Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Wednesday, May 20, 2009 4:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
crashed me) with
no joy. Any suggestions? Hardware is Dell Poweredge 1650/1550 and
TDM410P/TDM400P.
Any reason you're using the r/m option at all? Since this is an analog
card I would leave the r/m off and just let asterisk use the in-band
progress from the telco.
-Dave
*after* Asterisk was compiled.
Recompile Asterisk again and make sure
/usr/lib/asterisk/modules/chan_dahdi.so is created when you make install.
-Dave
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I just inherited a client that is using a Switchvox system. I normally
install a CentOS based system with freePBX and some custom endpoint
management stuff for Polycom phones. This Switchvox is making me feel a
bit stifled. I am having nightmares of another recent encounter with
Trixbox
I was not endorsing a particular product or asking for recommendations about specific products. I personally tried Trixbox and was not really satisfied with the results. The last time I was asked about an appliance I referred the person to Switchvox. There were a few other responses that seemed to
I would think that VoIP over VPN is a bad idea as UDP packets need to be
in realtime not corrected by the TCP of the VPN.
That depends very much on the VPN in use.
OpenVPN doesn't suffer from this problem. Although it's SSL-based
(and one might think it does everything through SSL-over-TCP),
http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.confmailcmd
Mailcmd allows the administrator to override the default mailer
command with a defined command. Mailcmd takes a string value set to the
desired command line to execute when a user needs to be notified of a
voice mail message.
This looks like a SIP registration problem to me. You should enable sip debugging. If the problem/solution does not become obvious then post the results so we can take a look.localhost*CLI sip show peersName/username Host Dyn Nat ACL Port Statusnew/42634 203.196.128.56 5080 OK (112
. Has this topic come up for conversation in the past and if so then what was the outcome?Thanks!Dave
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that checks if the number of messages in
the inbox is greater than the last time it was called (so a user doesn't
get a call after they check their messages) and if so, create a call
file to contact them and automatically connect them to the voice mail
system. If you want it I can send it to you.
-Dave
BTW, can someone explain to a libart major like me (;-)) where echo
comes on in a telephone conversation? I seem to recall it's due to the
length of the line between the CO and the local party, but I'm not
sure.
I'll try.
Echo occurs when part of the signal traveling in one direction
on the
go buy
some:
http://www.voiplink.com/Extended_Handset_Cord_Detangler_p/detangler-e.htm
I personally hate detanglers. I get more complaints about static on
calls that result from these things than anything else, but I need to
provide some solution.
Thanks
-Dave
P.S. For anyone looking
[r...@vicidialnow ~]# iftopinterface: eth0IP address is: 192.168.0.100 1.91Mb 3.81Mb 5.72Mb 7.63Mb9.54Mb└──┴──┴───┴──┴───What package did you get this utility from? I want to try it out..
Greetings,I am trying to harden an Asterisk box without affecting the staff too much. The cheap Linksys router forwards ports 5060-5080, 1-35000 and 22 to the Asterisk box. The road warriors were connecting directly to Asterisk via our public IP which allowed their soft-phone passwords to be
If I already have VOIP, can I use them or is it a special kind of service I'd need?
explain "already have voip" ?
I debated chiming in on this conversation. Your subject asking the same question as the text of your email. Basically you install / setup your Asterisk server. Once installed you
OK, but I do need a VOIP provider, then, right? Not just an internet provider?
And is it a special kind of VOIP you have to sign up for or can any VOIP provider/program fulfill the needs?
Most consumer VoIP providers will not meet your needs. I would strongly suggest you find a provider that
Hello, I've started to do some research into the new 4G wireless
standard, and there's one part of the standard that intrigues me.
Apparently all data is packet based, including the phone calls. Every
phone will have its own IPv6 address. This seems to pave the way for
a call to go
I have never been on a mailing list where this debate has not come up. It still makes me laugh after 20+ years of technology work.
Most coders learn "just enough to get by" not realising their productivity
and the quality of their product improves when they learn all of the tools
in the box
to be inband from the Polycom and Asterisk was configured to something
different. Asterisk therefore didn't detect and translate the DTMF to
out of band when it went over the IAX trunk.
-Dave
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is sent back to the caller.
-Dave
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1) IP-phones get there IP from a DHCP
The source of the address is not the issue.
I still see no register-message on the CLI. This really should happen
now, as they are defined host=dynamic !
I suspect you have not [correctly] configured the phones to register to the Asterisk server.
You
not
sure what it checks for though. I did set the dahdichanname=no in the
asterisk.conf if that makes any difference. It seemed to in calling the
channel in the dialplan but didn't seem to effect the meetme app.
Thanks,
Dave
Relevent bits from lsmod
Module Size Used by
dahdi_dummy
in my
extensions.conf but since I set that as a global variable it was easy to
fix. Just set it to refer to DAHDI instead of ZAP. I still think there is a
bug somewhere but I am unable to find it.
Thanks for the help.
Dave
On Fri, Apr 3, 2009 at 10:09 AM, Martin asteriskl...@callthem.info wrote
Yes that was on a fresh build. I updated from zaptel to dahdi at the same
time as moving from Asterisk 1.4.22 to 1.4.24.
On Fri, Apr 3, 2009 at 11:23 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Thu, Apr 02, 2009 at 10:22:55AM -0700, Dave Poirier wrote:
We recently updated our
else come across this? Any suggestions?
Thanks,
Dave
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On Thu, Apr 2, 2009 at 4:36 PM, Elliot Murdock murdo...@gmail.com wrote:
Hello!
I am trying to configure my digium TE220 dual-span pci express card
with Dahdi. I seemed to have managed to set up the card with the
Dahdi kernel, as demonstrated by executing dahdi_scan:
[1]
active=yes
...@xorcom.com
description:DAHDI OSLEC wrapper
depends:dahdi,echo
vermagic: 2.6.27.19-smp SMP preempt mod_unload 686
Try building DAHDI with the steps detailed here and see if you have
better luck:
http://lists.digium.com/pipermail/asterisk-users/2009-January/225299.html
-Dave
to and
you'll get the echo module with DADHI. It requires you download 2.6.28
but not that you are running 2.6.28.
2009/4/1 Marco Sambo derwid...@gmail.com
But I don't have also echo
modinfo echo
modinfo: could not find module echo
2009/4/1 Dave Fullerton dfullertaster
or ATT
always messes this up so... tips.
Thanks
-Dave
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SIP was written in such a way that the hashes it sends for passwords
could, with only a trivial rewrite of the server code, be SHA1 instead
of MD5 -- which would increase security to the level that, currently, it
would be far more trouble than it's worth to even bother to attempt to
crack.
I
can
use them for more!
Cheers,
Gordon
I think this is what you want:
http://www.voip-info.org/wiki/view/Asterisk+func+Devstate
-Dave
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. This same thing happens at home with
my TDM400P so I'm inclined to think it's not exclusive to the channel
bank. Anyone have any ideas?
Thanks
-Dave
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best bet is to increase the number of
lines you have.
-Dave
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kewlstart (Default) (Echo Canceler: mg2) (Slaves:01)
channel 04: fxs kewlstart (Default) (Echo Canceler: mg2) (Slaves:04)
snip
What are the contents of chan_dahdi.conf in /etc/asterisk? Did you
specify what signalling to use there?
-Dave
hanguponpolarityswitch=no, and tried loop start
signalling for the heck of it and that didn't work either.
Does anyone have any suggestions of additional things I could try?
Thanks in advance,
-Dave
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Dave Fullerton wrote:
Hello All,
I'm trying to connect Asterisk to an Executone phone system with an
analog DID card and I'm hoping someone can help me figure out what I'm
doing wrong. The Executone DID card provides battery to the telco, when
the telco wishes to dial a DID it goes off
*CLI
Must it be defined somewhere?
Cheers,
Yes, you need the sip_notify.conf file in /etc/asterisk. The sample file
that's in the asterisk source has the definition for polycom's in it.
-Dave
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machine B first and then A everything
works fine. I'm using dahdi_linux 2.1.0.4 on both.
I know I can just use SIP or IAX or anything else to connect these two
machines, but I'm using this as a learning experience to play with PRI
setups.
Thanks
-Dave
reza adinata wrote:
Hi guys,
I am trying to compile zaptel, using debian 4r5. However what I get in
zaptel 1.2.27 after make is below :
You do not appear to have the sources for the 2.6.18-6-486 kernel
installed (under ).
make: *** [modules] Error 1
tried to change the source with
In Florida some new subdivision developers have sold the
phone/cable/internet rights to a provider. They run fiber to each house
and then have the uplink to provider which isn't a traditional telco.
You can't get another provider as satellite dishes are limited in
covenants and restrictions
set. Everything is configured the same
in asterisk, you just use a dynamic span instead of a physical one. You
will still need one side to have a timing source (I did get mine to work
with just ztdummy).
-Dave
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Hi, all. My subject line says it all: is there a WiFi SIP phone with VPN
abilities? Failing that, a WiFi phone that runs Linux? I already know
one phone that does meet my requirements -- the iPhone. The new software
comes with a Cisco VPN client, and a SIP client can be had from
into using AEL instead (or in addition to)
for future work.
TIA
I use AEL. I find it much cleaner to look at and not having to deal with
priorities is a bonus.
-Dave
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asterisk
One of my user was asking, can he use VPN to access asterisk ?
What does it mean ?
And its possible ?
How ?VPN
Yes, it's possible.
As one example: I have the OpenVPN software installed on my Asterisk
server, and on my Nokia N810 wireless Internet tablet. The tablet is
configured to use
Paul Chambers wrote:
I'd recommend dnsmasq. I've been running it for a few years, and it
works very well for me. Besides DNS, it optionally supports DHCP
(integrated with DNS) and TFTP. A basic (i.e. normal :) configuration is
easy to set up, though there's plenty of depth if you need to
the same issue with dahdi-linux-2.1.0.3 using the staging
drivers from 2.6.28.
-Dave
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?
It now modprobe's without issues. I'll get to trying it out later.
-Dave
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I want to detect brute-force password hacking attacks - thus if there
are too many failed login attempts for a SIP account I want to lock
this account.
Does somebody have any ideas how this could be implemented?
The usual method (I think) is to monitor the log files, and
detect repeated
I may be over simplifying but I would have a serial number object that
gets incremented anytime it is called and will be set to 0 at start-up.
I would then use it to generate a UUID like this:
MAC.serialid.64bit timedate
I suggest reviewing RFC 4122, which discusses UUID formats in some
alertInfo for different rings in the sip.cfg and
then in asterisk set the alert info header in the dialplan.
You can change the reg.x.ringType on each registration in the phone's
config file.
See the SIP admin guide for details.
-Dave
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and then call the macro with the M() option to Dial (see
the Dial app help text).
-Dave
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the job:
http://wiki.sangoma.com/sangoma-wanpipe-usbfxo
People have been reviewing betas since early September so hopefully it
will be released soon.
-Dave
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and
replace it with a generic X.T or add a digit map for the number you
are dialing that is greater than 10 digits long.
-Dave
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the variable is defined but in
most cases (if any) that doesn't matter anyway.
But I guess it wouldn't hurt to add a DEFINED() function to
Asterisk.
if (DEFINED(myvariable)) {
// ...
}
Isn't that what EXISTS() is for?
-Dave
when I forgot to put quotes around
channels like that (took me forever to realize that one). Since you have
them I would say this is a bug. What version of asterisk are you running?
-Dave
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Tia , JimL
You're using the combined tarball that has both dahdi-linux and
dahdi-tools. That makeopts files is for the tools side (as shown the path).
-Dave
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)
manager.c:1732: error: (Each undeclared identifier is reported only once
manager.c:1732: error: for each function it appears in.)
make[1]: *** [manager.o] Error 1
make: *** [main] Error 2
Are you by chance using 1.6.0.2?
Try grabbing 1.6.0.3-rc1 or 1.6.0.1 instead.
-Dave
anyone give me a starting point?
Thanks,
Brent
Check the entries for office1 and office2 servers in sip.conf. If they
have a callerid= entry comment it out and do a SIP reload. When it is
set asterisk overrides the caller ID sent to it.
-Dave
Brent Davidson wrote:
Dave Fullerton wrote:
Check the entries for office1 and office2 servers in sip.conf. If they
have a callerid= entry comment it out and do a SIP reload. When it is
set asterisk overrides the caller ID sent to it.
-Dave
There aren't any callerid= entries in any of my
Brent Davidson wrote:
Dave Fullerton wrote:
Brent Davidson wrote:
Dave Fullerton wrote:
Check the entries for office1 and office2 servers in sip.conf. If
they have a callerid= entry comment it out and do a SIP reload. When
it is set asterisk overrides the caller ID sent
.
Mike
The SIPPEER function should allow you to extract what context is defined
in sip.conf.
-Dave
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that would use chanspy in
whisper mode to play the page through the current audio device if the
phone is busy. I don't know how to go about doing that however.
-Dave
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/chan_iax2.c: SENTINEL is not defined
in 1.6.0
-Dave
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Tzafrir Cohen wrote:
On Tue, Dec 02, 2008 at 01:22:16PM -0500, Dave Fullerton wrote:
Is anyone else having difficulty compiling 1.6.0.2?
It bombs out when compiling manager.c
manager.c: In function 'action_getvar':
manager.c:1732: error: 'SENTINEL' undeclared (first use in this function
chooses the highest number
available channel. This is used to reduce glare on analog or T1
(non-PRI) channels that are part of a hunt group.
-Dave
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There are also settings which will turn on local echo cancellation for
the handset, headset and/or speaker phone. I don't recall their names at
the moment. They are off by default on the handset and headset unless
you're using a very recent (3.0+) SIP app.
Tim Nelson wrote:
I'm not sure about
Coming from outside the network, setting up for a couple rounds of
NATting isn't going to work well. They are not seeing it between
phones. Others, using the polycom phones have reported echo between two
SIP on a 4ms ping trip.
Could this be due to a purely acoustic echo within the Polycom
().
Hopefully that's of some help to you.
-Dave
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/voice/soundpoint_ip/soundpoint_ip450.html
-Dave
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Jared Smith wrote:
On Tue, 2008-11-11 at 10:41 -0500, Dave Fullerton wrote:
Anyone looking for firmware should get it now before it disappears.
It's my understanding that this isn't a fluke, but that Polycom has
indeed changed their policy and will no longer you to go through your
reseller
Tilghman Lesher wrote:
On Tuesday 11 November 2008 10:50:26 Dave Fullerton wrote:
Jared Smith wrote:
On Tue, 2008-11-11 at 10:41 -0500, Dave Fullerton wrote:
Anyone looking for firmware should get it now before it disappears.
It's my understanding that this isn't a fluke, but that Polycom has
, end-user customers can only download previous
software. Please work directly with the Polycom Certified VoIP Reseller
you purchased the products from to obtain the most current and
appropriate software.
-Dave
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Andres wrote:
Phil Knighton wrote:
Hello all
What I'm looking for is some plain speaking advice on ISDN.
Currently using 4 analog lines connecting via a four port TDM400P FXO
card. We need to physically move our installations, and on requesting
the analog lines be moved - our
any problems. I also can't recommend Zyxel's
support enough, I had initial concerns about the PoE budget and within a
couple of rings, I was through to someone who actually knew the product
inside out.
Kind Regards,
Dave Walker
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cdr program ? i saw that there is a cdr.so
module that gets loaded - can it help me in anyway
The FreePBX forum would be a better place for this, I would imagine you
will get an answer sooner.
HTH
Kind Regards,
Dave Wa;ler
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be using and include in
the users.conf
Thanks,
Dave
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(Custom:nightmode)=BUSY);
else
Set(DEVSTATE(Custom:nightmode)=NOT_INUSE);
}
}
You'll need the DEVSTATE backport in order to use this example. See the
links at the bottom of this page:
http://www.voip-info.org/wiki/view/Asterisk+func+Devstate
-Dave
=phone-isdept
callerid=Dave Fullerton 3917
mailbox=3117
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=rfc2833
qualify=no
call-limit=10
limitonpeers=yes
[3900](lan-soundpointip)
username=3900
secret=sdjghdfkjhgdf
context=phone-operator
callerid=Operator 3900
[3917](lan-soundpointip)
username=3917
secret=dfkghdjfhdkfd
context=phone-isdept
callerid=Dave Fullerton 3917
mailbox=3117
In my
I just received an email notice from FWD about $30 membership fee.
My question: Is the email genuine? Did anybody else receive it?
I'm just trying to be sure that it is real and not a scam.
The (FWD) does not do anything to authenticate such emails (implementing
GPG/PGP signature etc.)
/log/asterisk/cdr-csv, but that's not much
help unless there's some other software that can turn the CSV into
something more user friendly. Is there software like that?
--
Dave Welsh
Quality of Course
(613) 749-8248
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that no one need to do any emergency rewiring?
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Dave Welsh
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at 02:10:34PM -0400, Dave Welsh wrote:
If I buy two AA50s can I set them up so that everything runs
through the
first one, but the second one will take over if the first one goes
down?
I can see the extensions recovering, because they use ethernet, but
what about the FSO lines
Paul Belanger wrote:
Can anybody confirm if this is the correct power adapter outputs:
12V DC 400mA
You adapter will have to outputs listed on it.
Thanks,
PB
That is correct for a 301.
-Dave
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, but the odds of someone other than callcentric placing a call to
my asterisk box and calling a 1777xxx phone number are pretty slim.
-Dave
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this. Any suggestions?
Doug
How about something like this:
exten = _X.,n,GotoIf($[${EXTEN:0:1} = 9]?not-parked,s,1)
You may need to tweak the extension pattern as this will match anything
that begins with 9.
-Dave
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I'm preparing for a client install of * by doing a fresh one in-house.
Unlike my earlier installation that runs asterisk as superuser, my
current experimental box runs without such privilege. This is causing
it to moan that it can't set TOS. I absolutely don't want to install it
on the
Hans Witvliet wrote:
There's not much that can stand lightning (not just a direct hit), so
you cant't blame the sipura box for that.
Even when it was build, using a Faraday-cage with double insulation with
optocouplers, the amount of energy picked up by a 3 km line is beyond
commercial
SIP wrote:
Joseph wrote:
I need another Sipura 3K and the replacement I think is Linksys SPA3102.
Any input on how reliable is it?
We have a few dozen subscribers using them at any given point in time. I
and my wife even use them at our respective homes. Rock solid stable.
No issues
Joseph wrote:
On 07/11/08 18:37, Dave Cotton wrote:
SIP wrote:
Joseph wrote:
I need another Sipura 3K and the replacement I think is Linksys SPA3102.
Any input on how reliable is it?
We have a few dozen subscribers using them at any given point in time. I
and my wife even use them
Simon wrote:
Hi There,
Has anyone managed to get 2 AVM ISDN Fritzcard's working in with a 2.6
kernel system?
Yes, with Suse 10.2/10.3 and chan_misdn.
/usr/sbin/misdn-init config wrote this:-
#
# Configuration file for your misdn hardware
#
# Usage: /usr/sbin/misdn-init
Matt Watson wrote:
On June 30, 2008 08:44:44 pm Simon wrote:
Hi There,
I am looking to build an Asterisk server with dual AVM Fritz!PCI cards
linked to 2 BRI in New Zealand. Just wondering if anyone has done
this, and if you have any ideas about the best disto choice for this
task?
Let
I'm looking for a way to delay the disconnection of a call to
a SIP extension (or pad it with silence) for a few seconds, after
an incoming call to that extension hangs up.
Rationale: I have an Asterisk PBX (current 1.4.20 codebase), with
a Leadtek BVP8051S ATA hooked to an analog phone which
dean,
I am an active member of AUG NYC... you can email me off list for any
info you need.
also, I am preparing to start a south jersey * UG. the phila group is
waning...
thanks,
daveC
Dean Collins wrote:
This is an email to all New
York
based Asterisk users.
asterisk to continue processing the media. Is that not the case? I've
scraped through what documentation I can find and googled but the only
additional info I could find was to set the externip=MYPUBLICIP. Can
anyone with a similar setup help point me in the right direction?
Thanks,
Dave
Brent Davidson wrote:
Asterisk Development Team wrote:
The Asterisk.org development team has announced the release of Zaptel
versions 1.2.25 and 1.4.10. These releases contain many bug fixes as
well as performance enhancements.
A couple of the more major changes include: modifications to
.
Please excuse me if the correct terminology has not been used, I am fairly
new to the world of VoIP and Asterisk.
Regards,
Dave
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, numbers and send it through to their
internal PBX. We would also do the billing and provide the actual
connection.
Anything else that you can add would be excellent.
Thank you so far.
Dave
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them.
Where is that? I don't seem to get that option. What I want is an announced
call transfer to another SIP device.
I've always used # which, in my features, conf is configured for
attended transfer and ## which is configured as blind.
Dave Cotton
to pay.
Dave Cotton
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