Hello,
I want to safely delegate ACD edition to a system administrator who has no
knowledge of Linux nor Asterisk.
More precisely, I want him to be able to edit and change menus such as :
Type 1 for management; 2 for support; 3 for sales department.
I could teach this administrator what Asterisk
On Tue, 21 Aug 2007, Vidura Senadeera wrote:
Dear All,
I would like to get community's feedback with regard to RAID1 ( Software or
Hardware) implementations with asterisk.
This is my setup
Motherboard with SATA RAID1 support
CENT OS 4.4
Asterisk 1.2.19
Libpri/zaptel latest release
2.8
I have it working fine in 1.4.x, but I also have the queues defined in the
Realtime database and not in the queues.conf
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Anthony
Francis
Enviado el: martes, 21 de agosto de 2007 1:46
Para: Asterisk Users Mailing
On Mon, Aug 20, 2007 at 09:23:20PM -0400, Jerry Geis wrote:
I have a TE405/TE410P card that was working on 1.0.X
I upgraded the OS to Centos 4.5, Updated asterisk to 1.4 and zaptel to
1.4.5 and libpri.
I copied all the zaptel and zapata and extensions.conf files from 1.0
I did update
Tzafrir Cohen wrote:
On Mon, Aug 20, 2007 at 09:23:20PM -0400, Jerry Geis wrote:
I have a TE405/TE410P card that was working on 1.0.X
I upgraded the OS to Centos 4.5, Updated asterisk to 1.4 and zaptel to
1.4.5 and libpri.
I copied all the zaptel and zapata and extensions.conf files
You could do more of a hack and find where in the code that the callerID
name and number are found in the voicemail code and use the seldom used
RDNIS variable.
C F's solution is clean and will work across upgrades but I would
probably do the above.
Thanks,
Steve
C F wrote:
While I don't
Matt Florell wrote:
Hello,
A client has asked for Two B channel Transfer capability (known as
TBCT or 2BCT, similar to other features such as ECT, RTL and Q,SIG
Path Replacement) in a new Asterisk system and so I researched the
capability and came up with quite a few gaps in documentation.
I am using TDM2400 with FXO modules to handle 16 concurrent calls to PSTN
for more than 12 hours a day with no problem at all.
On 8/16/07, Chan Jason [EMAIL PROTECTED] wrote:
Hi all,
I am planning to have a new TDM2400P to replace all Planet 450 SIP
gateways. Can TDM2400P survive in heavy
Dear all,
Thanks for the greate explanation regaing Software/H/W Raid. This details
better but on voip-info.org/wiki pages.
Thanks lot agian.
Regs,
Vidura Senadeera.
==
Dear All,
I would like to get community's feedback with regard to RAID1 (
You should have no problems. Make sure you put surge protection and
ground your POTS lines. It is a small investment. I have had SEVERAL
FXO modules die or behave strangely after thunderstorms. I cannot prove
it was a surge, but logic would indicate that was the issue.
Thanks,
Steve Totaro
Russell Bryant wrote:
Steve Murphy wrote:
How about this one: from an extensions.conf that someone posted on the
internet, I think, and I converted to AEL; I'm sorry, but I can't find
the original author.
(If anybody can find his post, I'd love to give him credit.) I did test
this out,
Tzafrir Cohen wrote:
/ On Mon, Aug 20, 2007 at 09:23:20PM -0400, Jerry Geis wrote:
//
// I have a TE405/TE410P card that was working on 1.0.X
//
// I upgraded the OS to Centos 4.5, Updated asterisk to 1.4 and zaptel to
// 1.4.5 and libpri.
//
// I copied all the zaptel and zapata and
Steve Totaro wrote:
You should have no problems. Make sure you put surge protection and
ground your POTS lines. It is a small investment. I have had SEVERAL
FXO modules die or behave strangely after thunderstorms. I cannot prove
it was a surge, but logic would indicate that was the
Dears
Any one succeeded to make Redundancy / Failover with asterisk 1.4.9 on
centos with kernel 2.6.9-55.EL.
Can you please send me the documentation link on how to or write down how to
.
Regards
*
No employee or agent is
C F wrote:
~# cat /proc/mdstat
Personalities : [raid1]
md0 : active raid1 sdb2[2](F) sda2[1]
76139968 blocks [2/1] [_U]
unused devices: none
The above is from an active system that one hdd failed. It would take
way longer to find such a thing on a hardware raid. Unless it came
Hi All,
I have recently installed a TC400B card into a system and am trying to
get it to work. As far as I ca tell from the docco on Digiums website,
there is no config as such unless you want to enable / disable only 1
codec, otherwise by default it runs as 92 channels of either.
I have
Here is a good read as far as what your risk is and how to mitigate it.
They clamp on top of your terminated 66 blocks and you also want to
properly ground.
Here is a vendor with a good selection and pricing. I have no idea if
they are any good, I have never used them.
Please stop posting this repeatedly.
There are pointers on www.voip-info.org
If you post to the biz list, and pay for someone's time and effort, you
may have better luck.
If you keep posting the same annoying message complete with HTML to the
user's list, I seriously doubt anyone will ever
Oooops forgot first link.
http://www.sandman.com/surge.html
Steve Totaro wrote:
Here is a good read as far as what your risk is and how to mitigate it.
They clamp on top of your terminated 66 blocks and you also want to
properly ground.
Here is a vendor with a good selection and pricing.
On Tue, 21 Aug 2007 07:33:23 +0530
Vidura Senadeera [EMAIL PROTECTED] wrote:
Dear All,
I would like to get community's feedback with regard to RAID1
( Software or Hardware) implementations with asterisk.
This is my setup
Motherboard with SATA RAID1 support
CENT OS 4.4
Asterisk 1.2.19
Remco Barendse a écrit :
Has anyone ever tried using a Nokia phone with SIP client as channel for
Asterisk? I mean i would like to receive calls to the mobile on
asterisk and use the Nokia phone to place calls to cell destinations.
E70 and E65 are working perfectly as SIP client through
Thank you all for your post, i've found them quite interesting and will give
work for some time :)
Thanks again.
Cheers,
Jonathan GF
On 8/20/07, Eric Chamberlain [EMAIL PROTECTED] wrote:
Using the phone itself as a GSM-SIP gateway is not possible with the
native VoIP application, but it
Zane C.B. wrote:
On Tue, 21 Aug 2007 07:33:23 +0530
Vidura Senadeera [EMAIL PROTECTED] wrote:
Dear All,
I would like to get community's feedback with regard to RAID1
( Software or Hardware) implementations with asterisk.
This is my setup
Motherboard with SATA RAID1 support
CENT OS 4.4
Any one succeeded to make _Redundancy* / Failover with asterisk
1.4.9 on centos with kernel 2.6.9-55.EL. ***_
Can you please send me the documentation link on how to or write down
how to .
hint
yum -y install heartbeat (on node1 and node2)
configure heartbeat
if you have configuration
Hello,
i would like the forum to help or advice me if my feeling is correct or not.
Is Astlinux the only distribution able to run on Soekris 5501 hardware or
other can run also (trixbox, freepbx, o maybe a manual installation of
asterisk).
My question is easy: i'd need to install it on that
On Mon, 20 Aug 2007, Steve Totaro wrote:
Well chan_bluetooth is really amazing (especially if your phone does not
support SIP).
You connect your phone via bluetooth to your asterisk box and it becomes
a channel type. You can use it as an extension(FXS) or a phone line
(FXO). I believe you
Peder @ NetworkOblivion wrote:
Anthony Francis wrote:
There is no queue_members file, asterisk doesnt know hat you are
talking
about, you would have to #include queue_members from inside that queue
definition.
Huh? How is including a file going to make realtime access the
Gordon Henderson a écrit :
On Mon, 20 Aug 2007, Steve Totaro wrote:
Well chan_bluetooth is really amazing (especially if your phone does not
support SIP).
You connect your phone via bluetooth to your asterisk box and it becomes
a channel type. You can use it as an extension(FXS) or a
Hi all,
We have an 8 agent support desk setup with 2 call queues running
Asterisk 1.4.5. Every so often agents will receive a call from the
queue that only rings once not allowing them time to answer. The call
doesn't seem to be dropped, just seems to go to voicemail. The agents
are also
On 8/20/07, 0xception [EMAIL PROTECTED] wrote:
Okay for a quick report back, that all seems to work...
I am assuming that means that when doing ${VM_CIDNAME:15} you got just
the ticket number.
Thanks for reporting back.
Thanks a lot.
Not much to report back other then that :)...
On
I thought that was what the flashing LEDs on the front of the server's
HDs were for (besides showing activity). Some I have seen also have an
LED near the power button to indicate HD problems.
I guess if you are building your own boxen and not using enterprise
grade servers, this is not the
When I use the CLI (asterisk -r) I get all sorts of info
scrolling past about current activity such as...
-- Executing Macro(SIP/7110-b1d316e0, callrecord|7134) in new stack
-- Executing NoOp(SIP/7110-b1d316e0, Call Record Macro REC7134 ) in
new stack
-- Executing GotoIf(SIP/7110-b1d316e0, 0?4:3)
Olivier wrote:
Hello,
I want to safely delegate ACD edition to a system administrator who has
no knowledge of Linux nor Asterisk.
More precisely, I want him to be able to edit and change menus such as :
Type 1 for management; 2 for support; 3 for sales department.
I could teach this
Tim Groeneveld wrote:
On Tuesday 21 August 2007 12:32:12 am Mark Michelson wrote:
When users call 510 then, it actually does ring everyone who has called
511.
The problem is when the operator comes to pick up the call. The operator
hears nothing, and the user still hears the Music on
On Tue, 21 Aug 2007, Steve Totaro wrote:
I thought that was what the flashing LEDs on the front of the server's
HDs were for (besides showing activity). Some I have seen also have an
LED near the power button to indicate HD problems.
I guess if you are building your own boxen and not using
Administrator TOOTAI wrote:
Gordon Henderson a écrit :
On Mon, 20 Aug 2007, Steve Totaro wrote:
Well chan_bluetooth is really amazing (especially if your phone does not
support SIP).
You connect your phone via bluetooth to your asterisk box and it becomes
a channel type. You
For 1.4: core set verbose 2
For 1.2: set verbose 2
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Andersen
Sent: Tuesday, August 21, 2007 9:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] CLI Question
On Tue, 21 Aug 2007, Bill Andersen wrote:
When I use the CLI (asterisk -r) I get all sorts of info
scrolling past about current activity such as...
-- Executing Macro(SIP/7110-b1d316e0, callrecord|7134) in new stack
-- Executing NoOp(SIP/7110-b1d316e0, Call Record Macro REC7134 ) in
new
Lee Howard wrote:
Artifex Maximus wrote:
zttest is often on 99.975586% with final result:
--- Results after 67 passes ---
Best: 99.987793 -- Worst: 99.951172 -- Average: 99.973764
This is unacceptable for faxing, and it is evidence of the underlying
problem also causing your faxes to
Replying to myself, as I've just discovered AsteriskNow screenshots (in
German !), AsteriskNow seems to offer interesting features for that.
I thought I should let this list readers know that.
___
--Bandwidth and Colocation Provided by
Nick Whitaker wrote:
Hi all,
We have an 8 agent support desk setup with 2 call queues running
Asterisk 1.4.5. Every so often agents will receive a call from the
queue that only rings once not allowing them time to answer. The call
doesn't seem to be dropped, just seems to go to voicemail.
On Tuesday 21 August 2007 10:58:16 Bill Andersen wrote:
When I use the CLI (asterisk -r) I get all sorts of info
scrolling past about current activity such as...
-- Executing Macro(SIP/7110-b1d316e0, callrecord|7134) in new stack
-- Executing NoOp(SIP/7110-b1d316e0, Call Record Macro REC7134
Thomas Kenyon wrote:
The weird thing is, looking at the motherboard manual for my test
machine, The lower the Interrupt does not neccesarily mean the higher
the priority. Eg. 8 to 15 have a higher priority than 3 to 7.
Correct. IRQ 2 bridges to IRQ 8. Thus the priority order is:
0, 1, 2,
Steve Totaro wrote:
I should correct myself, it was called chan_bluetooth but there was an
abandoned project with the same name. Just for clarity, the app you
should be researching is chan_mobile.
Thanks,
Steve Totaro
It was actually never called chan_bluetooth. That was one of the
I have a strange problem with overlap dialing. I installed an asterisk
server between a Siemens HiCom PBX and our telephony provider.
Everything is working fine except some strange problems with the dialing
of the fax (connected to the HiCom PBX). It seems to me that if dialing
takes too long
Matt Florell wrote:
Hello,
A client has asked for Two B channel Transfer capability (known as
TBCT or 2BCT, similar to other features such as ECT, RTL and Q,SIG
Path Replacement) in a new Asterisk system and so I researched the
capability and came up with quite a few gaps in documentation.
Hi,
Are all models with bluetooth capabiilty able to dial using bluetooth ???
In Brazil some telephony companies offer a little box to conect your fixed
land line. Probably a bluetooth to Analog line gateway. However, only
cellphones with especial firmware can be used.
So, what cellphones can I
Lee Howard wrote:
Correct. IRQ 2 bridges to IRQ 8. Thus the priority order is:
0, 1, 2, 8, 9, 10, 11, 12, 13, 14, 15, 3, 4, 5, 6, 7
My zttest results weren't quite as bad as the previous poster.
Home Machine.
--- Results after 113 passes ---
Best: 100.00 -- Worst: 99.987793 --
Hi
on debian iam try to make i get this problem
any suggestions.
make res_config_mysql.so
cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o
res_config_mysql.o res_config_mysql.c
res_config_mysql.c:75: warning: data definition has no type or storage class
On Tue, Aug 21, 2007 at 06:08:03PM +0200, Lars Bensmann wrote:
I have a strange problem with overlap dialing. I installed an asterisk
server between a Siemens HiCom PBX and our telephony provider.
Everything is working fine except some strange problems with the dialing
of the fax (connected
SIP wrote:
Russell Bryant wrote:
Steve Murphy wrote:
How about this one: from an extensions.conf that someone posted on the
internet, I think, and I converted to AEL; I'm sorry, but I can't find
the original author.
(If anybody can find his post, I'd love to give him credit.) I did test
On 23:08, Tue 21 Aug 07, ram wrote:
Hi
on debian iam try to make i get this problem
What version of Debian?
What version of asterisk-addons?
Is this an upgrade?
We need more info
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key:
On Tue, 21 Aug 2007, Russell Bryant wrote:
Nice! While we're on the subject of silly but fun dialplan bits, check out my
TV remote extension. When I moved a few months ago, there was a while when I
couldn't find the wireless keyboard that I usually use as my TV remote to
control MythTV.
Thomas Kenyon wrote:
My zttest results weren't quite as bad as the previous poster.
Home Machine.
--- Results after 113 passes ---
Best: 100.00 -- Worst: 99.987793 -- Average: 99.994452
This should be perfectly fine.
Work Machine.
--- Results after 115 passes ---
Best: 100.00 --
Hello All,
How can I SET EXTENSION from context?
This is my context: -
[docall-usa]
exten = _NXXNXX,1,Answer
exten = _NXXNXX,n,Set() ; What do I need to set here
exten = _NXXNXX,n,DeadAGI(dousacall.php|1)
exten = _NXXNXX,n,Hangup
I need to add 1 in front of ${EXTEN} and then
and the card has its own interrupt -
193:18715321896779 IO-APIC-level tc400b
But when ever we need to do a transcode, ie playing back a wav file on a
g729 channel, the audio is complete rubbish, with a lot of stutters in
it (sounds like a recording does when you upload a file in
Jason Parker wrote:
Steve Totaro wrote:
I should correct myself, it was called chan_bluetooth but there was an
abandoned project with the same name. Just for clarity, the app you
should be researching is chan_mobile.
Thanks,
Steve Totaro
It was actually never called
Hi, Gustavo:
[EMAIL PROTECTED] wrote:
Hi all and thanks for every suggest about my problem, I found that my TDM400P
was sharing IRQ with onboard sound device using cat /proc/interrupts, lspci -v
and lspci -vb. When I disable all unnecessary hardware on my machine and test
it, clicking sounds
On Sat, 2007-08-18 at 21:11 -0700, Ira wrote:
At 08:29 PM 8/18/2007, you wrote:
exten = _1NXXNXX,1,GotoIf($[${EXTEN} = 15554441212]?100)
Where?
I the only variable I am using is ${EXTEN} and as far as I can see I
have a dollar sign on each ${EXTEN}.
I think it's this one.
On Sat, 2007-08-18 at 22:11 -0700, Douglas Warren Garstang wrote:
It looks like when you use odbc for CDR storage, rather than getting a
Dispositon string like ANSWERED, CONGESTION etc, you'll get an integer
(1,2,4,8). Does anyone know where I can find what strings (ANSWERED etc)
these
On 8/21/07, Michiel van Baak [EMAIL PROTECTED] wrote:
On 23:08, Tue 21 Aug 07, ram wrote:
Hi
on debian iam try to make i get this problem
What version of Debian?
What version of asterisk-addons?
Is this an upgrade?
We need more info
Hi
no its fresh installation.
Steve Edwards wrote:
Almost every room in my house has a phone -- if I could teach my kids to
put them back where they belong.
This could easily be extended to recognize which phone was used so it
could control the Myth FE in that room.
Also, it could/should be extended to control x10
Can someone share pointers to Asterisk's Dialogic support? Which boards
are supported, driver status, and etc.
Thnx
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options
Quoting Steve Prior [EMAIL PROTECTED]:
shutting off the dialtone should be pretty simple, then what is really
needed is an audio Bidirectional Tee almost like a 3 way call, well
I guess exactly like a 3 way call but not dialed.
you have the dsp that is going to process audio on the channel,
Nitesh Divecha wrote:
How can I SET EXTENSION from context?
This is my context: -
[docall-usa]
exten = _NXXNXX,1,Answer
exten = _NXXNXX,n,Set() ; What do I need to set here
exten = _NXXNXX,n,DeadAGI(dousacall.php|1)
exten = _NXXNXX,n,Hangup
I need to add 1 in front
Greetings,
I have a problem getting Asterisk registered as a UAC against the
MetaSwitch call agent, because the customer insists on running it on a
NAT'd box. Thus, the Contact: field in the REGISTER request betrays
the private IP address of the Asterisk box, but the source IP of the
message
On 8/21/07, Steve Prior [EMAIL PROTECTED] wrote:
Steve Edwards wrote:
Almost every room in my house has a phone -- if I could teach my kids to
put them back where they belong.
This could easily be extended to recognize which phone was used so it
could control the Myth FE in that room.
Steve Edwards wrote:
Almost every room in my house has a phone -- if I could teach my kids to
put them back where they belong.
This could easily be extended to recognize which phone was used so it
could control the Myth FE in that room.
Also, it could/should be extended to control x10
On 8/21/07, Steve Edwards [EMAIL PROTECTED] wrote:
To control the tv in this room, press 1. To control a tv in another room,
press 2. To control the outside lights, press 3. To control the
sprinklers, press 4, ...
To control the power bar the Asterisk server is plugged into, press 5
click
On Tue, 21 Aug 2007, David Gomillion wrote:
Now, you can address Asterisk by saying, Computer, raise lights 20% and
impress all of your trekkie friends when the lights turn up.
Sorry - it's gotta be: [1]
Zen, lights up.
boing Confirm.
But I guess not many leftpondians might appreciate
My servers run in a datacenter, 50km away from my office... if a led
flash, if the speaker beep... I think I'll not see/hear it ...
My servers are monitored using nagios which has a plugin for software
raid... so if one array goes down, I receive a mail/sms/call/...
futher more, everything is on
I am pretty sure you can only get Dialogic support in ABE.
Thanks,
Steve Totaro
Wai Wu wrote:
Can someone share pointers to Asterisk's Dialogic support? Which boards
are supported, driver status, and etc.
Thnx
___
--Bandwidth and
Hi:
I've got a dozen Mitel 5020 IP sets and can't find out if they do SIP,
or even find an administrator's manual for them. Mitel has been rather
unhelpful. They only deal with partner resellers.
Has anybody used these with Asterisk?
-Stephen-
___
Gordon Henderson wrote:
Either start asterisk with no -v's or type:
set verbose 0
at the prompt.
Thanks. Exactly what I needed.
Bill
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To
Gordon Henderson wrote:
On Tue, 21 Aug 2007, David Gomillion wrote:
Now, you can address Asterisk by saying, Computer, raise lights 20% and
impress all of your trekkie friends when the lights turn up.
Sorry - it's gotta be: [1]
Zen, lights up.
boing Confirm.
But I guess
On Tue, 21 Aug 2007, Wai Wu wrote:
Can someone share pointers to Asterisk's Dialogic support? Which boards
are supported, driver status, and etc.
Which type of boards are you interested in? I don't know about other cards,
but the DIVA Server ISDN cards are well supported.
Armin
I guess I am just lucky to have 24 hour manned data centers with staff
that walk around looking for flashing LEDs.
I am sure there is some error thrown in /var/log/messages about a
failure that could be used to trigger a notification quite trivially.
Thanks,
Steve
Arnaud Ligot wrote:
My
On 00:18, Wed 22 Aug 07, ram wrote:
On 8/21/07, Michiel van Baak [EMAIL PROTECTED] wrote:
On 23:08, Tue 21 Aug 07, ram wrote:
Hi
on debian iam try to make i get this problem
What version of Debian?
What version of asterisk-addons?
Is this an upgrade?
We need more info
Steve Prior wrote:
What I was thinking about was what if instead of a dialtone you are
brought directly to a home automation voice menu which works in
parallel with your normal dial plan. If you wanted to make a call,
just ignore the voice menu and dial normally. If you wanted to
turn on
Got this figured out. externip= does work if the other NAT-related
options are also enabled, plus it appears that Trixbox (which is what
the end-user was using, it seems) handles this well in its config file
structure regardless.
--
Alex Balashov
Evariste Systems
Web:
The Asterisk development team has released version 1.4.11.
This version contains numerous bug fixes. One of these is for a security issue
in chan_sip. The issue is that SIP dialog history was being stored in memory
regardless if the option for this was turned on or off. This could be abused to
On 8/22/07, Michiel van Baak [EMAIL PROTECTED] wrote:
On 00:18, Wed 22 Aug 07, ram wrote:
On 8/21/07, Michiel van Baak [EMAIL PROTECTED] wrote:
On 23:08, Tue 21 Aug 07, ram wrote:
Hi
on debian iam try to make i get this problem
What version of Debian?
What version of
Asterisk Project Security Advisory - AST-2007-020
++
| Product | Asterisk |
Hello PPL, someone have any idea for notifying users that they have
voicemail waiting when they will register after weren't being registered on
asterisk? I need this for nokia terminal e series users. I studied sms
service but seems to be only for PSTN lines. I comes with idea to receive a
call
Hi guys,
I've made some tests with a partner and when he call to me he can't
hear ring back tone.
My asterisk sent 180 ringing message to him.
He told me that in 180 ringing there isn't a media attributes and i
need to reconfigure my side to send 180 ringing with media attributes.
How can i
The Asterisk Development Team wrote:
The Asterisk development team has released version 1.4.11.
This release is available for download from
http://downloads.digium.com/pub/telephony/asterisk/.
Not quite. :)
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied -
I have this exact same problem with two different Business Edition
systems. Both are using TDM400s.
Do we have an answer for this yet?
Yours,
Michael Munger, dCAP
404-438-2128
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Michael Munger wrote:
I have this exact same problem with two different Business Edition
systems. Both are using TDM400s.
Do we have an answer for this yet?
I know this sounds silly, but if there is a chance that it is an
improperly tuned echo canceller, has anyone tried using oslec.
On Tue, Aug 21, 2007 at 08:42:50PM +0300, Tzafrir Cohen wrote:
Not sure what the problem is, but a way around it:
Any chance you could disable the overlap dialing and get the PBX to send
the whole number in one go?
Mmmh. The PBX is not very friendly to program. But I will have a look.
My customer has tones of DM3 cards (DM/V600, DM/N1200, and D600-2E1),
they want to see if they can use them in Asterisk. My advise to them is
to sell those cards and buy Sangoma E1 cards, and still have money left.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Michael Munger wrote:
I have this exact same problem with two different Business Edition
systems. Both are using TDM400s.
Do we have an answer for this yet?
You need to contact Digium technical support. They provide free support for
hardware issues like this. Furthermore,
since you are a
Andres wrote:
Try to compare the frame size you are receiving from asterisk and set
your phone to transmit the same frame size. I would guess the card
appears to have problems when the frame size is different. Please try
and report back. I am curious about this.
The problem occurs
I’ve been tearing my hair out trying to get a Polycom phone (behind a NAT) to
register to an * box behind a Cisco SIP ALG. With known good credentials
configured on the phone and in *, I get 403 Bad Auth when trying to register.
If I put the phone onto the same LAN as * it works fine without
Hi All,
I have a Polycom 501 that is behind a NAT. When it registers to the
Asterisk server it is using the IP address on the private network and
not the public IP of the NAT address.
Can someone tell me what I need to do so the phone registerers using an
internet address rather than the
On 8/21/07, Steve Edwards [EMAIL PROTECTED] wrote:
To control the tv in this room, press 1. To control a tv in another
room, press 2. To control the outside lights, press 3. To control the
sprinklers, press 4, ...
Before this thread I already had a Firecracker on the server, a
Here you go folks:
ftp://ftp.ntcp.net/DUNDi_So_Easy.pdf
If someone would be so kind as to upload to the wiki, it will be much
appriciated.
Thank you all who replied to my poll questions.
As always, I hope this help.
JR
--
JR Richardson
Engineering for the Masses
Polycom's were simply not originally built for multi location VoIP. There
is no NAT support in the Polycom's. We have several networks, being an ISP,
and have found that when transversing one network say 192.168.2.x with the *
box on a 192.168.1.x the polycoms were able to communicate however
On Tue, 21 Aug 2007, Matthew Warren wrote:
We have several networks, being an ISP, and have found that when
transversing one network say 192.168.2.x with the * box on a 192.168.1.x
the polycoms were able to communicate however sustained a lot of one way
audio problems. Moving thim onto
In your sip.conf, for the user:
nat=yes
To send keepalives for the UDP connection (depending on how flimsy the
device handling NAT is):
qualify=yes
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Klaverstyn, David C
Sent: Tuesday, August 21, 2007
I think what Alex was trying to say was that Polycom IP Phones are an
example of immature product development. While they look very nice and
have a nice display the product doesn't compete very well compared to
other manufacturers.
The two most obvious flaws are that they cannot be NAT'ed so they
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