Displaying 20 results from an estimated 1000 matches similar to: "Faxing and PCI (was Re: Digium cards, sodisappointing !)"
2006 Mar 21
4
Junghanns and Digium TDM400?
Hi all,
is it possible to bridge a call between a Junghanns quadBRI card and a
TDM400 in the same server?
It should be I think, -- I am trying this and when an incoming call comes
in, it hangs both up at the moment the bridge is attempted
(and a subsequent 'qozap: dropped audio' error is show in the
/var/log/messages)
Any thoughts appreciated -- I've seen posts, but no clear
2006 Jun 22
5
Out of Office Auto Reply:
I will be on vacation from <22/06/06> to <30/06/06>.
I will not be reachable on my mobile. I will have limited access to mails, and please expect a delayed response.
In my absence, please contact the following:
Ray Richard or Safeer Mohammed
Thanks
H.Gireesh
2006 Feb 15
4
SIP and firewalls?
Hi
We are currently using Asterisk 1.2.4 with IAX and app_meetme for
conferencing, but are looking to move to SIP because of issues with an IAX
control we're using.
The reason we moved from SIP to IAX in the first place was because of the
poor NAT traversal with SIP. At that stage we were using Asterisk 1.0.*. How
does Asterisk 1.2.4 handle NAT traversal and firewalls compared to the older
2005 May 11
12
Snom 360
I am having major problems with the first run of Snom 360s that rolled
out last month.
I am working with the US vendor and they in turn are working with Snom
but I wanted to see of anyone else was using these or having issues with
them.
Issues:
Speakerphone/Hands Free volume spikes up and down during a call. You
have to manually set the volume during every call. This makes it totally
unusable.
2006 Jan 21
7
MeetMe Dialplan question
Hi,
is the following possible? I would like to transfer a call to my
"personal" MeetMe conference room and get transferred there
automatically as well. Currently I can transfer the call to the
conference, have to hangup and then call the conference number myself. I
would love to have this in one quick function.
Moreover is there a way to disable the "You are currently the only
2006 Feb 17
3
how to add stun functionality in asterisk
Hi friends !
I want to add stun functionality in asterisk.
can anybody give me some hint that how can i start that.
thanks in advance
Deepak Dhiman
2012 Mar 05
10
Compatibility of Hitachi Deskstar 7K3000 HDS723030ALA640 with ZFS
Greetings,
Quick question:
I am about to acquire some disks for use with ZFS (currently using zfs-fuse
v0.7.0). I''m aware of some 4k alignment issues with Western Digital
advanced format disks.
As far as I can tell, the Hitachi Deskstar 7K3000 (HDS723030ALA640) uses
512B sectors and so I presume does not suffer from such issues (because it
doesn''t lie about the physical layout
2006 Jun 19
3
Bristuff-0.3.0-PRE-1q and & florz patch compile trouble
Again trouble compiling bristuff-0.3.0-PRE-1q with the florz patch on a
x86_64 box (I guess nobody is using x86_64 platform or is able to fix this
themselves?)
First of all when bristuff is downloaded and compile is started it appears
that the bristuff Makefiles are badly broken.
The asterisk Makefiles all do see to find the kernel sources on a RHEL4
box in the proper directory, the pure
2006 Jan 17
4
How to find out if a new voicemail exists
Hi,
I would like to see if during a call a new voicemail was recorded. I want to send a SMS to mobile phones if someone recorded a message on our voicemail system.
I can use VMCOUNT to see if there are new messages in the Inbox but this will result in new SMS being sent even if the caller hangs up during the Voicemailpromt, at least if there are still unread/unheard messages in the inbox.
Is
2005 Aug 16
2
florz patch for bristuff breaks compile on x86_64?
After upgrading a CentOS 3.x box to CentOS 4.1 (both x86_64 with an
Athlon64) I also wanted to get the latest bristuff. Unfortunately
bristuff without florz causes the box to kernel panic within hours
(console will complain about bad frame received something).
It seems however that the florz patch will not work for x86_64 arch.
Bristuff -0.2.0-RC8j compiles fine without the florz patch, but
2005 May 30
1
I865, HFC-S etc.
Hi,
I'am having some problems with new mainboards and 3xHFC-S cards.
The the first problem was with interrupts, I mean if HFC-S card was using
interrupt i.e. 21 or higher - it didn't work.
Solved by disabling APIC.
However, still the driver behaves a little bit strange.
If card 0 & 1 is TE and 2 is NT, TE works fine, but NT is not working at
all.
If card 0 is NT and 1 & 2 TE - all
2006 Mar 31
1
Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.
Does anyone test Asterisk 1.2.X + bristuff-0.3.X and TDM card? We can't
get it to work.
-David
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Julian J.
M.
Sent: Friday, March 31, 2006 1:44 AM
To: Chris Earle; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: BRI
2012 Jun 17
26
Recommendation for home NAS external JBOD
Hi,
my oi151 based home NAS is approaching a frightening "drive space" level. Right now the data volume is a 4*1TB Raid-Z1, 3 1/2" local disks individually connected to an 8 port LSI 6Gbit controller.
So I can either exchange the disks one by one with autoexpand, use 2-4 TB disks and be happy. This was my original approach. However I am totally unclear about the 512b vs 4Kb issue.
2005 Jan 23
4
Florz patch for zaphfc
Has anyone had any success using the Florz patch for zaphfc ?
I have a * system with 2 HFC cards which is working fine with 2 PTP ISDN
lines however the users are complaining of crackles on the line which I am
assuming is related to the IRQ issues raised by Florz.
I have tried to use the patch but it errors trying to patch zaphfc.h
Any help would be appreciated.
Regards,
Stuart
--
No virus
2006 Nov 16
1
zaptel, bristuff zaphfc, and florz question
Hi,
We've been using zaphfc single ISDN cards as cheap Zaptel timing
sources for our Asterisk boxes for a long time, and in the asterisk
1.0.x series, had zero problems doing so.
I now have some boxes with Zaptel 1.2.x (with a mixture of 1.0.x
asterisk and 1.2.x asterisk), and this setup no-longer seems stable -
By plugging or unplugging the ISDN cable, and sometimes just randomly
the card
2007 May 06
0
Asterisk 1.2.14-BRIstuffed-0.3.0-PRE-1y
Hi all,
I have a hangup problem when i get incoming calls on my ISDN interface.
I use ISDN network controller [HFC-PCI] and asterisk with florz patch.
Logs when the hangup happens follows:
May 6 20:52:47 NOTICE[11532] cdr.c: CDR on channel 'Zap/1-1' not posted
May 6 20:52:47 NOTICE[11532] cdr.c: CDR on channel 'Zap/1-1' lacks end
May 6 20:52:47 DEBUG[11532] pbx.c: Expression
2005 Feb 14
1
Bristuff-0.2.0-RC5 florz patched weird error and no outgoing calls?
I applied the florz patch but my problems remain. Now I get all sorts
of weird errors on the console and I cannot make outgoing calls. Incoming
calls work as expected. I am using a single HFC-S card with BRI.
Any clue what these errors below are?
Ri = 44651 TEI msg = 3 TEI = 7f
Ri = 3800 TEI msg = 3 TEI = 7f
Ri = 42399 TEI msg = 3 TEI = 7f
Ri = 42409 TEI msg = 3 TEI = 7f
Ri = 22078 TEI msg = 3
2006 Jan 25
1
ISDN D-channel disconnects for a minute every 5 minutes
I have a problem with Asterisk-bristuffed using a zaphfc card.
I am located in the Netherlands, so I have an ISDN line from KPN. When I
start Asterisk, and plug in the ISDN line, everything works perfectly for
about 5 minutes. And then the ISDN line is down for 1 minute, and after that
minute, the line comes back up and works for another 5 minutes. Every time
the line goes down I get the error
2006 Feb 09
3
Corrupt CDR records in Asterisk 1.2.x
I have a problem with CDR recording in Asterisk 1.2.x. This is the
situation:
An Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1 with florz) machine with a single
HFC-S ISDN BRI card. I log the call records to both the Master.csv and
MySQL.
The problem is that when an incoming call from the ISDN line is logged to
the CDR, the "src" and the "clid" field show up as something like
2006 Jan 21
1
Is sip1.voipbuster.com corking reliably for others on list?
I am trying to move from IAX2 to SIP for voipbuster, moving at the same
time to sip1.voipbuster.com.
When I try calling out, I see that there is SIP exchange, and in many
cases also RTP data being exchanged.
Hover in a very large number of attempts the connection is not
established. Half of the time there is no RTP, the rest of the time there
*is* RTP data flowing in two ways, but no ringtone is