I know for a fact that this product can do attended transfers. (as I
have seen it in use)
http://www.asteriskit.com.au/Page/The-Receptionist-Console
PaulH
On Wed, 2008-02-27 at 19:41 +, Chris Bagnall wrote:
Greetings list,
I've been playing around this afternoon with Flash Operator
He said in the OP that he is using a Digium TDM400. I guess turning
EC off could help diagnose that pretty quickly.
I have diagnosed this issue (when it was hardware failure) by creating
different dialout extensions in the dialplan, to hit zap/1, then zap/2
etc.
Then I found it was only on
I do not know if this will make a difference but the protocol-variant
for Mexico should be:
protocol-variant mx,10,4
You only get 10 digits from the phone company.
On Wed, 2008-02-27 at 18:03 -0800, Andres Tello Abrego wrote:
protocol-variant mx,20,4
--
Telecomunicaciones
phone1.cfg:
call.callsPerLineKey=1
Raúl Gómez C. wrote:
Michael,
I haven't used nor configured a Polycom phone, but you should check in
/etc/asterisk/sip.conf the call-limit param of the phone's config.
On Thu, Feb 28, 2008 at 12:31 PM, Michael Munger
[EMAIL PROTECTED] wrote:
We
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Thanks Carlos...
Using mx,10,4 didn't work.
Chan 31, class 'mfcr2', variant 'mx,10,4', end 0, caller 0, from '' to ''
Loading protocol mfcr2
Thread for channel 0
MFC/R2 Chan 31: Call control(9)
MFC/R2 Chan 31: Unblock
MFC/R2 Chan 31: 1001
Hi,
On Wed, Jan 16, 2008 at 10:11 PM, [EMAIL PROTECTED] wrote:
Hello users!
Recently I read that AgentCallbackLogin is going to be deprecated soon.
Wanting to set up a few callback type queues, I set them up as suggested
in queues-with-callback-members.txt.
I was able to set the
Hi
On Wed, Feb 27, 2008 at 5:41 PM, Chris Bagnall [EMAIL PROTECTED] wrote:
Greetings list,
I've been playing around this afternoon with Flash Operator Panel, trying to
get it to do attended transfers. I am running the latest version.
Has anyone managed to get this working reliably, and
shadowym wrote:
A bit hard to describe. Using a SIP hardphone I log into my voicemail at
which point Allison says you have x messages.. There are various other
prompts that exhibit the same problem but that is one easy to explain and
reproduce one. The problem is there is a slight
There is a problem with Dell Servers and I experienced the same issue as you speak of with Dell 2900 Servers. I keep getting PCIe Training Error and Dell had to replace the raid controller. Take a look at IBM 3550.
Original Message
Subject: Re: [asterisk-users] Had it with Dell
Hi,
I have written a C code which would let me connect to the Asterisk Manager
Interface. The code compiles successfully but on running the code I get
unauthorized login shown in the Asterisk command line console.
Here is my C code:
#includestdio.h
#includenetdb.h
#includeunistd.h
Hi,
we started with HFC cards and were upset, how many patches we had to
introduce in the source.
Then we found mISDN and we are operating an AVM Fritz-Card and an HFC card
out of the box
via mISDN interface. No change at asterisk's code was necessary.
Hope, this helps,
Michael
Dear all,
We are trying to use res_config_ldap in asterisk 1.6.0 beta 2/4 but
are having some problems as under. Just wanted to know if these are
known issues or we are doing something wrong.
1. the res_config_ldap.sample asterisk config file follows a different
schema. The file posted on
Hmm, I don't know if the zaptel fax detection will trigger. This is going
through my company's Rolm CBX switch, using a plain T1 cable (yanno, RBS, EM
Wink, and all that jazz) between the two systems as a Tie line, so that may
mess with the fax stuff. The Rolm's just wired to take a special
Fernando Berretta wrote:
Tzafir,
I'm sorry, my question wasn't clear.
Apparently Asterisk 1.6.0b2 and b4 has support for t38 because of some
modifications on app_fax so the questions are:
1 - If I use Asterisk 1.6.0b2 o b4 and a fax is received from a FXO Card
and this FXO port is
On Tue, Feb 26, 2008 at 07:19:58PM -0600, Shaun Ruffell wrote:
John Covici wrote:
Hi. I am using asterisk 1.4 (latest as of today) and zaptel 1.4
(latest as of today) and I cannot dial out using my 400P card with one
fxs module and one fxo module. I am using kernel 2.6.24 and get the
On Wed, Feb 27, 2008 at 08:13:49AM +0200, Louwrens Benadé wrote:
Well, on an E1 PRI config your D-channel is indeed assigned to channel 16,
the center channel. On a T1, your data channel is on channel 24, the last
channel.
Did you restore your zaptel config from samples or another source?
Hi All;
How can I configure Asterisk in that way:
If I entered code (from my mobile when I call to the
Asterisk or from any Internal Phone), then the machine
do restart. I need this when I am far from the office
and I need to restart the machine and I do not have
Internet connection.
Any help?
try System() application in dialplan to execute linux reboot command.
On Wed, Feb 27, 2008 at 5:24 PM, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi All;
How can I configure Asterisk in that way:
If I entered code (from my mobile when I call to the
Asterisk or from any Internal Phone), then
I assume that you've already updated the firmware to 2.1.2 - There
were several problems and crash-bugs with previous firmware versions,
but I have found 2.1.2 to be relatively stable.
Regards,
Steve
___
-- Bandwidth and Colocation Provided by
Hello List.
I have created a patch some time ago, to use the ISDN feature call deflection
or partial rerouting, as it Is also known, to make proper call transfers on
PRI, without using an extra channel for the outgoing call.
Back then I ported the function zapCD from bristuff, to a normal
i believe your problem is at the hardware/driver/provider level so you will
be looking at the :
zaptel.conf zapata.conf files
i don't know if this will help but here is my config for a E1 euroISDN line:
zaptel.conf
span=1,0,0,ccs,hdb3 # First E1 Port
span=2,0,0,ccs,hdb3 #
On Wed, Feb 27, 2008 at 10:25 AM, linuxian iandsd [EMAIL PROTECTED] wrote:
i believe your problem is at the hardware/driver/provider level so you will
be looking at the :
zaptel.conf zapata.conf files
Thanks for the advice. I will give it a try this evening.
my second advice is :
On Wed, Feb 27, 2008 at 10:25:28AM +, linuxian iandsd wrote:
i believe your problem is at the hardware/driver/provider level so you will
be looking at the :
zaptel.conf zapata.conf files
i don't know if this will help but here is my config for a E1 euroISDN line:
But this is not the case
Rizwan Hisham wrote:
I am having a strange problem. I am using my asterisk server AST1 to
register with another asterisk server AST2 using 2 accounts (2 register
commands in sip.conf). I have made 2 local users in AST1, and configured my
dialplan in such a way that both local accounts on AST1
Try EyeBeam. It is the paid version of X-Lite.
- Original Message -
From: Mike
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Sent: Friday, January 18, 2008 10:57 PM
Subject: [asterisk-users] Looking for business-grade SIP Softphone
Hi,
I am looking for a
On Wed, Feb 27, 2008 at 12:01 PM, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
I had a theory on how this had happened in the specific case. But so far
the OP has not confirmed or denied it.
I did deny it. Please see it here:
I haven't tried all of them but.. at least Linksys ATA AdminGuide
doesn't specify such limitation
FAX Enable T38
To enable the use of the ITU-T T.38 standard for faxing, select yes.
Otherwise, select no.
The default is yes
Thomas Kenyon wrote:
Fernando Berretta wrote:
Tzafir,
I'm
On Tue, Feb 26, 2008 at 12:27:16PM +, Andres Jimenez wrote:
Comes from a previous message.
On Tue, Feb 26, 2008 at 12:25 PM, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
Here's my guess:
You built Asterisk vs. a newer Zaptel (that happened to have the
Astribank drivers).
Now
On Wed, Feb 27, 2008 at 12:45 PM, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
Are the new zaptel drivers loaded?
cat /sys/module/zaptel/version
Yes. I did fix it upgrading it back (and rebuilding asterisk and
libpri) to zaptel 1.4.8
--
Andres Jimenez
GPG : http://www.andresin.com/gpg/[EMAIL
Have a look at our Zoiper (http://www.zoiper.com/oem.php) - it does all
4 items you are looking for.
Zoa
Dovid B wrote:
Try EyeBeam. It is the paid version of X-Lite.
- Original Message -
*From:* Mike mailto:[EMAIL PROTECTED]
*To:* 'Asterisk Users Mailing List -
Dear Atik;
Thanks a lot, another thing if possible:
A code to be entered, then it will execute some
asterisk commands like: asterisk -rx iax2 reload?
Any advise?
Regards
Bilal
-
try System() application in dialplan to execute linux
reboot command.
On Wed, Feb 27, 2008 at 5:24
On Wed, Feb 27, 2008 at 09:32:08AM +0100, Hasse Hagen Johansen wrote:
Hi
I've got a cheap card from x100p.com for my pots line. I haven't found a
definate answer if it is possible to get danish(DTMF without signaling
before it). I have had a look at bug #9 but that is
written longtime ago.
I recall that this is now part of Asterisk (1.4 or 1.6 or both). It
really is a great feature rather than using two channels in trunk to
trunk transfer.
Thanks,
Steve Totaro
On Wed, Feb 27, 2008 at 4:38 AM, Jon Schøpzinsky [EMAIL PROTECTED] wrote:
Hello List.
I have created a patch some
Hi
I've got a cheap card from x100p.com for my pots line. I haven't found a
definate answer if it is possible to get danish(DTMF without signaling
before it). I have had a look at bug #9 but that is
written longtime ago.
I am running zaptel 1.4.7 and asterisk 1.4.14 BRIstuffed 0.4.0-test4 both
Hi Ian,
I'm out of the office for the day, but as soon as I can I'll check my logs
looking for Zombie calls, although my GXP-2000 are configured with static IP
so there's no (re)registry in my case (host=PhoneIPaddr in sip.conf).
On Thu, Feb 28, 2008 at 2:57 AM, Ian [EMAIL PROTECTED] wrote:
On Wed, 2008-02-27 at 08:50 -0500, Steve Totaro wrote:
I recall that this is now part of Asterisk (1.4 or 1.6 or both). It
really is a great feature rather than using two channels in trunk to
trunk transfer.
This is often called a Two B-Channel Transfer, or TBCT. As long as
your PRI provider
Hi list,
After failing to get a Sipura/Linksys SPA3000, which I've configured
as a PSTN gateway, to pass on the Caller ID, I decided to try my luck
with a Linksys SPA3102 after hearing some promising stories.
Unfortunately, I've run into a completely new problem: it seems
Asterisk won't
I am setting up an Asterisk server to provide voice messaging in a campus
setting. I am interested in how others have Asterisk set up in regards to
firewalls and web interface access to minimize security risks.
___
-- Bandwidth and Colocation Provided
Now to come up with a way to busy out individual channels via this
or another method. This is one feature that is in great demand.
Thanks,
Steve Totaro
On Wed, Feb 27, 2008 at 10:03 AM, Jon Schøpzinsky [EMAIL PROTECTED] wrote:
Actually this isnt the same as Two B-Channel transfer.
This is
Actually this isnt the same as Two B-Channel transfer.
This is done by sending a FACILITY message to the ISDN, which in hand then
disconnects the call and sends it to the number provided in the call deflection
message.
All b-channels are closed when the FACILITY message is sent.
You can also
http://www.geeks.com/details.asp?invtid=8835Y11-R
Use Promo Code: 8835DEAL to bring it down to $219 (plus shipping)
I picked up two:
8835Y11-R -- IBM eServer 325 Dual Opteron 2.0GHz 1GB 12 2 $219.99
$439.98 (a little less than $500 for two)
Thanks,
Steve Totaro
Hello all,
i today have searched on the internet about a solution to let asterisk act as
a SMSC (Short MessageServiceCentre) to deliver SMSes directly to GSM Phones.
I only have found some cases with use of an extern SMSC (i.e. by the Mobile
Net Provider)
Is there a possibillity to do that,
bilal ghayyad wrote:
Hi All;
How can I configure Asterisk in that way:
If I entered code (from my mobile when I call to the
Asterisk or from any Internal Phone), then the machine
do restart. I need this when I am far from the office
and I need to restart the machine and I do not have
On Wed, Feb 27, 2008 at 10:26 AM, Alex Balashov
[EMAIL PROTECTED] wrote:
bilal ghayyad wrote:
Hi All;
How can I configure Asterisk in that way:
If I entered code (from my mobile when I call to the
Asterisk or from any Internal Phone), then the machine
do restart. I need this
Steve Totaro escribió:
On Wed, Feb 27, 2008 at 10:26 AM, Alex Balashov
[EMAIL PROTECTED] wrote:
bilal ghayyad wrote:
Hi All;
How can I configure Asterisk in that way:
If I entered code (from my mobile when I call to the
Asterisk or from any Internal Phone), then the machine
do
Do you mean individual B-channels?
That could be done in dialplan, with the ZapCD command... When its done that is
:)
Jon
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: 27. februar 2008 16:19
To: Asterisk Users Mailing List -
arkda wrote:
Nothing in the console aside from what I've posted. When a DTMF tone is
played the server freezes instantly, hard reboot required.
Distribution is SuSE 10.2, kernel 2.6.18.8-0.7-default
The actual dialplan on this server is very simple, only one phone and a
few Dial
Steve Totaro wrote:
The way I read the OP, he wishes to reboot the box, not just restart
asterisk. In which case, simply having
exten=777,1,Authenticate(whatever)
exten=777,n,System(reboot)
Ah, yes, you are indubitably correct.
--
Alex Balashov
Evariste Systems
Web:
Is there a way to detect Service Provider message such as invalid number, using
AMD or some other application.
Regards,
Sanjay.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or
I have setup a few Asterisk systems for customers using Digium TDM400 cards
and Aastra phones. No problems with sound quality at all except at this one
site.
Every time I try their system I don't hear any problems but they tell me
that it is really bad. They describe it a a loud scratching sound.
Greetings list,
Have there been any further developments recently regarding presenting the
original caller's caller ID to SIP devices after an attended transfer? I've
googled around on the topic, but most of the threads I've found (some from
this very list) are all dated back in mid-2006 and I
Hello,
I am having a problem with an call openend from the siptapi application
(http://www.enum.at/SIP-TAPI.479.0.html) to a phone which is refered to
another external number after the phone picks up. I see in the Sip
Traffic that everything is fine with the exosip client used in siptapi
and
We have an asterisk system and Polycom phones that were provisioned by
someone else. They want to get call waiting to work, but for the life of
me, I cannot figure out why the Polycom is returning a SIP 486 Busy Here
when you call and the person is already on the phone.
I have the feeling
Sure Shaun, I'll give it a shot. I'll contact you directly to let you know
the results.
On Wed, Feb 27, 2008 at 10:33 AM, Shaun Ruffell [EMAIL PROTECTED] wrote:
arkda wrote:
Nothing in the console aside from what I've posted. When a DTMF tone is
played the server freezes instantly, hard
Hi, all
I want to configure a few FXS ports in an Antribank-16 to be able to
receive faxes sent throught a PRI:
E1 ==Zap * ==FXS * ==Fax machine
My asterisk box has a Digium TE120P (for the PRI).
Versions are *= 1.4.17 | Zaptel=1.4.8 | libpri=1.4.5
The Astribank is not configured yet,
Out of stock now.
Any war stories about running Asterisk on a serious blade setup?
Will you ever hire Wesley Snipes to flog them at a convention?
-HJC
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
Quoting Jaap Winius [EMAIL PROTECTED]:
Hi list,
After failing to get a Sipura/Linksys SPA3000, which I've configured
as a PSTN gateway, to pass on the Caller ID, I decided to try my luck
with a Linksys SPA3102 after hearing some promising stories.
Unfortunately, I've run into a completely
Royce Souther wrote:
Are there any tests that can be done to pinpoint the problem?
Swap out the card - that usually fixes anything you have control over.
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
Andres Jimenez wrote:
exten= 11,1,Dial(Zap/41)
exten= 22,1,Dial(Zap/42)
That's what I do with my Adit channel bank.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
On Wed, Feb 27, 2008 at 05:25:08PM +, Andres Jimenez wrote:
Hi, all
I want to configure a few FXS ports in an Antribank-16 to be able to
receive faxes sent throught a PRI:
E1 ==Zap * ==FXS * ==Fax machine
My asterisk box has a Digium TE120P (for the PRI).
Versions are *= 1.4.17
You do not have to do anything else. When Asterisk detects a fax tone
it will disable echo cancellation on those channels so the fax can go
through. Just make sure that the Astribank is the sync source for
timing and you should be able to send and receive faxes.
In your dialplan
On Wed, Feb 27, 2008 at 6:58 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
Let's say I configure the FXS ports in the Astribank as channels 41
42 (only 2 for the moment).
Hmmm... One full E1 span will get you channels 1-31 (even if you use
only up to channel 24, the span will register 31
I've got this in extensions.conf:
[macro-stdexten]
exten = s,1,Dial(${ARG2},30,p)
exten =
601555,1,Macro(stdexten,200,SIP/200SIP/201SIP/203SIP/${VOICEPULSE_GATEWAY_OUT_A}/+1504555)
Where the real numbers have been replaced with 555. What I'm trying
to do is ring my cell phone in
On Wed, Feb 27, 2008 at 06:16:47PM +, Andres Jimenez wrote:
On Wed, Feb 27, 2008 at 6:58 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
Let's say I configure the FXS ports in the Astribank as channels 41
42 (only 2 for the moment).
Hmmm... One full E1 span will get you channels
Michael,
I haven't used nor configured a Polycom phone, but you should check in
/etc/asterisk/sip.conf the call-limit param of the phone's config.
On Thu, Feb 28, 2008 at 12:31 PM, Michael Munger
[EMAIL PROTECTED] wrote:
We have an asterisk system and Polycom phones that were provisioned by
Check for IRQ issues, move the card to a different slot.
You could ask permission to record calls so maybe you can hear the
sound yourself.
I would then go ahead and swap out cards. I have had TDM400 with bad
modules and also bad ports on the cards themselves, so it could a
hardware issue.
Greetings list,
I've been playing around this afternoon with Flash Operator Panel, trying to
get it to do attended transfers. I am running the latest version.
Has anyone managed to get this working reliably, and if so, would you mind
sharing how you did it please?
Alternatively, are there any
Am Mittwoch, den 27.02.2008, 16:30 +0100 schrieb Hans-Peter Straub:
Hello all,
i today have searched on the internet about a solution to let asterisk act as
a SMSC (Short MessageServiceCentre) to deliver SMSes directly to GSM Phones.
I only have found some cases with use of an extern SMSC
On Wed, 27 Feb 2008, Royce Souther wrote:
I have setup a few Asterisk systems for customers using Digium TDM400 cards
and Aastra phones. No problems with sound quality at all except at this one
site.
Every time I try their system I don't hear any problems but they tell me
that it is really
I think it may have been a NAT and/or reinvite issue. I've now
forwarded udp 1-2 to my pbx and turned off reinvite for all.
NAT issues can be so difficult to diagnose sometimes. I'll be glad to
see it go with ipv6.
Randall
Randall Smith wrote:
[macro-stdexten]
exten =
Quoting Tim Johnson [EMAIL PROTECTED]:
I see you put a password line in your sip.conf, but I do not see a
username line. Also, you might want to check the port #'s for both the
Line 1 and PSTN line. I use 5060 and 5061, respectively. Hopefully
this either helps, or puts you on the right
Perhaps irq sharing?
Royce Souther schreef:
I have setup a few Asterisk systems for customers using Digium TDM400
cards and Aastra phones. No problems with sound quality at all except
at this one site.
Every time I try their system I don't hear any problems but they tell
me that it is
Hello! I've run into a problem where a user is making an outbound call at the
same time that an inbound call is being made on the same analog line. It
appears that as the zap channel is opened for the outbound call, it is simply
answering the inbound call. Obviously, both parties involved in
This is called glare. What you should do is reverse your outbound hunt. If you
dial zap/g0 simply use zap/G0. The capitol G makes the line go 5 4 3 2 1
instead of 1 2 3 4 5. You may still see glare but this usually reduces it.
James Finstrom
Rhino Equipment Corp.
http://www.rhinoequipment.com
James Finstrom wrote:
This is called glare. What you should do is reverse your outbound hunt. If
you dial zap/g0 simply use zap/G0. The capitol G makes the line go 5 4 3 2 1
instead of 1 2 3 4 5. You may still see glare but this usually reduces it.
James Finstrom
Rhino Equipment Corp.
This is a well know issue in analogue trunks, called collisions or
glare. As you say, more is the traffic more are probability of
collisions. One trick to reduce this problem is to reverse the outgoing
hunting group against the incoming hunt group.
Jorge Mendoza
Tim Nelson wrote:
Hello! I've
Chris Bagnall wrote:
Greetings list,
I've been playing around this afternoon with Flash Operator Panel, trying to
get it to do attended transfers. I am running the latest version.
Has anyone managed to get this working reliably, and if so, would you mind
sharing how you did it please?
Could this be ECFO?
Echo Canceler Freak Out, this happens when the rxgain is too high and
the echo canceler freaks out. Some users describe it as screeching,
feedback, static, or other useless terms. If users report static
on a system where there cannot be static (all digital, PRI, SIP,
This is something that has been bugging me for awhile. I have noticed it on
multiple systems with different hardware, with and without zaptel cards, and
various versions of Asterisk in the 1.2 and 1.4 branches.
A bit hard to describe. Using a SIP hardphone I log into my voicemail at which
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I have a astunicall-1.4 setup with a te110p to a nortel pbx in Mexico.
(Hate R2!).
This is what I get when trying to call to * box using testcall:
./testcall
Chan 31, class 'mfcr2', variant 'mx,20,4', end 0, caller 0, from '' to ''
Loading protocol
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