Hi -
Just wondering if anyone has gotten a Polycom VVX phone to
successfully do an Auto Answer with asterisk. I have an older
generation of Polycom phones that do this just fine, but I can't seem
to make the VVX phones work.
I tried the guide here:
On Fri, Mar 14, 2014 at 12:36 PM, Noah Miller noahisaacmil...@gmail.com wrote:
Hi -
Just wondering if anyone has gotten a Polycom VVX phone to
successfully do an Auto Answer with asterisk. I have an older
generation of Polycom phones that do this just fine, but I can't seem
to make the VVX
Hi All -
I pulled from a working system a TDM400 with one s110 fxs and three
x100 fxos. I put it into a new box and the fxs no longer works. The
fxos work just fine. I thought it was odd, but I chalked it up to a
random chance failure and ordered another s110. The replacement
doesn't work
.
I'm just thinking that the failure that dahdi_scan see may be because
the s110 isn't getting power.
On Tue, Nov 16, 2010 at 1:46 PM, Barry Miller
asterisk-us...@notanet.net wrote:
On Tue, Nov 16, 2010 at 01:17:08PM -0500, Noah Miller wrote:
Hi All -
I pulled from a working system a TDM400
I'm just thinking that the failure that dahdi_scan see may be because
the s110 isn't getting power.
If you see FAILED in dahdi_scan for the FXS port, then most likely
there will be some indication of what actually failed in the kernel log.
Is there anything in dmesg?
Aha! Thanks, Shaun.
Hi Garge -
exten =
,1,Asterisk_Application(Action) ;Dial(Zap/1/${Phone_Number_you want})
Two things:
1. There is no such thing as Zap anymore. Zap has been renamed to
Dahdi because of a trademark issue. So your extension should look
like:
exten = ,Dial(Dahdi/1/)
2. Do you
Ok..So what ip phone model do NAT?
I think you'd struggle to find one. If it's a requirement you're probably
doing something wrong...
Definitely get a router. Plug the IP phone into the router, and then
you can plug the computer into the phone or the router.
- Noah
--
It is a building, with 24 separated rooms, each room will have a PC and a IP
Phone. Every room connected to a switch Cisco 2950.
I want keeping all PCs isolated behind a NAT (no access to neighbour's PC),
and still keep communication in same LAN between all IP Phones.
Should I take another
I have a question about the blind transfer using ##. This works great on our
cordless phone, but there have been occasions that we can't transfer using
##. I was able to reproduce the issue by doing the following:
1) Call in from the outside line,
2) Ask the operator to transfer me to an
I think you need to remove the line echocanceller in system.conf
You could also try to use fxotune, it'a really improving things.
You also need to put echocancel=yes in chan_dahdi.conf
This is a PRI, so fxotune is not the thing to use in this case.
- Noah
What are the limits with asterisk server running on one decent (4GB, 4 CPU
etc.) machine.
There are a LOT of factors involved. You will likely have to do your
own testing with just the specific features you want.
How many MeetMe conferences it can support? What is the limit of number of
The echo between our extensions (using Polycom 550 handsets) disappears
once I removed the Digium echo module.
Are you routing internal calls from SIP - DAHDI - SIP? The digium
echo module will not have any effect on pure SIP - SIP calls. Do
you have acoustic echo cancellation active on the
I'm actually there, but I was wondering if the tables there are up to
date and if any changes took place. I see all kinds of comments about
changes.
You could go ahead and install and then look at the table structure
using your dbms.
- Noah
___
--
I assume if all the SIP trunks are to the same host/port, Asterisk
cannot distinguish which trunk is active when an incoming call is
made- it will dump all incoming calls to the context specified in the
last trunk entry of sip.conf
No. SIP uses authentication (well, I guess you can not use
Hi -
I am having echo issues on our Asterisk box using a PRI circuit. I was
using the software echo cancellation and that helped a bit but didn't solve
it completely. So I went and bought a Digium echo cancellation module for
the TE121 card. That made it even worst, getting more echo on
In 2007, I released a Polycom Provisioning Tool. I retired the package
earlier this year, and have had so many requests for it, I have revived the
concept, new, improved, and still FREE.
Any chance of you releasing the source?
The asterisk GUI does Polycom phone provisioning, and that source
I’m running CENTOS 5.3 with apache 2, asterisk 1.4.26.2, mysql 5 and php
5.2.11. top shows 928mb out of 1035mb in use with idle asterisk and 17
users. There could be a problem, but I’m relatively new to CENTOS, so any
suggestions would be happy.
I use CentOS for asterisk boxen, too, and my
Hi Warren -
I have one client that is telling me that their Polycom 500's format the
file system every time they reboot, and also that they are unable to make
changes locally on the phone itself, only via the config files. If the
config file is not available when they try to boot the phone,
Hi Blaz -
Do you maybe know for a fairly good quality IAX2/SIP hard phones in up to 40
USD?
I don't think there are any IAX hardphone in production anymore. You
might be able to find a used Atcom 320, but probably not for anywhere
close to $40.
It looks like voipsupply.com has some old Cisco
Hi Mike -
I've got a Polycom 501 that's been working with Asterisk for some time.
However, I don't seem to be able to put a call on hold and get it back. It
goes on hold just fine. But when I press the resume button, nothing
happends.
Anyone seen this befor? Any ideas on where to start
We have swapped out the phone multiple times for the user.
Only one user.
Bad PoE port on the switch?
How about local interference that the user cannot control? Does the
same phone experience static when moved elsewhere?
Do you have a power brick for the phone so you can try it as non-PoE?
We swapped PoE switches, phones, cable and switch ports multiple times.
What do you mean by local interference? Cell phone? The person swears
nothing is near the phone.
There are lots of things that can cause interference. Radios,
elevators, bad electrical wiring, you name it. Is the static
So, does anyone know of a way to detect whether a call from a SIP phone
is the first step of an attended transfer or an original call?
It could probably work if you put a SIP proxy in between (ref. Kamilio).
Another way might be to set up a special transfer extension that all
users use to
If I get an echo cancellation module for my Digium TE121 card, will I need
to do any adjustments/configuration in Asterisk?
You should probably still set the gain using rxgain and txgain. IME,
it's much easier setting gains on a PRI than it is on a POTS line,
though. I've worked with a couple
I have two Asterisk server, running on Asterisk 1.6:
SRV1 = 192.168.0.5 on Asterisk 1.6.1.4
SRV2 = 192.168.0.20 on Asterisk 1.6.1.8
I want create a link for exchange call.
To clarify and expand on Aggio's response. You either need to have a
peer and user on both machines, or you
I use two ‘lines’ though ‘Line appearances’ would be a better term, though
still confusing in my book.
I have five line appearances on the Snom190 on my desk. I regularly
use two line appearances, and on occasion, I have used three to juggle
back and forth between calls.
I would guess that a
When typing 'help' on the command line (* console) is there a way to
keep it from just scrolling most of the information off the top of the
screen? I can't hit ctrl-s fast enough so I miss most of the info. This
makes 'help' be not much help.
my default scroll back buffer is set to around
I don't know much about game consoles, and I was wondering if someone
had successfully ported Linux and Asterisk to the current hardware,
ie. Nintendo Wii, Sony PS3, or Microsoft XBox360?
The Xbox is an x86 machine, so running linux and/or asterisk on it
should not be too difficult. There's
We're also working fine with it but I also do not know what the
available imapflags are and what they mean. I have seen notls and
novalidatecert. Out of curiosity, I spent the last 20 minutes googling
for information on c-client imapflags and didn't find any definitions or
even a simple
Hi All -
At Leif's suggestion, I'm soliciting testers for a patch to IMAP voicemail.
Currently, when asterisk checks for voicemails in an IMAP folder, it
only looks for messages in the same context and with the same
voicemail box number as the person dialing in to VoicemailMain(). I
believe
Hi All -
I'm setting up a corporate emergency broadcast system that uses an
autodialer to contact all company employees. Everything works fine
except if the auto-dialed calls go to the end users' voicemail. If
that happens, asterisk starts playback of the emergency message while
the voicemail
Hi -
I had a client recently move their asterisk system (asterisk 1.4.26,
dahdi 2.2.0.1, aex800 w/vpm module) to a new location, a building
that's nearly 150 years old. I was not personally able to go there,
but the person who did the move said the building's demarc room was
scary-- water leaks,
My question for anyone with knowledge on this: would HPEC do a better
job than the VPM module (or oslec)? Can HPEC cope with very long echo
tails?
HPEC and the Digium VPMADT032 use the same algorithms from the same vendor.
Aha. Thanks for this tidbit, Kevin!
Next question: does anybody
Next question: does anybody know how to handle extremely long tail
echo that a VPM module cannot?
How long is 'long' in this case? The VPMs and HPEC (and OSLEC) can
handle 128ms echo tails, which is pretty darn long. It's rare to see an
echo tail longer than that except on very high latency
Please, does anybody have a good document describes well
the optimum method to achieve Asterisk Redundancy/Clustering on 2 servers.
Documentation?!... well... there's not much.
It depends on what you're trying to achieve with your cluster. If you
want a simple active/passive failover
[14177.069426] dahdi: Version: 2.2.0-rc2
Are you sure you're using the latest stable release of Dahdi and not the rc?
- Noah
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Let's just simplify this a LOT:
Your phones have no dialtone. This means they are not registering
with asterisk. I see in your sip.conf, for both you phones, you have:
host=X.X.X.X
If you specify an address here, your phones will not register.
Instead, to make your phones register, set it to:
Hi -
I just deployed a system using IMAP Voicemail. During my testing,
voicemail worked fine. I could check vm from the phone, and the
messages would get marked as read, or I could read the messages in a
mail client, and the phone's mwi light would turn off. Very neat.
I'm not exactly sure
I'm not exactly sure when things got munged up, but something broke.
I can record messages with Voicemail(), but now when I access an IMAP
mailbox using VoicemailMain(), it always says there are no messages,
even when there clearly are (unread) messages in the IMAP mailbox.
This appears to
).
Martin
On Thu, Apr 2, 2009 at 4:38 PM, Noah Miller noahisaacmil...@gmail.com wrote:
Hi -
Does anybody know if an FXS generates line voltage when Dahdi/Zaptel
is disabled?
Thanks,
Noah
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Hi -
Does anybody know if an FXS generates line voltage when Dahdi/Zaptel
is disabled?
Thanks,
Noah
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The policy that we have been following is that only final releases will be
announced to the asterisk-announce list. Betas and release candidates are not.
The rationale is that asterisk-announce is supposed to be a low-volume list
and
that most subscribers to it would not appreciate all the
Hi Marcelo -
Is there any alternative to invoke mid-call services without using the # and
* signals? I was expecting to use Hook-Flash either via INFO or RTP
telephone-event.
You can change the keys used to invoke a service in features.conf. I
know many people use ## or #1 for blind transfer
Hi Steve -
New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
existent.
Welcome to Open Source!
Seriously, look at the README files accompanying asterisk, dahdi, and
libpri. They will give you compilation/installation instructions.
You can also search this list with
It seems to me that everything one may want to know would be contained
on voip-info.org
Hmm. Dangerous statement. There are many things on the WIKI that are
quite outdated, and a great many other things that aren't there at
all.
People don't ask stupid questions because of a lack of a FAQ
I don't believe that Polycom's version of SLA does anything with
Asterisk. You have to use asterisk's SLA implementation
(http://www.asterisk.org/node/48342).
So asterisk can't do SLA with Polycom phones?
Asterisk can do SLA with Polycom, just not using Polycom's SLA
implementation (in
Hi Mark -
Has anyone done SLA with Polycom phones? I've got a large project coming
up where the customer is keen on SLA for trunks and extensions. Trunks
will be on a PRI.
We may do this with Cisco phones if they work better.
You really want to do SLA with all 23 lines of the PRI? That's a
Hi Mark -
You really want to do SLA with all 23 lines of the PRI? That's a
lotta lines to be shared. You'd need two sidecars for each phone
(Cisco or Polycom).
Actually there will be multiple PRI's :)
This customer is a multi-tenant situation so each tenant will have a few
trunk SLA's
Hi Abel -
I had installed Asterisk 1.4 and when I call to a exist extension, the
voice have noise, but, when I call to a extension does no exist,
asterisk played a voice that say me that extension does no exist, but
without noise
I want I some body can test with a softphone my server,
Hi Tzafrir -
I'm wondering if anybody has IMAP Voicemail AND the directory working
together. I haven't had any success. IMAP voicemail works fine, but
when it's active, the Directory does not work. The problem seems to
be with libc-client. Specifically, asterisk is not able to access the
Hi All -
I'm wondering if anybody has IMAP Voicemail AND the directory working
together. I haven't had any success. IMAP voicemail works fine, but
when it's active, the Directory does not work. The problem seems to
be with libc-client. Specifically, asterisk is not able to access the
mm_dlog
Hi Sean -
On Wed, Dec 3, 2008 at 7:36 PM, sean darcy seandar...@gmail.com wrote:
Installing 1.4.23-rc2, I actually looked at the startup and saw this
warning:
WARNING[10730]: loader.c:359 load_dynamic_module: Error loading module
'app_directory.so': /usr/lib/libc-client.so.2007: undefined
I have IMAP voicemail working with Exchange 2003 using a single username and
password for multiple mailboxes.
Sorry to hijack this thread (at least I changed the Subject), but this
really caught my eye. I was under the impression that Exchange's IMAP
doesn't have the master user feature and
Hi Jeff -
I have IMAP voicemail working with Exchange 2003 using a single username and
password for multiple mailboxes.
Sorry to hijack this thread (at least I changed the Subject), but this
really caught my eye. I was under the impression that Exchange's IMAP
doesn't have the master user
Hi Ken -
Hey, all. When I last was heavily into Asterisk (1.0.x), setting up to
receive faxes was, well, a PITA, what with having to patch the Asterisk
install with various driver patches and this, that, and the other.
Is that still true? Is there a fax HOWTO out there that reflects
Hi Dan -
I found the maxusers defined in meetme.c, but I'm
not sure how this value is set. Does anybody know
if one can limit the number of users permitted in a
meetme conference? I know there's MeetmeCount(), but
I'd rather avoid the dialplan logic and just set
maxusers instead.
That
Due diligence is required on anything 10,000 people are going to be
pounding on. Undersizing is common,
I think due diligence is THE key with any open source solution,
including asterisk. I'll admit that I pretty badly screwed up one
asterisk installation because I didn't adequately prepare it
Is Asterisk even needed?
Potentially, no. But if you intend to provide subscriber/PBX features,
it is needed as a UA feature box(s).
And FreeSWITCH can't handle that?
Freeswitch can provide many PBX features with additional modules, but
asterisk can provide more, and its implementations
Hi -
I found the maxusers defined in meetme.c, but I'm not sure how this
value is set. Does anybody know if one can limit the number of users
permitted in a meetme conference? I know there's MeetmeCount(), but
I'd rather avoid the dialplan logic and just set maxusers instead.
Thanks,
Noah
Hi Shariq -
I m facing problem with TDM2400P pstn card. When someone dials, the voice
quality is crappyInstead of hearing.
Echo cancel almost works, but the callee hear what they describe as a
'background crackle/buzz' coming back when they talk.
Crackling noise is usually caused by an
Hi Alejandro -
Dear all, now I'm using an Asterisk 1.4.13 SIP server with 50 SIP users
and it works very well only in an intranet environment (no connections
to the PSTN world).
But in the near future, we have to plan a telephone system that works in
the intranet (voip) and also it must be
Hi Andrea -
I have two asterisk servers, an IAX trunk between and some SIP users
registered
to each server.
The scenario is this: user A, registered to PBX 1, calls user B, registered to
PBX 2. Then A wants to transfer the call using the features.conf method (in my
case, **), but is
Asterisks greatest strength is that it's a highly flexible platform that
let's you pretty much do anything.
It's downside, is that it's a highly flexible platform that let's you
pretty much do anything.
In other words, the quality of what you are trying to do depends on the
quality and
Hi Larry -
This is my setup of the features.conf but it had not any reaction after I
pushed the *2 while calling was acting ! Could you tell me the reason ? Or
give my the method of the setting.
Thanks!
LARRY
[general]
parkext =
Hi Ken -
The SIP.CONF has been made identical across all 3 remote locations, and the
main server has the same config for each remote site connecting.
I first want to confirm that it's possible to have 3 remote Asterisk servers
setup as a SIP client connected to a 4th Asterisk server.
I just
Hi Nhadie -
Could it be my problem is since i'm using 2 asterisk, if an extensions
registers on asterisk#1 it will not be reachable by extensions on
asterisk#2? or it should not matter if i'm using realtime?
It does not matter that you're using realtime. If a phone registers
to asterisk
Hi Daniel -
How can I made a 3-way conference betwwen IAX channels?
My current version is: 1.4.21.1
Anytime you need a call with more than 2 parties, you need to use some
kind of conferencing application. The default conference
application for asterisk is meetme. You can use meetme with any
with IAX softphones is somewhat limited, but maybe if
you indicate which phone you're using, somebody could provide you with
assistance.
- Noah
Daniel
On Tue, Jul 22, 2008 at 8:18 PM, Noah Miller [EMAIL PROTECTED]
wrote:
Hi Daniel -
How can I made a 3-way conference betwwen IAX channels
Hi James -
Thanks for the wild guess. But The user(who is myself) is dialing 3000. It
only failes to work when I use patterns. So I thought I am making a mistake
on the syntax, I have checked all the books I have and the internet and I
can't see anything wrong. :-\
Sounds like time for some
Hi Joseph -
I have Astra 480i's and Snom M3's. I am using a SIP provider so I do
not have any peripheral cards.
I am on voip-wiki now reading about the echo canceller tuning, thanks!
For your particular case, you're probably not going to find much
useful info on the wiki about echo
This is almost standard with voip calls. The echo-cancellation has to
train up to the call parameters. Some hardware is better with it than
others and you can try tweaking the value for the echo canceler up and
down.
Hmm. This has not been my experience. I have rarely seen echo on
pure
Hi John -
I have a request that I have not been able to figure out as yet. I need
to be able to play a beep when a call is transfered via attended transfer.
This is exactly what is in the bug tracker at:
http://bugs.digium.com/view.php?id=3819
Has any one found a way, elegant ot otherwise,
Hi Loic -
According to that its using MG2.
I think it will say MG2 regardless of whether or not there is a
hardware module present.
Shouldnt it be using something like
HPEC?
I don't think the hardware echo cancellers use the HPEC algorithm. As
Eric and Matt have mentioned, dmseg will tell
Hi John -
That could be...I only have ports 5060 and 8088 open on the firewall.
Should another port be open?
If asterisk is inside a firewall/nat and the phone devices are on the
other side, you need to also open port for the rtp audio stream. By
default, this is UDP 1 - 2, but this
One thing I have noticed is that in the cases where the wildcard cannot
determine the CID (i.e. because the rxgain is up around 10.5), I get
this in my asterisk console:
[Jul 15 08:04:09] NOTICE[26696]: chan_zap.c:6670 ss_thread: Got event 18
(Ring Begin)...
It is odd that it would
Hi Adrian -
When I use re-invite, does the Asterisk server stay in the SIP conversation,
and just RTP traffic diverts, or does the SIP transfer away from the A*k
server too ?
I'm sure somebody will correct me if this is wrong, but I believe the
signalling must stay with asterisk, as asterisk
Hi Uros -
I have problem using Asterisk.I have isdn-pri and openvox d110p card in my
computer.They are connected with RJ-45 (1,2,4,5 pins to the card and all
pins to the isdn done by telco workers). I got green led on isdn which is
sign that isdn is working and that is connected to openvox,
Hi -
I'm trying to install a fresh copy of asterisk on a 64bit platform. I'm
using CentOs 5.1 and all the latest builds of zaptel, libpri and asterisk.
When I try to build Asterisk this is the error I'm getting.
src/add.c:1: error: CPU you selected does not support x86-64 instruction set
Hi John -
I have a fresh copy of asterisk, libpri, zaptel, asterisk-addons, and
asterisk-gui installed on centos (I built everything using ./configure,
make, make install, make samples). I connected to the GUI interface and
created two new users. I used the two users accounts to connect up
Hi Leotis -
When i run fxotune -i i get the following output:
sudo fxotune -i
Tuning module /dev/zap/1
Done!
/dev/zap/2 absent: No such device or address
/dev/zap/3 absent: No such device or address
/dev/zap/4 absent: No such device or address
/dev/zap/5 absent: No such file or directory
Hi Bilal -
When Asterisk behind NAT and Polycom behind NAT, I forwarded the 5060 UDP to
asterisk
(at asterisk router) and to Polycom IP Phone at polycomg router site, but the
problem stayed.
Also I was use nat=yes in the sip.conf
Also I forwarded the udp rtp ports (that configured in
with the DONT_OPTIMIZE flag.
I'd probably pick option 1, but it may just be easier to use option 2
depending on what gcc packages are available for your system.
- Noah
On Mon, Jul 14, 2008 at 11:40 AM, Noah Miller [EMAIL PROTECTED]
wrote:
Hi Leotis -
When i run fxotune -i i get the following output:
sudo
Hi Syed -
I have been using single TDM800P card. It is a small card with 4FXO and 4FXS
ports. I have been using it for sometime without any problem. I am using
Asterisk 1.4.18.1. Now due to greater requirement to handle more calls our
office has bought another larger card TDM2401E which has
Hi Enrico -
I'm trying to get up and running a TDM400 with a standard italian pots
line but i'm having
problems at getting asterisk to detect when the line get answered on
outgoing calls.
I'm using asterisk 1.6 beta 9 with zaptel 1.4.11.
Zaptel channels use fxs_ks signalling .
I must
Hi -
I can not seem to get AsteriskNow to register my SIP provider correctly?
I can do this manually when compiling Asterisk and installing it w/o a
GUI, but not with this. I just get the following message.
-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #22)
The
Hi Syed -
zttool shows that TDM800P is loaded first and TDM2401E is loaded second. now
problem is
ports are not being configured by asterisk. i have done following changes in
two files
zaptel.onf and zapata.conf.
zaptel.conf
loadzone=us, defaultzone=us,
fxoks=1-4, fxsks=5-8,
Hi Jose -
After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri,
zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop
working.
The board is working, I tested in another server with the 1.2.13 asterisk
version.
Also changed the pci slot where the board
Hi Bob -
I have a problem compiling Zaptel on an up to date CentOS 5.2 box.
Zaptel 1.4.11, CentOS running on AMD dual core X64.
...
CC [M] /projects/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.o
In file included
from /projects/asterisk/zaptel-1.4.11/kernel/xpp/xpd.h:26,
from
I have 1 nic card which is linked to the router.
Then I use 1 port on the router which is linked to the asterisk appliance.
It will work via WAN which ive now got. SO I can access the asterisk
appliance via 192.168.1.15
The problem is now…How do I connect the phone.
Ive got the phone
Hi Doug -
In my research it appears this often happens when using more than one
processor. I am using a dual core Pentium.
I guess my dilema here is which way to go. Clearly the audio is not
working the way I would like it to and the way I came to expect from my
old system. When playing
Hi Matt -
In short, fxotune adjusts line impedance, where as adjusting gains I believe
is essentially adjusting the amplification / deamplification of the signal.
http://www.voip-info.org/wiki/view/Asterisk+fxotune
Well, that clears it up a little. I think where I get confused is
that
Hi Matthew -
These techniques are not mutually exclusive, I usually want people to
use gain modification as the last step in trying to eliminate echo
(after balancing the hybrid and making sure you are using a good echo
canceller).
In the case of running fxotune, your zapata.conf software
Hi Edd -
I run a couple of asterisk servers all connecting
to international sip providers.
All three servers are on the same type of internet connection
(Martis/Diginet).
There isnt a shortage of bandwidth, and its not a codec issue, as ive
tried swapping codecs.
If its not a line issue,
Hi Drew -
I really don't know anything about how phone lines work in Singapore,
but maybe you could try using ground start signaling (fxsgs)?
- Noah
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Hi Zeeshan -
If you have multiple tenants using the same extensions range, you have
two options:
1) have the tenants call each other via their PSTN numbers, and then
dial the internal 1XX extension
2) assign a special prefix for each of the tenants to call each other.
For example, tenant one
Hi All -
I hope somebody can clarify for me what exactly fxotune does, and how
it is related to gain settings. I've been reading what appears to be
conflicting information from various sources.
I've got a box with an AEX800 with 6 lines (from Qwest) running
asterisk and zaptel versions 1.4.20.1
Hi All -
For the first time, I'm setting up SIP trunking between two asterisk
boxes. The calls themselves work fine, but I'm not able to get DTMF
working. I've tried using inband, rfc2833 and auto, and none of them
work. Maybe I'm missing something obvious? Here's my config:
Asterisk1
Hi Harry -
99% of all my users are calling from GSM phones, and my system
basically just plays some sound files back.
The PBX is connected to an ISDN-30 connection. Are there any modules
for playing MP3 files, so I can use them with commands like Play() and
Background()?
See
Hi Jared -
For the first time, I'm setting up SIP trunking between two asterisk
boxes. The calls themselves work fine, but I'm not able to get DTMF
working.
If you're connecting an Asterisk 1.2 box to an Asterisk 1.4 box (as it
appears that you are), you'll need to set
against Digium support, but the list actually
responds more quickly at this point. I think the Digium support staff
are in a situation of high demand and short staffing.
- Noah
Noah Miller wrote:
Hi Jared -
For the first time, I'm setting up SIP trunking between two asterisk
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