2008/10/24 Steve Murphy [EMAIL PROTECTED]
Well, if you have 50K extensions, you'll find the trunk/1.6.x versions
a bit easier to bear in this respect; I've redone the reload process
so that it takes longer, but the magic is that it locks the dialplan
and swaps in the new dialplan in about
2008/10/23 Brendan Martens [EMAIL PROTECTED]
Indeed I am going for pure voip and trying to figure out how to
implement t.38, as you suggest.
On Oct 23, 2008, at 2:08 AM, Olivier wrote:
I think Brendan is asking about endpoints (how to connect fax
machines to pure VoIP).
Short answer:
On Thu, Oct 23, 2008 at 11:30:29PM +1100, Fernando Serto wrote:
Hi,
I've been very puzzled lately. I installed a phone system for a friend
a few weeks ago, and they're having a problem that I can't get rid of,
actually 2 problems. Before I go into the problems, let me tell you
about the
On 24 Oct 2008, at 03:57, David Gibbons wrote:
Dare I ask why you want to do this?
Cheaper than buying an AIM-CUE? And certainly more flexible.
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asterisk-users mailing list
To
Hi
I've found a solution for what i think is exactly the same problem here:
http://bugs.digium.com/view.php?id=13491nbn=6
Regards
Enrico
Julien Claassen ha scritto:
Hello everyone!
I've just switched from Asterisk 1.6.0-beta9 to 1.6.0.1 and my mISDN is
not
working. Here's what
On Thu, 23 Oct 2008, Karl Fife wrote:
We have a number of DID's that do the standard VoIP tricks: ringing
multiple locations, findme-followme etc. What is happening more and
more is that customers call those DID numbers, and draw the reasonable
conclusion that they are calling mobile numbers
Olivier wrote:
Linksys PAP2 or 3102 for instance
or Patton M-ATA
In fact, I would say most analog gateways with FXS port should also
support T.38.
In this case, your setup would be :
That list rather poorly supports your argument. The PAP2 and the PAP2T
do *not* support T.38, despite
Ahh now I see.
I am a major proponent of Cisco hardware but it works pretty well with * using
either the SIP image or the SCCP image. I would need to have some pretty
specific feature needs in order to complicate things with a setup that required
CME and * to interact.
On the other hand if
Hi,
I've managed to build the zaptel modules including ztdummy; ztdummy is
installing fine in the modules list and the relevant device structures
are present.
lsmod | grep ztdummy gives:-
ztdummy 5160 0
zaptel186916 1 ztdummy
rtc12372 1
It's definitely just for fun, I wouldn't think to try to implement
such as setup for a client unless I were really comfortable with the
setup!
On Fri, Oct 24, 2008 at 8:36 AM, David Gibbons [EMAIL PROTECTED] wrote:
Ahh now I see.
I am a major proponent of Cisco hardware but it works pretty
Well, there's no harm in _looking_ at it.
ram wrote:
look at Vicidial
ram
On Thu, Oct 16, 2008 at 4:46 PM, yavuz yildirim [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
hi everybody
This is Yavuz YILDIRIM
I am software developer.I have a some problems in
I'll look into those devices mentioned.
I think that I have one last question... I don't intend to have a
hardware fax machine on our end, I really just want it to get to
asterisk then email it from there. I know this can be done with
hylafax/iaxmodem etc, I actually have gotten that to
Do you have any recommendations for good ones, or, non-buggy ones?
Brendan Martens
On Oct 24, 2008, at 7:48 AM, Steve Underwood wrote:
Olivier wrote:
Linksys PAP2 or 3102 for instance
or Patton M-ATA
In fact, I would say most analog gateways with FXS port should also
support T.38.
In
Gordon Henderson wrote:
On Thu, 23 Oct 2008, Karl Fife wrote:
We have a number of DID's that do the standard VoIP tricks: ringing
multiple locations, findme-followme etc. What is happening more and
more is that customers call those DID numbers, and draw the reasonable
conclusion that
2008/10/24 Steve Underwood [EMAIL PROTECTED]
Olivier wrote:
Linksys PAP2 or 3102 for instance
or Patton M-ATA
In fact, I would say most analog gateways with FXS port should also
support T.38.
In this case, your setup would be :
That list rather poorly supports your argument.
Yes,
2008/10/24 Brendan Martens [EMAIL PROTECTED]
Do you have any recommendations for good ones, or, non-buggy ones?
It should be wise to also ask your ITSP as T.38 interop is far from easy ...
Would you go with pure-VoIP or would you keep an analog line ?
Please does anyone have Freepbx or Trixbox Powerpoint Presentation?
Thanks
_
Connect to the next generation of MSN Messenger
On Fri, Oct 24, 2008 at 01:38:14PM +0100, Richard Horton wrote:
Hi,
I've managed to build the zaptel modules including ztdummy; ztdummy is
installing fine in the modules list and the relevant device structures
are present.
lsmod | grep ztdummy gives:-
ztdummy 5160 0
I've restarted the service and zombie channels were killed.
Daniel
On Wed, Oct 15, 2008 at 3:29 PM, Steve Murphy [EMAIL PROTECTED] wrote:
On Tue, 2008-10-14 at 17:24 -0500, Daniel - Asterisk wrote:
Hello everyone,
I'm getting DIALSTATUS=CHANUNAVAIL when a call is trying to get one of
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 telco (BT) is suggesting we replace our analog
Olivier wrote:
2008/10/24 Brendan Martens [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
Do you have any recommendations for good ones, or, non-buggy ones?
Some of or resellers are using 2102 apparently with no issues :)
Senad
www.bicomsystems.com
On Fri, 24 Oct 2008, Drew Gibson wrote:
Gordon Henderson wrote:
On Thu, 23 Oct 2008, Karl Fife wrote:
We have a number of DID's that do the standard VoIP tricks: ringing
multiple locations, findme-followme etc. What is happening more and
more is that customers call those DID numbers, and
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 telco (BT) is
On Fri, 24 Oct 2008, 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 telco
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
I've been following this thread and trying to sort out what is wanted, what is
available, and why. Comments to the following would be appreciated and might
be useful to others.
1. Why would anyone originate a FAX via VoIP? If it has to go through a bunch
of translation steps at both ends,
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 telco (BT) is
On Oct 24, 2008, at 9:49 AM, Wilton Helm wrote:
I've been following this thread and trying to sort out what is
wanted, what is available, and why. Comments to the following would
be appreciated and might be useful to others.
1. Why would anyone originate a FAX via VoIP? If it has to go
2008/10/24 Tzafrir Cohen [EMAIL PROTECTED]:
On Fri, Oct 24, 2008 at 01:38:14PM +0100, Richard Horton wrote:
Try running zttest . Does it print anything or is simply hung?
Hangs - I also found after sending my message my syslog filling up
with rtc interupt missed messages - don't think my
Hi all,
I received a report of a client which stated that two of its agents are
logging in to the queues when they actually arent there working. They
appeared to be logged on all night. They thought they werent logging off
correctly, but they checked one of them and he was following the
Jonn R Taylor wrote:
Install a T1 between the Panasonic and Asterisk and program the T1 in the
Panasonic as a other custom PBX. VOIP card would be the best.
Jonn
One thing to beware of with the Panasonic VoIP card, is that I have
found no way of getting it to pass out of band DTMF,
People can't figure out e-mail as it is, they aren't going to figure out how
to fax via e-mail..
I can understand people saying that. Myself, I'd take E-Mail any day. I've
been messing with FAX at various facilities for years, and have found it
unreliable, as have most people I talk to.
On Fri, 24 Oct 2008, Alan Lord 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
I've got a problem that keeps popping up with my reception phone.
It is a IP 650 and the receptionist - on three occassions - has accidentally
hit the Forward softkey just before she enters the Page All keystrokes
and then all future calls get routed as an overhead page.
I will admit, the
In your phone configuration file, for all lines:
divert
divert.fwd.1.enabled = 0
divert.fwd.2.enabled = 0
divert.fwd.3.enabled = 0
divert.fwd.4.enabled = 0
divert.fwd.5.enabled = 0
divert.fwd.6.enabled = 0
/
The worst part is this is the
Kristian Kielhofner wrote:
On 10/23/08, Bruce Komito [EMAIL PROTECTED] wrote:
We've had LOTS of problems with Sonicwalls doing bad things to SIP and RTP
connections. I've seen the delay thing, as well as the Sonicwall throwing
away entries from the ARP table because of inactivity.
From my experience, Sonicwall is a nightmare with SIP if you do not have
Enhanced OS.
General rules I use:
-Do not use SIP transformations (the VOIP tab), these cause random RTP issues,
and once you start forwarding calls between users, all things go to heck. You
are better off using
On Oct 24, 2008, at 1:12 PM, Bill Andersen wrote:
I've got a problem that keeps popping up with my reception phone.
It is a IP 650 and the receptionist - on three occassions - has
accidentally
hit the Forward softkey just before she enters the Page All
keystrokes
and then all future
That did the trick. And yes, I agree it is a very poor design. After
looking at how
it all transpired, it made more sense as to why it has happened lately. I
recently
purchased a wireless headset for the receptionist. She would not use her
corded
headset because she also does some filing
Can i install Asterisk beside Nortel PCM, just for recording all calls on E1
(incoming and outgoing calls)
I want to get the E1 into Asterisk (Digium)
how can this scenario be achieved in details please ?
Date: Sat, 25 Oct 2008 07:42:09 +1300
From: [EMAIL PROTECTED]
To:
On Oct 24, 2008, at 9:29 AM, Gordon Henderson wrote:
On Fri, 24 Oct 2008, Drew Gibson wrote:
Gordon Henderson wrote:
On Thu, 23 Oct 2008, Karl Fife wrote:
We have a number of DID's that do the standard VoIP tricks: ringing
multiple locations, findme-followme etc. What is happening more
John Todd wrote:
Instead of disabling the keys on the phone, why not just put logic in
your dialplan that refuses calls to the paging extension except when
the originator is a handset? If the call != handset originated, then
send to the voicemail of the handset that bounced the call.
after a fresh installation of Freepbx
1- How can i make Freepbx send voicemail to Email. (the Linux mail
configuration steps)
2- How can i operate Fax machine and How it will be able to send the FAX to
email.
3- Is there any software application i can run to monitor live the calls and to
see
I'm having an unusual problem at one of my branch offices. Every now
and then they will make a call and the person they call is unable to
hear them, but they are able to hear the person. The Asterisk server
has only one ethernet interface and is on the same physical network as
the 2 snom 300
after a fresh installation of Freepbx
1- How can i make Freepbx send voicemail to Email. (the Linux mail
configuration steps)
2- How can i operate Fax machine and How it will be able to send the FAX to
email.
3- Is there any software application i can run to monitor live the calls and
to see
When using an Asterisk iaxy adapter every 15 to 30seconds there is a loud
annoying beep during conversation.
Does anybody know how to stop it?
--
#Joseph
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asterisk-users
How is your asterisk server connected to the PSTN? SIP/IAX out...ISDN/T1
out? Etc...
Are you looking for lost RTP between * and internal phones or * and external
provider?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson
Sent: October 24,
The asterisk server is connected to the PSTN via a Rhino R4FXO-EC card.
The lost RTP would have be between the Asterisk server and the phones.
There are only 2 phones in the building, 2 lines coming in to the
asterisk server and the server is on the same ethernet switch as the
phones. The
Well, if this is snom specific I can't offer more insight. It really sounds
like misconfigured iptables and/or sip helper (conntrack/nat/etc).
Are you sure your IP address is right in your sip.conf? If you don;t have
NAT set to yes for these phones, they will trust the sip header for IP
address
We have seen cases where an IP address conflict caused something like this.
You can take Wireshark traces on the PC (possibly run them in a loop so that
you have a pretty long context) and if you have one-way audio be quick to log
on to the web interface of the phone and also take a wireshark
queuestats?
Original Message
Subject: Re: [asterisk-users] Fresh installed box
From: "Matt Gibson" [EMAIL PROTECTED]
Date: Fri, October 24, 2008 6:16 pm
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
asterisk-users@lists.digium.com
after a fresh installation
http://www.trixbox.org/forums/trixbox-forums/open-discussion/asterisk-guru-queuestats-install-guide-video
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark
If you don't like the forward key why not simply get rid of it.
With firmware 3.1.0 all you need to do is add one line to your config file:
softkey.feature.forward=0.
While you are at it, you might (or might not) like to get rid of buddies and
mystatus.
softkey.feature.buddies=0
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