much memory, are you sure asterisk is
using 800mb of ram ? it should be ten times less.
Zoa
You're right, I forgot there are also huge mysql tables on the same machine
(with no agi and transcoding) 80 alaw concurrent calls , cdr_mysql,
Hi there everybody,
We are running Asterisk 1.2.1 with a TE410P card attached to one
PRI ISDN line, and many SIP phones. Yesterday we ended up in a situation
where all incoming calls were giving the engaged tone. Every time some
tried to ring in we got:
Jan 4 14:56:32 WARNING[896]
Paul Dugas wrote:
On Wed, 2006-01-04 at 11:59 -0500, Mark Phillips wrote:
Anyone got any VoIP traffic shaping rules for m0n0wall that they could
let me look at please?
Running m0n0wall-1.21 now, I used the wizard to set the base
queues/pipes/rules then added two more rules:
If Dir
I am trying to install G729 licence on my Virtuozzo server running
asterisk but I keep getting an error as it has no eth0. I get the
following error when running register:
[EMAIL PROTECTED] root]# /root/register G729-
Digium Product Registration
Copyright (C) 2004, Digium, Inc.
Analyzing
Thanks Paradise, this seems to have worked a treat!!!
I commented out the:
exten = 110,hint,SIP/110
lines which were in extensions_additional.conf for each sip extension I
had.
This seems to have stopped the crashes which were previously 3-5 times a
day, now:
System uptime: 1 day, 18 hours,
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
Try experimenting with this:
[general]
featuredigittimeout = 1000 ; Max time (ms) between digits for
; feature activation. Default is 500
It seams it works. Thank you.
--
Tomislav Parcina
Hi all,
I am having difficulty getting incoming PSTN calls working.
I have set up an account with a third party provider. In my system, the user
register with SER and use Asterisk for PSTN access, voicemail etc
My provider told me to change my sip.conf as follows
register =
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
CounterPath's X-Pro Tapi softphone has this I think?
http://www.xten.com/index.php?menu=X-Series (select the EU region)
I think they have a trial...downloading it now.
Thank you.
--
Tomislav Parcina
[EMAIL PROTECTED]
On Tue, Jan 03, 2006 at 06:28:24PM -0500, Michael Stearne wrote:
On 1/3/06, Technical Support [EMAIL PROTECTED] wrote:
We do a lot of installs on Fedora (slowly becoming our favorite). Initially
clients asked for FC because of compatibility with Red Hat, great package
management, etc.
On Tue, Jan 03, 2006 at 04:33:49PM -, Brett, Gary wrote:
I wish to install asterisk 1.2 (the latest tar.gz from the site not the
CVS version) on an HP box with a TE110P (single port E1/T1)
My question is which OS would be preferred in this configuration Fedora Core
1 or Fedora Core
In article [EMAIL PROTECTED], trixter@
0xdecafbad.com says...
to add to this, given the state of MD5 and its 'security' or lack
thereof, its a bit over simplistic to just say md5 without adding that
its actually 3 md5 hashes... Precomputing is harder (but not
impossible) because of the way
Hi,
we just set up an asterisk with 55 Polycom 500 IP phones.
The blind transfer does not work.
The way we try to blind transfer a call:
1. answer the call
2. press transfer
3. press blind softkey - the display shows Blind transfer to: and cursor is
in the second line
4. enter the number -
Hi all, I have a small query regarding ringing tones on an iaxy2.
I have a customer who uses an iaxy to breakout to pstn via our *.
However the customer complains that he gets no ringing tone whislt
making calls, i just visited the site and can confirm this.
I also have another customer who is
I have to correct myself.
The problem occurs only when we try dial numbers with 10 or 11 at the beginning.
Kib Eki wrote:
Hi,
we just set up an asterisk with 55 Polycom 500 IP phones.
The blind transfer does not work.
The way we try to blind transfer a call:
1. answer the call
2. press
In article [EMAIL PROTECTED]
exch2k3.phoenix.com, [EMAIL PROTECTED] says...
Please contact me off list if you'd like to give it a try.
Any link or something?
--
Tomislav Parcina
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by
What is command when I wona to list something page by page in * CLI?
Something that works like |less or |more.
Have a nice day!
--
Tomislav Parcina
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users
On Thu, 2006-01-05 at 14:05 +0100, Tomislav Parcina wrote:
In article [EMAIL PROTECTED], trixter@
0xdecafbad.com says...
to add to this, given the state of MD5 and its 'security' or lack
thereof, its a bit over simplistic to just say md5 without adding that
its actually 3 md5 hashes...
Hi,
Can you give me some councils of remotely rebooting sip phones in asterisk
server? How to configure sip_notify.conf and sip.conf? Kind regards,
Guan
; Reboot Polycom Phone
Event=check-sync
Content-Length=0
; Untested (Reboot Sipura Phone)
Event=resync
Content-Length=0
; Untested (Reboot
On 5 Jan 2006, at 09:45, Zoa wrote:
Have a look at our idefisk softphone. (available for windows, mac
and linux).
The download links at http://www.asteriskguru.com/tools/
idefisk_beta.php only lead to Windoze versions, how do I get the Maxc
version?
Thanks!
jens
On Thu, Jan 05, 2006 at 02:59:34PM +0100, Tomislav Parcina wrote:
What is command when I wona to list something page by page in * CLI?
Something that works like |less or |more.
Scroll back in your terminal? Use screen if your terminal is not capable
of that?
less /var/log/asterisk/messages ?
Hi BK -
The blind transfer does not work.
The way we try to blind transfer a call:
1. answer the call
2. press transfer
3. press blind softkey- the display shows Blind transfer to: and
cursor is in the second line
4. enter the number- when we enter the second digit of the number
ast guy wrote:
for what purpose logical channels are used?
Call waiting, three-way calling, etc.
___
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Kerry Garrison wrote:
Announced today, Linksys SPA9000 IP Telephony Key System
http://voipspeak.net/index.php?/content/view/60/2/
Do not post advertisements for products on this list, whether you are
selling them or not.
___
--Bandwidth and
The second edition of my Asterisk book VoIP Telephony with Asterisk is now
in print. It's reorganized and expanded.
TKS
Paul Mahler
Paul Mahler
[EMAIL PROTECTED]
www.signate.com
___
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Ok, I've been trying to figure out why my [EMAIL PROTECTED] won't answer the
lines when I can call out and the panel shows the call coming in - well
something bizarre has happened.
I set up inbound routing to ring my extension if a call comes in - and my
extension rings but when I pick it up I
Hi,
You could only take timing from one E1 per card. So you should use :
span=1,1,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
span=3,0,0,ccs,hdb3,crc4
span=4,0,0,ccs,hdb3,crc4
instead of :
span=1,1,0,ccs,hdb3,crc4
span=2,1,0,ccs,hdb3,crc4
span=3,1,0,ccs,hdb3,crc4
span=4,1,0,ccs,hdb3,crc4
Le
Hi. If I use kernel 2.6.15 I cannot compile zaptel modules. I get
the following error(s) using gcc4.
CC [M] /usr/src/zaptel/zaptel.o
/usr/src/zaptel/zaptel.c: In function 'zt_ppp_xmit':
/usr/src/zaptel/zaptel.c:1533: warning: comparison of distinct pointer types
lacks a cast
Single port GSM Gateway support 900 / 1800 GSM mode with
external antenna.
Brand new unit and all of them will be tested before dispatch.
Extremely easy to setup and can be used out of the box
without any configuration. So should be good alternatively of
phonecell or nokia pbx etc..
Hi,
we've connected Sphinx4 through eagi script (modified eagi example) to
Asterisk. Users can now say their wishes - but for gradual evolution we
would like also to provide older way of DTMF navigation too - can we
recognize
DTMF while reading sound in eagi ?
Any advice or examples ?
Thanks
I've found that chanspy crashes asterisk after about 10 channel spys..
asterisk just stops responding, and I have to restart it.
On 1/4/06, Tom Vile [EMAIL PROTECTED] wrote:
correct it only works with bridged calls.
On 1/4/06, Leo Ann Boon [EMAIL PROTECTED] wrote:
Tom Vile wrote:
use
I know the subject of faxing has been covered in some detail, but I was
wondering if anyone has a hardware configuration similar to ours that
has faxes working successfully and would be willing to share any
settings/insight.
We are unable to fax reliably with a Sipura 2100 connected to
John Covici wrote:
Hi. If I use kernel 2.6.15 I cannot compile zaptel modules. I get
the following error(s) using gcc4.
Without telling us exactly what version of Zaptel you are trying to
build. your report is nearly useless.
Zaptel was updated to take these API changes into account
I have not had that issue. Are you saying 10 concurrent channels
being spied on or after the 10th it starts to crash?
On 1/5/06, Matt [EMAIL PROTECTED] wrote:
I've found that chanspy crashes asterisk after about 10 channel spys..
asterisk just stops responding, and I have to restart it.
On
I did get the latest zaptel from cvs, but maybe this isn't up to date
-- sorry for the confusion.
How do y9ou determine the zaptel version for future reference?
on Thursday 01/05/2006 Kevin P. Fleming([EMAIL PROTECTED]) wrote
John Covici wrote:
Hi. If I use kernel 2.6.15 I cannot compile
Wow! Thanks for all the responses! Very informative.
Erik: I'm just looking for simple dial-out and pass-along incoming cell
calls to *. Looks like the doc-n-talk should do it, except I checked
with them and, silly me, the new Samsung t309 phone I just got is not
supported yet. Hopefully it will
This is what I set on my Sipura:
You have to be in as admin and then advanced settings.
On the SIP page change:
RTP Packet Size: 0.010
On the Line Page:
FAX CED Detect Enable: Yes
FAX CNG Detect Enable: Yes
FAX Passthru Codec: G711u
FAX Codec Symmetric: No
FAX Passthru Method: NSE
FAX Process
I tried to get it working for a very long time (over a year) with every
possible set of config parameters I could find both for * as well as for
the Sipura's. Echo cancelling etc. etc. all changed but still problems.
I tried to get it working on an * box with a BRI line.
Finally I have given
Kerry Garrison wrote:
Announced today, Linksys SPA9000 IP Telephony Key System
http://voipspeak.net/index.php?/content/view/60/2/
Do not post advertisements for products on this list, whether you are
selling them or not.
___
--Bandwidth and
Olivier Perrin ha scritto:
Hi,
You could only take timing from one E1 per card. So you should use :
span=1,1,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
span=3,0,0,ccs,hdb3,crc4
span=4,0,0,ccs,hdb3,crc4
instead of :
span=1,1,0,ccs,hdb3,crc4
span=2,1,0,ccs,hdb3,crc4
span=3,1,0,ccs,hdb3,crc4
I recollect that there used to be a fixed, finite limit to the number of call
groups that could exist. Does anyone know if that limitation still exists in
1.2.1, or maybe where I could look in the code to find out if it's a fixed
length array or not? Thanks.
Doug.
From ast_get_group(char *s) in channel.c:
for (x = start; x = finish; x++) {
if ((x 63) || (x 0)) {
ast_log(LOG_WARNING, Ignoring invalid
group %d (maximum group is 63)\n, x);
} else
Is this what
Hi All,
Until now I've only used IAX2 to connect to ITSPs. I've been toying
with a SIP connection to Gizmo Project, but not yet successfully. It
brings to mind a question. At what point does it make sense to consider
a SIP-aware firewall such as those from Ingate?
I'd hate to move away from my
Chris Bagnall wrote:
Single port GSM Gateway support 900 / 1800 GSM mode with
external antenna.
Brand new unit and all of them will be tested before dispatch.
Extremely easy to setup and can be used out of the box
without any configuration. So should be good alternatively of
phonecell or
Ales Vizdal, AVONET, s.r.o. wrote:
I bind asterisk sip to all interfaces/addresses by bind=0.0.0.0
(ie. a.b.c.d, a.b.c.e IPs assigned on PC), I experienced problem when UA
registers to a.b.c.e, asterisk sends register response from a.b.c.d and
client ignores reply, because a.b.c.e !=
I was looking for a way to catch the zap busy return and do a redial.
I would dial out on a zap channel. If the call is busy it would then hangup
the zap channel and ask if I wanted to redial press 1 to redial or hangup
to quit.
On the 1 it would hangup the extension redial the number and call
Until now I've only used IAX2 to connect to ITSPs. I've been
toying with a SIP connection to Gizmo Project, but not yet
successfully. It brings to mind a question. At what point
does it make sense to consider a SIP-aware firewall such as
those from Ingate?
You should be able to run SIP
Hi all,
I have an TE11 card and I installed the zaptel driver from digium.
The zaptel.conf look like:
span=1,1,0,esf,b8sz,yellow
bchan=1-23
dchan=24
when I tried modprobe -v wcte11xp without any error message
and then ztttol
I received the error Red alarm
What would be the problem?
Thanks in
On Thursday 05 January 2006 17:09, Chris Bagnall wrote:
Single port GSM Gateway support 900 / 1800 GSM mode with
external antenna.
Brand new unit and all of them will be tested before dispatch.
Extremely easy to setup and can be used out of the box
without any configuration. So should be
- Original Message -
From: Chris Bagnall [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, January 05, 2006 5:33 PM
Subject: RE: [Asterisk-Users] OT: SIP aware firewalls?
Until now I've only used IAX2 to
Hi Kib,
Can you paste the dialplan string from your sip.cfg (this is the
pattern matching string in the phone setup)?
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Jan 5, 2006, at 5:34 AM, Kib
We have here a TandBerg videoconferencing system connected in asterisk with a Beronet card BN4S0 (4 BRI ports). I`m trying to make a videoconferece (video + audio) with this Tandberg to another Tandberg using the ISDN channels through the BN4S0 BRI Card. But, i'm only obtainingaudio calls, the
Thanks for that. I wonder if I could just change the x63 to something higher...
-Original Message-
From: BJ Weschke [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 05, 2006 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Group Limit
- Original Message -
From: Chris Bagnall [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, January 05, 2006 5:33 PM
Subject: RE: [Asterisk-Users] OT: SIP aware firewalls?
Until now I've only used IAX2 to
Hello Kevin ,
On Thu, 5 Jan 2006, Kevin P. Fleming wrote:
Ales Vizdal, AVONET, s.r.o. wrote:
I bind asterisk sip to all interfaces/addresses by bind=0.0.0.0
(ie. a.b.c.d, a.b.c.e IPs assigned on PC), I experienced problem when UA
registers to a.b.c.e, asterisk sends register
My snom phone shows how many messages, it is a snom 360, there is the MWI light and on the lcd display it will say 5 New and 5 old messages.
On 1/4/06, Aaron Daniel [EMAIL PROTECTED] wrote:
If the voicemail is stored locally on the server that the phone isregistering to, the phone should
You could use a cisco ata 186.
There aren't very cheap, but I have made them work on several of my
customer sites with faxes.
The ata just registers to the * server as a SIP endpoint.
Also, echo cancelling and other intelligent things are bad when
dealing with faxes and modems.
Just use the
I'd like to have Asterisk log useful messages during operation.
Is there any way in extensions.conf that I can manually log messages to a file,
say via syslog()? The console output is ugly, with all the extra Executing
NoOp(SIP/pstn.voip.com-08a28bd0, crud at the front of each line. I'm not
That's cool... the cisco's only turn a light on.
Aaron
Joe Pukepail wrote:
My snom phone shows how many messages, it is a snom 360, there is the
MWI light and on the lcd display it will say 5 New and 5 old messages.
On 1/4/06, *Aaron Daniel* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
At 11:48 PM 01/04/2006, you wrote:
I have a TDM400P board with two FXO modules. But the modules are not
detected when the kernel modules starts. Any ideas?
When I bought my TDM it had the 2 modules in sockets 3 and 4, as soon
as I moved them to sockets 1 and 2 it started working.
Ira
I will send you the alpha version offlist if you promise to give me some
feedback :)
Zoa
Jens Vagelpohl wrote:
On 5 Jan 2006, at 09:45, Zoa wrote:
Have a look at our idefisk softphone. (available for windows, mac
and linux).
The download links at http://www.asteriskguru.com/tools/
Hi,
I have tried to use UserEvent() command to send data
to Asterisk Manager from my dialplan.
It works fine if the body only contains 1 line but I
dont know how to send multiple arguments in the body.
I have tested:
UserEvent(eventname|body1|body2)
in the iaxy's context, do you Answer before Dial? I think this might
remove ringing indications. I think you either Dial first, or if Answer
has to be first, add the r option to the Dial cmd?
Hope this helps :)
bails wrote:
Hi all, I have a small query regarding ringing tones on an iaxy2.
Because the Polycom softkey menus were so cumbersome, we chose to use
Asterisk's attended and blind transfers facility-wide. we press ## for
blind transfer (and Allison asks Transfer, and you type the exten num)
and ** for attended transfer.
One more reason we chose this setup was if we
I use polycoms, but I imagine the process is similar for many.
These three lines are the content of my sip_notify.conf:
[polycom-check-cfg]
Event=check-sync
Content-Length=0
An example SIP friend is defined as [112], so we could now type, from
the CLI:
sip notify polycom-check-cfg 112
or to
I don't find the console output ugly, maybe messy, but never ugly :P If u don't like those NoOp, just take them away from ur extensions.conf. BTW, to save the console output to a given file, just edit your logger.conf file. Say you only want the console output, then just add to your filename the
Hi,
I have developped an application that monitors the
status of my queues through the events triggered on the Manager
Interface.
This way, I can know the status of my Agent real
time.
Now, I have a new requirement that I must allow a
manager to click on the Agent he wants to monitor and
Well,
I want the output that the NoOp's generate. I want to be able to manually log
lines to a file through some mechanism. I just wish I could do it without all
the extra NoOp stuff at the front.
I just
tried using:
mylogfile = verbose
in
logger.conf but all I got was the
hi,
I have hacked the interface for the TE110P board to see how much of the
Falc56 that I could access as I want a driver with a different design
than the zaptel. I am willing to contribute to a hardware interface
manual if someone else want to pick up the task.
Jan
Hello all, is anyone aware of any open source call
accounting software for Asterisk? Something that can parse out Asterisk's
call detail records and generate on-demand reports?
Thanks,
Mark
___
--Bandwidth and Colocation provided by
Then stop looking for easy solutions and get your hands dirty changing your c files Alyed Well, I want the output that the NoOp's generate. I want to be able to manually log lines to a file through some mechanism. I just wish I could do it without all the extra NoOp stuff at the front. I just
Not
everyone is a C programmer extraordinairre.
-Original Message-From: Alyed Tzompa
[mailto:[EMAIL PROTECTED]Sent: Thursday, January 05, 2006
11:59 AMTo: Douglas Garstang;
asterisk-users@lists.digium.comSubject: RE: [Asterisk-Users]
Asterisk DebuggingThen stop
I'm trying to
record calls for SPECFIC agents, which queues.conf and agents.conf don't seem to
support. Someone suggested I just put a monitor() command before the Dial() so
that when the Queue dials the agent, it will start
recording.
exten =
a00090101,1,Monitor(wav||m)
exten =
I work for a telecom company that allows me to peer my Asterisk box to
their system for free. Pretty neat. I have everything working except
that I can't get inbound VoIP calls using the DID number that my
company assigned for me. Today, I finally discovered the source of
the problem: For
I have got one and and is working fine. It's exactly for
cards lying around still inside their 12-month contracts..
Actually it's full extend PnP-SIM-card-GSM-gateway(I forgot to unlock the
pin:) so keep this in mind). There are 2 fxs ports, but I use just one;
points to a SPA3000. The other
sounds like a digitmap issue.
We looked at using # originally, but interferred with too many IVR
type applications from people.
On Jan 5, 2006, at 12:00 PM, Mojo with Horan Company, LLC wrote:
Because the Polycom softkey menus were so cumbersome, we chose to
use Asterisk's attended and
They're not? They have no business in an open source world then ;-}
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Douglas Garstang wrote:
Not everyone is a C programmer extraordinairre.
-Original Message-
*From:* Alyed Tzompa [mailto:[EMAIL PROTECTED]
*Sent:*
No but shell scripts are pretty easy and will cleanup your file for you.
On Jan 5, 2006, at 1:07 PM, Douglas Garstang wrote:
Not everyone is a C programmer extraordinairre.
-Original Message-
From: Alyed Tzompa [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 05, 2006 11:59 AM
To:
I have got one and and is working fine. It's exactly for
cards lying around still inside their 12-month contracts..
Actually it's full extend PnP-SIM-card-GSM-gateway(I forgot to unlock the
pin:) so keep this in mind). There are 2 fxs ports, but I use just one;
points to a SPA3000. The
I do agree, plus even if you don't know anything about scripting there are plenty of shell tutorials out thereAlyed No but shell scripts are pretty easy and will cleanup your file for you.On Jan 5, 2006, at 1:07 PM, Douglas Garstang wrote: Not everyone is a C programmer extraordinairre.
Ok, I've been trying to figure out why my [EMAIL PROTECTED] won't answer the
lines when I can call out and the panel shows the call coming in - well
something bizarre has happened.
I set up inbound routing to ring my extension if a call comes in - and my
extension rings but when I pick it up I
Please contact me off list if you'd like to give it a try.
Any link or something?
The installation process might need more documentation, so I
asked interested parties to contact me off list. That way
I can improve the documentation before the general public
attempts to install it.
If
Looks like a digitmap problem within the Polycom configs.
On 1/5/06, Kib Eki [EMAIL PROTECTED] wrote:
Hi,
we just set up an asterisk with 55 Polycom 500 IP phones.
The blind transfer does not work.
The way we try to blind transfer a call:
1. answer the call
2. press transfer
3. press
The new meetme i feature in asterisk1.2.1 for annoucing user
join/leave
is good, but the initial steps to record the name and confirm seems
lenghty,
the user shoudl just say the name and get into the conference, How can
i
disable the confirmation of the name recorded before entering the
On Friday 06 January 2006 00:19, Jean-Michel Hiver wrote:
I have got one and and is working fine. It's exactly for
cards lying around still inside their 12-month contracts..
Actually it's full extend PnP-SIM-card-GSM-gateway(I forgot to unlock the
pin:) so keep this in mind). There are 2 fxs
Hi,
Thczv F. Thczv wrote:
Would there be any other nasty consequences of making that change?
More importantly (perhaps), is there any way to make the change in
[EMAIL PROTECTED] without doing a recompile (and potentially screwing up
my system beyond my ability to repair it)?
We modified
Hi,
Thczv F. Thczv wrote:
Would there be any other nasty consequences of making that change?
More importantly (perhaps), is there any way to make the change in
[EMAIL PROTECTED] without doing a recompile (and potentially screwing up
my system beyond my ability to repair it)?
We modified this
I know
plenty about scripting. Pick your interpreted language
However, if the functionality already existed in Asterisk, a script
wouldn't be necessary. I'm not at the point yet where I want to start developing
scripts for this.
-Original Message-From: Alyed Tzompa
I thought this problem with PRI and channels getting out of sync was fixed in
the 1.2.x release of Asterisk. Here are the errors:
Jan 5 13:59:05 WARNING[1253]: chan_zap.c:8360 pri_dchannel: Ring requested on
channel 0/2 already in use on span 1. Hanging up owner.
Jan 5 13:59:11
On Thursday 05 January 2006 21:31, stotaro wrote:
I have got one and and is working fine. It's exactly for
cards lying around still inside their 12-month contracts..
Actually it's full extend PnP-SIM-card-GSM-gateway(I forgot to unlock
the pin:) so keep this in mind). There are 2 fxs
We looked at using # originally, but interferred with too many IVR
type applications from people.
That's why we switched to ##, it's almost as quick to hit it twice as
once, and doesn't interfere with any (the very few) IVRs my clients
access regularly :)
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