Does anyone know if is possible to purposely set red alarm status on PRI
circuit for testing purposes (other than unplugging it). I have looked for a
console command which might allow this
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a
Once the call is hung up it is too late. I need to interpret the SIP
response codes prior to hangup so I can play an appropriate recorded voice
announcement.
On 4/9/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
Eric Bishop wrote:
> Hi all,
>
> I want to implement certai
Hi all,
I want to implement certain actions based on SIP response codes. Is there a
similar variable such as ${DIALSTATUS} that comes back with the relevant SIP
response code for a call?
--- Thanks
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"show dialplan" keeps showing contexts created by AEL. I tried deleting
/etc/asterisk/extensions.ael but kept getting these messages in the Asterisk
log:
Feb 14 21:39:53 WARNING[6074] pbx_ael.c: Unable to open
'/etc/asterisk/extensions.ael': No such file or directory
Feb 14 21:39:53 WARNING[6074]
I do need MWI notifcation, just not on this particulary trunk. Is there
anyway to to turn off MWI on a particular trunk or can it only be done
globally?
On 2/22/07, Olle E Johansson <[EMAIL PROTECTED]> wrote:
22 feb 2007 kl. 08.24 skrev Davy Chan:
> **>I have one Asterisk box registering to a
Surely there must be a simpler way than patching the Asterisk code? After
all this is Asterisk-to-Asterisk registration not some third party
softswitch idiosyncrasy. Would setting up fake voicemail boxes help?
On 2/22/07, Davy Chan <[EMAIL PROTECTED]> wrote:
**>I have one Asterisk box register
I have one Asterisk box registering to another via SIP and on the registar
console I keep getting:
-- Got SIP response 603 "Declined (no dialog)" back from xxx.xxx.xxx.xx
Anyone know how to turn off this "feature"?
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Hi all,
We currently have 2 Asterisk boxes and we pass calls to a fro. All works
great except we now need to pass variables between them.
For example now on box 1 we have:
exten => _23XX,1,SetVar(Foo=1234)
exten => _23XX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
When the call dials into Box 2 the
Any kind Polycom dealers out there?
-- Forwarded message --
From: Eric Bishop <[EMAIL PROTECTED]>
Date: Feb 14, 2007 8:10 PM
Subject: Can anyone help me out with Polycom 2.1 firmware please?
To: Asterisk Users Mailing List - Non-Commercial Discussion <
aster
"The conference bridge runs Ulaw codec by default. If you let people connect
with GSM or other codecs, Asterisk will use CPU power to convert audio
between codecs" ... What about alaw channels is there any transcoding work
being done there?
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Hi all,
I am trying to implement native format (ulaw) voice prompts and music on
hold. Different documentation has different file extensions. Does Asterisk
recognise them all? So far I have .ulaw .ul .pcm . Which should I use so
Asterisk recognises them as native uLaw files
_
Would be greatly appreciated
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header
--
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Eric Bishop
*Sent:* Sunday, February 04, 2007 15:43
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] SIP privacy headers
Hi,
Out ITSP has told us to user "SIP privacy heade
Hi,
Out ITSP has told us to user "SIP privacy headers" to hide outbound caller
ID. Does anyone know how or if this can be done in Asterisk. I tried
exten => s,3,SIPAddHeader(privacy=on)
prior to executing Dial but no luck.
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I second that request
On 1/25/07, Kenneth Padgett <[EMAIL PROTECTED]> wrote:
> I ran into this problem with an early batch of IP650s. Polycom's
firmware
> version 2.0.3b made this issue go away.
Speaking of Polycom firmware, anyone have an up to date source for the
stuff? The site I ordered f
On inbound calls from my SIP provider I get multiple warnings as follows:
WARNING[5351]: chan_sip.c:7086 check_via: 'MODH6QD' is not a valid host
Everything else works but these warnings are a pain and I don't know what
they are about Nothing on previos lists or Google explains...
_
I am running a HP DL360 G3 ans want to know the optimal g729 module for it.
There don't seem to be any optimised for Xeon's. I am currently using i686,
but is there a better one to match my Xeon CPU's?
[EMAIL PROTECTED] ~]# cat /proc/cpuinfo
processor : 0
vendor_id : GenuineIntel
cp
Anyone know if it posible to make voice promps native g726 or g711 format?
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On 28 Nov 2006, at 03:01, Eric Bishop wrote:
> I am trying to do it with FOP and Calling Circles. Both have closed
> code. Anyway to do it from Asterisk?
>
You could use the 'Local' channel as the argument to the originate
command
and then set it in the dialplan.
Tim Panton
Do extra CPU's without hyperthreading help?
On 11/28/06, Don <[EMAIL PROTECTED]> wrote:
hyperthreading screws ours up...we actually run better with
hyperthreading off...
hyperthreading results seem to vary from different people you talk too.
- Original Message -----
*From:*
Hi all,
We have Xeon-based system with only 1 (hyperthreaded) CPU (in a HP DL360).
We are seeing high load on multiple meetme session as well as g729
transcoding. My question is will putting an extra CPU help or does Asterisk
just run on a single CPU.
_
th examples:
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate
Cheers
On 11/27/06, Eric Bishop <[EMAIL PROTECTED]> wrote:
> We have calls that originate click-to-dial apps that use the manager
> interface. As most of you know these apps first ring your handset s
time.
Create a call file, and schedule it to run with cron. The following
page on the wiki shows something similar:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
You can adapt it to suit your needs.
- Noah
> on Saturday 11/25/2006 Eric Bishop([EMAIL PROTECTED]) wrote
We have calls that originate click-to-dial apps that use the manager
interface. As most of you know these apps first ring your handset so that
you pickup the handset and then place the outbound call once you have picked
up.
When they first ring my handset (before me picking up the handset) the ca
Not quite what I'm looking for. I ant to hang up all channels (zap or sip)
in meetme room 5
On 11/23/06, Michiel van Baak <[EMAIL PROTECTED]> wrote:
On 19:18, Thu 23 Nov 06, Eric Bishop wrote:
> Other than rebooting the server or restarting Asterisk from cron does
anyone
> k
Other than rebooting the server or restarting Asterisk from cron does anyone
know how to kill a meetme room at midnight. Or perhaps other creative ways
people deal with callers who don't hang up.
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When we have calls that originate click-to-daial apps that use the manager
interface they always originate "from asterisk" is there any way to change
the "from" name?
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Dean,
I know Qtec definately do, however their offering is pretty much focused
only on businesses and they offer their service only via their private IP
network - not via the Internet.
http://www.qtec.com.au
-- Eric
On 11/17/06, Dean Collins <[EMAIL PROTECTED]> wrote:
I have a client who
I second that request.
On 11/4/06, Kevin Bockman <[EMAIL PROTECTED]> wrote:
Hi,Would anyone be kind enough to send me the 2.0.2 SIP firmware? I askedVoipSupply for it on Wednesday, nagged them again on Thursday and theydid not even send the request yet. I was supposed to have it 'Friday
morn
Anyone have a sane rc script for FOP on CentOS/RHEL systems?
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October 03, 2006 4:09 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [asterisk-users] Polycom Buddy Watch Setup help request(Subject changed from 'Re: [asterisk-users] Polycom Buddy Watch Broken with
2.0.1 Firmware?' as it was a bit off topic).>From:
Does anyone have an end-to-end summary of how they have successfully set up the buddy feature including all the relevant Asterisk and Polycom config snippets. All I have been able to do so far is scrounge up bits and peices from the list and Wiki - nothing that covers the entire process... I think
Hi All,When we loose Internet access (DNS) Asterisk basically halts until Internet comes up even for internal registrations and calls. We are even running a caching DNS server on the Asterisk box but this does not seem to help. Any suggestions?
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sterisk? And what are pros and cons of each service for use in conjunction with each. Could I run a PRI protocol over either one since I will cintrol both ends?
On 9/23/06, Jay R. Ashworth <[EMAIL PROTECTED]> wrote:
On Sat, Sep 23, 2006 at 08:22:18AM +1000, Eric Bishop wrote:>We are look
Hi all,We are looking to interconnect 2 Asterisk boxes at seperate sites via a TDM leased line, rather than IP mainly for commercial reasons. Our network provider is offering us either a 31x64kbps leased line or an E1. Am I just ignorant or are these the same thing? An E1 has 30 B channels and 1 D
When ever we do a roll out of Asterisk in a small business environment
replacing an old key system or legacy PBX the receptionist always asks
us, "How do I know if someone is on a call before transferring them?".
My typical answer is "why do you need to know, just do an attended
transfer and if the
Anyone know if it possible to create binary/obfuscated/ human
unreadable extensions.conf/sip.conf etc.? We would like to deploy a
system in an environment where not giving out root is still not enough.
We want to hide the contents of these normally plain text files.
__
Do you need BRI stuff to use the ISDNguard? Also can you make the
switch manually rather than relying on heartbeat auto failover?On 7/11/06, Tzafrir Cohen <[EMAIL PROTECTED]
> wrote:On Tue, Jul 11, 2006 at 08:55:41PM +1000, Eric Bishop wrote:> If so can you comment on how well it has (
If so can you comment on how well it has (or hasn't) worked for you?
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What us meant by "blended rate"?
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Hi All,
We are in search of the latest Polycom firmware SIP 1.6.6 as per http://www.polycom.com/resource_center/1,,pw-492,00.html
Can someone help? We have legitimately obtained these phones but even
our official distributor can't get their hands on updated firmware. The
only thing we have f
I have tried it with 1.2.7.1 and 1.2.9.1. Same issue with both and only on the SIP trunk, not on endpoints.On 6/19/06,
William Piper <[EMAIL PROTECTED]> wrote:
What version of * are you using? I am running 1.2.7.1 with call-limit= and it works fine.
bp
On 6/19/06, Eric Bishop &l
Hi all,
We have a SIP trunk with * and even when there are calls in progress
"sip show inuse" always shows 0 calls in progress. I have outgoinglimit
and incominglimit limit set and have also tried call-limit. "sip show
inuse" works fine with SIP handsets though very frustrating.
___
Hi all,
We have executives who use conference rooms. The typical scenario is
that one of them will organise a conference a few hours in advance and
email everyone the details, however is there anyway the they can
"reserve" a conference room number?
For example if they organise a conference in ro
Hi all,
I have previously (and briefly) use a Sipura 941 before. I could have
sworn that it has a blind transfer soft key when on a call. Now running
the latest firmware (4.1.12a) the only soft keys that come up while on
a call are for attended transfer and 2 way conference.
Can anyone tell me if
Hi All,
I know this can be acheived in the Asterisk dial plan however for
non-technical reasons I need to be able to do it using the SPA-941
Personal Directory feature. An entry such as the following matches the
CALLING number fine but I need the match the CALLED number. In all the
specs of the SP
Could someone explain to a non-US dummy the following phrases I have seen on the list.
"I can provide you with tier 1 termination 6/6. I can blend or NPANXX breakout."
"We provide US48 termination, blended rate for 1 MOU and above is .008 with 6/6."
What is 6/6?
What is US48?
What is blended
I also have an 8700g. Have you managed to figure out how to play .wav voicemails?On 5/13/06, Kerry Garrison <
[EMAIL PROTECTED]> wrote:Our system is running all of the latest code and freepbx and would send the
attachment to my MDA just fine and I was able to play it without anyproblem. My problem
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Is it possible for Asterisk to set the ptime attribute on outbound calls in SDP invite?
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Hi All,
We are in search of the latest Polycom firmware SIP 1.6.5 and BootROM
3.1.3 as per http://www.polycom.com/resource_center/1,,pw-492,00.html
Can someone help? We have legitimately obtained these phones but even
our official distributor can't get their hands on updated firmware. The
only th
Is this with the TE411P? Also what do you mean by "pulled the zaptel trunk source"?On 2/17/06, Stagg Shelton <
[EMAIL PROTECTED]> wrote:This is my last update to my issue. Finally my echo problem is
resolved. On Monday morning 2/13/06 I pulled the the zaptel trunksource. That night after my cust
Hi All,
I have a Digium card in my Asterisk server configured as pri_net and I
want to introduce latency on it in order to simulate PSTN conditions
and test some echo canceller hardware. Is it possible to purposefully
introduce latency and echo in a controlled fashion in order to do so?
Thanks...
Nope only bad feedback here. The software EC in Asterisk worked much better for me than did the VPM on the TE411P.On 2/13/06, Isaac Xiao (KVB Kunlun Pty Limited)
<[EMAIL PROTECTED]> wrote:
What version of Asterisk and Zaptel you were using? Did
you try latest Asterisk 1.2.4 and Zapte
Or perhaps slow them down or pipe to a file?
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Kevin,
I have experienced the same issue. I get worse echo with the VPM
installed than with software EC. Have had it at 2 different sites with
2 different TE411P's.
- EricOn 2/6/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
Stagg Shelton wrote:> I just implemented a system using a TE411P hardwa
Do you have step by step instructions on how you created these RPMs. I
would like to create a few of my own but compiled for my own custom
kernel and patchea and am not very familiar with RPM packagingOn 1/27/06, Andrew McRory <[EMAIL PROTECTED]
> wrote:Available in the usual place.
ftp
will they work with CentOS 4.2?On 1/19/06, Andrew McRory <[EMAIL PROTECTED]> wrote:
I have compiled a set of RPMS from svn and put them in the regular place.Link:ftp://ftp.linuxsys.com/pub/releases/CentOS-4.0/asterisk-1.2.2/
Best Regards,--Andrew McRory - President/CTOLinux Systems Engineer
Would you mind sharing with the list the tellabs hardware and how you got it up and running (ie pinouts etc)?
On 1/15/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
Hello Dan,I was fighting with echo on a number of circumstances, and came to thefollowing conclusions.If you are on a distant loop,
both devices as they are on the
same network. nat=never is a better choice than nat=no. You might also
check your extensions.conf to verify that the calling from 1760 to 7960
is the same as from 7960 to 1760. You could also try moving both
devices to using U-Law instead.
-Jon
Eric Bishop wrote:
Hi
Hi all,
I have a Cisco 1760 gateway and and Cisco 7960 VoIP phone running via
Asterisk. Both are running g711A codecs and SIP. On inbound calls I get
a native bridge, however on outbound calls I never get a native bridge.
With other SIP gateways I do get a native bridge on the outbound call.
My si
most of the echo comes in at about the
28th tap, and assuming a sample rate of 8000hz,
that would be about 3.5ms.
Will that tell you the sort of things you
need to know?
Thanks
james
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of
Eric Bishop
Sent: Sunday, 15
CTED]] On Behalf Of
Eric Bishop
Sent: Sunday, 15 January 2006
10:11
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] echo
tail stats
Does anyone know how to determine the echo tail size (in ms) of a
Does anyone know how to determine the echo tail size (in ms) of a particular call?
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Anyone know how to ignore the 2100 Hz tone detection without editing
zconfig.h and recompiling? I am getting a lot of false "zaptel Disabled
echo canceller because of tone (rx) on channel xx"
The wiki mentiones that this can be disabled at run time. See
http://www.voip-info.org/wiki/view/Asterisk+
Hi all,
We have 1760 working perfectly here with Asterisk for in and outbound calls except for:
1) Outgoing calls sound like they have silence suppression on them
(inbound calls are totally fine though). Have tried "no vad" and and
different VICs.
2) On outgoing calls on the Cisco console I get
Anyone know if Asterisk 1.2.1 supports turning off the hardware echo
canceller WITHOUT recompiling the driver like I had to in 1.0.X?
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Anyone have eny elegant dial plan config for attack dialing? Basically
I just want to automatically and continuously dial a busy until it is
answered and then hand it over to a SIP hanset.
Thanks.
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I am using a Cisco 1760V with FXO card in Australia to provide ports into
Asterisk.
I was wondering if anyone out there has a config for the cisco to detect
the disconnect or hangup signal for Australian tones.
If the calling party hangs up while leaving a voice mail for example, it
takes
Well that didn't work. When I rebooted MySQL didn't start at allOn 11/21/05, JP Carballo <[EMAIL PROTECTED]
> wrote:JP Carballo wrote:> Eric Bishop wrote:>>> I have:
>>>> [EMAIL PROTECTED] ~]# chkconfig --list | grep mysql>>
mysqld 0:off
, Matt Riddell <[EMAIL PROTECTED]> wrote:
Eric Bishop wrote:> Hi All,>> I am running Asterisk (1.0.9.) on CentOS 4 with CDR recording being> output to MySQL. However whenever the system boots up after a reboot I> am needing to manually restart Asterisk because MySQL is after Aste
Hi All,
I am running Asterisk (1.0.9.) on CentOS 4 with CDR recording being
output to MySQL. However whenever the system boots up after a reboot I
am needing to manually restart Asterisk because MySQL is after Asterisk
in the service startup sequence and I get
ERROR[3367]: Failed to connect to m
On 11/18/05, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote:
On Friday 18 November 2005 00:30, Eric Bishop wrote:> I purchased the following item:> http://www.oriontelecom.com/echo_canceller/1u_telnet/e1_1u_19inch_ec.html
>> As you can see not a very highly spec'd product but do
most other scenarios as we are using
Tier 1 hardware (all HP), Digium Rev 2 firmware and our rack is about
10 metres from the CO.
On 11/18/05, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote:
On Thursday 17 November 2005 21:01, Eric Bishop wrote:> I got sick of tweaking and playing with Digium's
Doug Meredith <[EMAIL PROTECTED]> wrote:
Eric Bishop <[EMAIL PROTECTED]> wrote:>If I call our Asterisk box via Disa and then place a call to one of the>problem analogue numbers (native Zap bridge) I don't get any echo. So the
>echo seems to occur only when using a SIP handset a
Yes I am referring to TE411. I have not used TE406 which is the same product, just different slot type.
On 11/15/05, George Pajari <[EMAIL PROTECTED]> wrote:
Eric Bishop wrote:> I have recently seen the light and started using dedicated echo> cancellation hardware. It works great
Hi All,
I have recently seen the light and started using dedicated echo
cancellation hardware. It works great with our E1 PRI's, much better
than either of Digium's hardware or software echo cancellation
products. I have had trouble however finding a simlar device for use
with analogue lines and t
I agree about Asterisk being terrible with DNS failure, but how can you avoid using DNS on *nix system?On 11/7/05, Eric ManxPower Wieling <
[EMAIL PROTECTED]> wrote:Brian Capouch wrote:> I don't think this is a new issue--I've seen it talked about on the list
> before. I don't know if I've ever se
Can any one who has gone from the open source version of Asterisk to ABE comment on their experiences?
Specifically:
- How does the quality compare to the open source stable versions?
- How often do updates come out?
- How far is it behind CVS HEAD in terms of features?
- How good had Digium s
Nope, never really found a satisfactory solution to this..
On 11/3/05, Tom Rymes <[EMAIL PROTECTED]> wrote:
Hi there. I noticed a post you made to asterisk-users backin June regarding problems you were having with Asteriskif your internet connection went down. I am having thesame problem here,
I am looking to buy wither the 1U or desktop E1 echo canceller from
Orion. Has anyone had any experiences either good or bad with these
units?
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Our configuration is as follows:
SIP phones -> TE410P -> PSTN
When a SIP handset makes a call to other ISDN numbers - no problem.
When a SIP handset make a call to analogue numbers - echo.
I know for certain that the problem is at our end. Why?
If I call our Asterisk box via Disa and then plac
I would also be very interested in what hardware you used.
On 10/29/05, Robert Augustyn <[EMAIL PROTECTED]> wrote:
Darren,Can you elaborate on what echocan did you use and how?Thanks.robert> -Original Message-> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED]] On Behalf Of> Darren Wright
I am running CVS HEAD. How can I tell which software echo canceller I am using?On 10/28/05, Matthew Fredrickson <
[EMAIL PROTECTED]> wrote:On Oct 27, 2005, at 12:38 AM,
[EMAIL PROTECTED] wrote:> My question is, what is the direction in relation to analog boards and> such?Right now, it looks like t
Does anyone have a full list of places Asterisk puts all config files
and binaries. I need this to be able to fully rollback if I have a
failed upgrade of Asterisk/Zaptel/LibPRI. So far I have:
/etc/zaptel.conf
/etc/asterisk/
/usr/sbin/safe_asterisk
/usr/sbin/asterisk
/usr/lib/asterisk/modules/
/u
I replaced a TE410P (1st Gen) with a TE411P (2nd gen with hardware echo
canceller) and the echo actually got much worse! Very disappointing!
On 10/28/05, Boris Bakchiev <[EMAIL PROTECTED]> wrote:
Hi,I have TE406P (2nd gen card with echo cancellation on-board).We still notice quite often echo on our
We are running 1.0.9 STABLE on all of our machines. I am about try and
upgrade one machine to CVS HEAD as all this echo cancellation
improvements sound enticing. Can anyone recommend
a) A procedure to cleanly upgrade from STABLE to HEAD
b) A procedure to ensure I can back out and go back to 1.0
On a dual processor Xeon (EM64T) would you reccomend turning
hypertreading on or off? I tend go for it off dual processor machines
just in case 2 processes end up on the one physical processor rather
than 2 processes on 2 different physical processors. What do you think?On 10/9/05, Kevin P. Fleming
Have you founy any real life performance benefit of x86_64 (particularly EM64T on Xeon) as apposed to plan old x86?
On 10/9/05, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
Dinesh Nair wrote:> and they still go for US$10 a pop ?Patent indemnification licenses are completely separate from the codecb
Hi all,
We have a small call centre here running with Asterisk 1.0.9. All the
agents use Cisco 7960's with SIP 7.5 firmware. Is there any way we can
show queue status on the those nice big LCD's. Especially we would like
to display whether the agent is currently logged in or not.
Is this possible
Hi all,
Just a couple of quick questions. I have a HP DL360 G4 (dual Xeon EM64T 3.0Ghz
processors). I am using a TE411P in the
system.
1. Should I run the a x86_64 Linux (CentOS) or just go with the plain
old x86 version? Is there any benefit (or things to be aware of) on
x86_64 vs x86?
2.
Hi all,
Just a couple of quick questions. I have a HP DL360 G4 (dual 3.0Ghz
processors). The processors are EM64T. I am using a TE411P in the
system.
1. Should I run the a x86_64 Linux (CentOS) or just go with the plain old 32 bit version?
2. This being a dual processor system, should I turn on
Can someone from Digium comment on this?
On 9/9/05, Cory Andrews <[EMAIL PROTECTED]> wrote:
I
don't think you can switch the echo tail size, I could be mistaken, but I think
the more channels you are utilizing the smaller amount of MS you have allocated
to each channel.
Cory AndrewsPartne
Hi all,
1. Just bought a new TE411P and about to install it replacing the
existing TE410P. I am assuming I need to set echocancel=no and
echocancelwhenbridged=no now that it will be done in hardware, correct?
2. Is there any way to monitor hardware echo cancellation to ensure it is working (apart
Already have that..
On 8/27/05, Eric Wieling aka ManxPower <[EMAIL PROTECTED]> wrote:
> Eric Bishop wrote:
> > Hi all,
> >
> > Our Asterisk box sends calls outbound via either SIP (through our VoIP
> > provider) or an E1 PRI (directly connected via a TE410P
ChanSel: Reserved
> Ext: 1 Coding: 0 Number Specified Channel Type: 3
> Ext: 1 Channel: 1 ]
> [28 09 41 6c 6c 61 6e 20 44 69 62]
> Display (len= 9) [ Eric Bishop ]
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> Calling Num
nes
> priindication = outofband
>
>
> Hope that Helps
> Jens
>
>
> -Original Message-----
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Bishop
> Sent: 25 August 2005 09:32 AM
> To: Asterisk Users Mailing List - Non-Commercial Discus
Andres,
Thanks for the suggestion. I did try it but it is not moving to the
next priority after the Dial command. I also do know for a fact that
it is not actually being answered. On the console I just get:
-- Called g1/123456789
On 8/26/05, Andres <[EMAIL PROTECTED]> wrote:
> Er
Hi all,
Our Asterisk box sends calls outbound via either SIP (through our VoIP
provider) or an E1 PRI (directly connected via a TE410P). When we dial
a number that is engaged via our VoIP provider we get the following on
the Asterisk console (numbers and IP addresses changed to protect the
innocen
Hi all,
Our Asterisk box sends calls outbound via either SIP (through our VoIP
provider) or an E1 PRI (directly connected via a TE410P). When we dial
a number that is engaged via our VoIP provider we get the following on
the Asterisk console (numbers and IP addresses changed to protect the
innocen
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