I've missed?
We have it working on a dl320, several ML350's, ml310, but never tried on a
dl380 yet.
We had serious issues in the past when iLO was enabled on a 350. disabling
iLO on that machine helped us. (we had irq errors)
You could try disabling iLO, just to make sure.
regards,
stoffell
or post to the mailing list if you have any further
news/experiences..
cheers,
stoffell
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/lib/libc.so.6
(gdb) q
Thanks in advance for any pointers.
cheers
stoffell
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Flags: Call
PRI Logical Span: Implicit
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Regards,
stoffell
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running in.
the whole point is I wanted to move away from mISDN (for other reasons) to
the digium-way so I can use native digium (and only digium) software. :-)
regards,
stoffell
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On Tue, Nov 11, 2008 at 11:54 PM, Shaun Ruffell [EMAIL PROTECTED] wrote:
If you are a user of the B410P card, and are able, please test these release
candidates in your environment. To test you will need version 1.4.4 or
greater of libpri and version 1.6.0 or greater of Asterisk.
Shaun, this
Hi all,
In the changelog of bristuff, as of version 0.4.0test4(test5) the
beronet cards should be supported.
Can anyone confirm if the beronet 2,4 and 8 ports version are
supported by qozap now?
Regards,
stoffell
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'looked' fine
(no irq sharing etc..) but the problem was related to iLO. Disabling
iLO made it all work.. So if you have issues, try that for starters..
What distro and versions are you using?
cheers,
stoffell
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:
http://www.voip-info.org/wiki/view/Asterisk+dimensioning
http://www.voip-info.org/wiki/view/Asterisk+hardware+recommendations
cheers,
stoffell
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features and possibilities of hylafax! ;-)
cheers,
stoffell
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internal NIC of your small pbx (like in:
laptop?) or by using a switch..
cheers,
stoffell
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So any recommendations for another wireless VOIP phone?
As someone else pointed out, the Siemens C450 IP (and higher models) work great!
Also, the snom m3 gets some good reviews and will be the next one I'll try out..
cheers,
stoffell
Has anyone been able to get ISDN-BRI support to work reliably on
Asterisk 1.4? If so, I'd love to know how you did it (hardware,
distro, kernel, modules, versions, config files).
Maybe your best bet is using bristuff, the bristuff-0.4.0 series are
tests for asterisk 1.4, I haven't tested them
I'm in Europe (yeah, that does matter when choosing a good phone!) ..
Some of my (and my customers') favorites:
- Polycom (pretty much all of them)
- Thomson ST2030
- Siemens Gigaset C450 IP dect (for wireless phones)
cheers,
stoffell
---
http://www.electromarket.be
Maybe you could do a test with :
a; using the latest polycom administration guide (examples found on
voip-info.org) to supply configs and firmware
b; use latest firmware (2.1.0.something)
c; if the issue doesn't go away, try ethereal between a 'misbehaving'
phone and the switch to see what the
I also think you get what you pay for and I don't use hfc based isdn
cards in production any more. Having said that, a small home
installation isn't quite the same as a 30 user office environment. My
home-pbx for example is quite happy reloading asterisk+zaphfc every
night. Of course not
On 9/15/06, Noc Phibee [EMAIL PROTECTED] wrote:
a small question:
what is the best card for Asterisk for supply 2/4 BRI access to a old PABX ?
A good bri card is the quadbri of Junghanns/Beronet or Digium (haven't
tried the Digium one, but seems interesting because of the on-board
echo can..).
Hi list,
Any suggestions on how to deal correctly (socially and technically)
with users complaining about features/issues? For instance, users
complaining about echo; personally I ask the user(s) to give me all
the details when reporting echo (like; using handset/speaker,
internal/external call,
Search Daily Asterisk News for echo:
Yes, that's for the issue with echo, but I was more or less meaning
the social side, the communication with the users.. echo was an
example.. :) (bad choice maybe? :))
cheers
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On 9/13/06, Forum [EMAIL PROTECTED] wrote:
Unfortunately they pointed me back to Polycom and I have not yet heard back
from them.
Can somebody post a link to download sip2.0.1?
If they point you back, report that to Polycom, they'll contact the
reseller (if it's an authorized reseller, that
On 9/13/06, Forum Expansive [EMAIL PROTECTED] wrote:
What is the latest polycom firmware and where can I get it?
sip2.0.1, ask your reseller, they must give it to you.
cheers
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On 9/1/06, Xue Liangliang [EMAIL PROTECTED] wrote:
Hi, all. I am from Singapore, we deployed a few PABX based on Asterisk.
Here in Singapore there are two Teleco providing E1 pri service, we
encountered a strange problem : when calling a number that is
unavailible or line suspended, one of the
to avoid this to make
sure the users experience the same behaviour as before. (with the
'traditional PBX')
I have tried changing the priindication setting (tried
inband,outofband and passthrough) but this didn't change anything.
Does anyone have any idea as to how I can debug this?
cheers,
stoffell
On 9/5/06, Jean-Michel Hiver [EMAIL PROTECTED] wrote:
Have you tried progressinband=yes? As far as understand it, it forwards
early RTP (that is, stuff that is received prior to the ANSWER), which
might just do the trick.
Hm, I have just added this in zapata.conf and sip.conf, and also tried
On 7/20/06, shadowym [EMAIL PROTECTED] wrote:
their Asterisk server? What I would like it to do is use both ethernet
controllers on my motherboard so that if one fails the other one takes over.
I don't see anyway to make it work seamlessly with 2 IP addresses it would
here are some url's to
On 7/7/06, mike [EMAIL PROTECTED] wrote:
someone deployed Junghanns's 0.3.0-PRE-1q (* 1.2.9.1) with a quadBri
card on a production system ?
drawbacks ?
Not really... But be sure to test if you don't have the hangup bug.
(call your cell phone, don't pickup, just hangup as soon as its
ringing,
On 7/6/06, Kevin Savoy [EMAIL PROTECTED] wrote:
I'm having an issue where Asterisk hangs up a call (either party hangs up) but
the telco side
of the T1, both the local company and ATT, does not receive the hangup signal
from
Asterisk. Therefore Asterisk thinks the channel is available but it's
On 5/31/06, Jean-Louis curty [EMAIL PROTECTED] wrote:
I does nothing special,
no output, nor error,
same.. .:-(
you should at least get any output from ztcfg, but aside from that,
like Tommaso said, you must also set the correct jumpers.
cheers
___
-PRE-1q.
I have emailed junghanns.net to let them know.
Best regards,
stoffell
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On 6/29/06, Olivier [EMAIL PROTECTED] wrote:
I have emailed junghanns.net to let them know.
Did they acknowledge the issue ?
I didn't get any reply yet. (but I'm used to that ;))
But yes, the -q release CHANGES file contained this:
- libpri fix for P2P BRI in Belgium
But the bug still
On 6/28/06, Forrest Beck [EMAIL PROTECTED] wrote:
So far we have a
Grandstream 2000
Cisco 7912
Very good phone but not so big display.
Polycom SoundPoint IP
What model? they recently released an alternative to the 501, being a
430. Looks promising.
And we are looking at getting a Linksys
Hi,
I have been using app_sms for a few weeks now, since I recently
upgrade to asterisk 1.2.9.1 (latest bristuff, -q) however, app_sms
doesn't seem to work that well anymore..
On receiving an sms, I execute the app_sms script, and get this as output:
-- Accepting voice call from '171701' to
On 6/29/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
I thought that this was me going mad. I'm trying to use SVN trunk and
have exactly the same problems.
So, I think it's a bug.
Can you confirm sending out works fine?
I send out an SMS without any problem, on receiving however, I have
On 6/1/06, whois wes [EMAIL PROTECTED] wrote:
we're running fedora core 4 with the stock kernel, but a move to either
CentOS, Debian, or just running Pound Key might be in the works.
i currently have the following options added: vga=normal nmi_watchdog=0
acpi=off
i had also tried noapic and
On 5/31/06, whois wes [EMAIL PROTECTED] wrote:
random server lockups, we moved 4 of the servers to Sangoma A104d's last
week. The problems we were having have basically disappeared.
Keep in mind, this is after trying PCI based NIC's and ensuring there were
NO IRQ's being shared, messing with
On 5/30/06, Stefan Reuter [EMAIL PROTECTED] wrote:
Are there any apt repositories which provide newer versions of the
software?
sure: http://pkg-voip.buildserver.net/debian
Hi Stefan, very nice. A related question, is there any way you could
share the process of how to create the asterisk
On 5/18/06, Remco Barende [EMAIL PROTECTED] wrote:
Also the 2850 is *always* sharing IRQ's on every PCI slot, you need to buy
a dual port ethernet adapter which will use only one irq to free up an IRQ
on another slot. This just totally sucks and irq sharing in a box with
only 3 pci slots is
On 5/17/06, Hadley Rich [EMAIL PROTECTED] wrote:
They do, but it isn't released yet. Put B410P into google and you will get a
couple of hits. Digium's marketing page says it is available and the
distributor I use had one on show the other day so they can't be too far
away.
Aside from being
On 5/17/06, Marcel van der Boom [EMAIL PROTECTED] wrote:
We had the exact same problem. It started happening for us starting at
the 'k' release of bristuff (i mailed a msg on it in february i think
to junghanns).
Marcel, thanks. This does seem to work indeed! I just tested this on
our
On 5/16/06, Edu [EMAIL PROTECTED] wrote:
We have an Asterisk Bussines Edition ABE-A.1-6 on a PowerEdge 2850 with 4Gb
RAM. It was working 24/7 without any for a month, but for not related causes I
Just for your info, I have experienced the same issue (just once) on a
Dell PE 2850 also. Same
On 5/15/06, picciuX [EMAIL PROTECTED] wrote:
have you tried EXPLICITLY disabling busydetect? It could cause confusion
on digital (BRI PRI) lines...
If you have busydetect=yes in previuos channel definitions, it will be
inherited by your BRI channels also.
hi, thanks for the tip, but
On 5/11/06, Tim Robinson [EMAIL PROTECTED] wrote:
There is a lot of junk in your zapata.conf that you do not need, as it
relates to analogue lines. This might be causing confusion?
I have tried a similary config to yours, doesn't helps. I haven't got
this problem on an E1, just on the newer
On 5/9/06, Jeroen Zwarts [EMAIL PROTECTED] wrote:
I have a problem with the Bristuffed version of Asterisk. I have tried
Bristuff-0.3.0-pre-1m,n,o,p (Asterisk 1.2.6 to 1.2.7.1) but they all have
the same problem it seems:
Hi Jeroen, any progress made yet? I noticed I'm experiencing the same
On 5/2/06, Wayne Gemmell [EMAIL PROTECTED] wrote:
Opened pseudo zap interface, measuring accuracy...
This may be a stupid question but how did you do this?
in your zaptel source dir (after making..): ./zttest -v
or search for zttest on voip-info.
cheers
On 4/30/06, Remco Barende [EMAIL PROTECTED] wrote:
I e-mailed Dell support and asked them if it is possibel to assign a
unique IRQ to one of the three PCI slots.
Their reply was, not possible, you are ALWAYS sharing IRQ's, I guess this
is the reason for the poor results I'm seeing.
If you're
On 4/29/06, Vidar [EMAIL PROTECTED] wrote:
Has anyone managed to add the bristuff patch to 1.2.7.1 successfully?
My attempts has ended up bad, so if anyone has a working patchfile for
1.2.7.1 I would be grateful to receive it.
Have a look at this URL: http://www.junghanns.net/downloads/
You
hi there,
We just encountered the following.. a customer has a tradifional PBX
that runs next to asterisk. Both PBX's have their own E1 line. Now
'some' numbers are forwarded from the traditional PBX to the new
asterisk server. (both have different DID numbers assigned)
When those numbers are
On 4/24/06, Chris Gamble [EMAIL PROTECTED] wrote:
We are using AAH with Asterisk 1.2.7.1 with a TE405P as listed below. We are
getting frequent restarts on the spans which lead to dropped calls. I have
pasted some hopefully
maybe this is related:
On 4/21/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi listers,
I am looking for people who have used Power over Ethernet switches,
primarily in conjunction with Polycom IP 501's. I've been looking at the
Linksys SRW224P, since I've had good luck with the SRW224 in our office.
On 4/11/06, Begumisa Gerald M [EMAIL PROTECTED] wrote:
Again, if the IO-APIC is reporting that the card is on its own IRQ,
it really, truly, honestly *IS* on its own IRQ. The reason that it
is suggested to disable the IO-APIC is that on many low-end systems,
Allow me to comment
On 4/17/06, Alex Mosburger [EMAIL PROTECTED] wrote:
-) * needs to listen to DTMF tones during the call (for transfers or any
other features)
Does this mean you cannot do any blind or attended transfer? or only
the # transfer option (asterisk built-in, from features.conf) doesn't
work?
cheers
On 4/17/06, Simone [EMAIL PROTECTED] wrote:
look at the wiki and the phones suggested, we'd definitely like phones
with internal ethernet switch and PoE capable, I'll try to get an idea
of what could work for us.
I just have a few suggestions on the phones.. First of all, try using
1 model for
On 4/15/06, Min Hwan Chang [EMAIL PROTECTED] wrote:
wondering if its stable enough to use. Currently I'm editing my own *.conf
Using it at multiple sites (ranging from 10-50 extensions).
scripts but it sure would be nice if there were some sort of web interface
for other people to use. The
On 4/14/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
We have a number of Polycom 501's connected to our * box and they work
great. Some of our users have added a few entries into the directory on
the phone. The problem is on those particular phones they now sometimes
get resource full on
On 4/3/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
as mISDN neighter support fax-protocol nor beeing able two work in NT-Mode
it's no alternative for bristuff :(
I guess http://www.visdn.org/ is an alternative. I haven't used it
yet, but will be looking into it for sure.. (use daily builds)
On 4/2/06, Christian Gröger [EMAIL PROTECTED] wrote:
need that Zaptel stuff? It always prompted errors so i am now using
mISDN -without errors, is there a module for freePBX for mISDN?
to use mISDN with freepbx, you can Add custom trunk in the Trunks menu.
Anyway, is there a good manual for
On 3/29/06, Benoit Panizzon [EMAIL PROTECTED] wrote:
Isn't there an acutal patch to get zaphfc support in *?
You even have 3 possible ways out..
1; you stay with the current bristuff (a somewhat older
zaptel+asterisk, but is this really making a difference?)
2; you use a visdn snapshot
On 3/25/06, Daniel Hazelbaker [EMAIL PROTECTED] wrote:
We are looking at installing a VoIP system with Asterisk and are
currently looking at the line of 3Com phones. Has anybody had
success with using the following phones? We need to buy a lot and we
don't want to end up with phones
On 3/23/06, Henning Holtschneider [EMAIL PROTECTED] wrote:
I've got Asterisk 1.2.4 running with two Junghanns QuadBRI cards using the
qozap driver from bristuff 0.3.0-PRE-1l. One of the cards is running in TE
mode, the other one in NT mode.
Have you (or can you) tried it with 0.3.0-pre1k ?
I
On 3/23/06, Michiel van Baak [EMAIL PROTECTED] wrote:
I've got Asterisk 1.2.4 running with two Junghanns QuadBRI cards using the
qozap driver from bristuff 0.3.0-PRE-1l. One of the cards is running in TE
mode, the other one in NT mode.
Hm, how weird. I'm not experiencing this.
I'm running
On 3/21/06, Dave Cotton [EMAIL PROTECTED] wrote:
span=1,1,3,ccs,ami
I was going mad with a Quadbri until I changed ccs,ami to ccs,hdb3 and
it's been running 6 months now.
Dave, nice to read on this, can you explain what was going wrong when
you used ccs,ami? And how did you find out about
On 3/18/06, Watkins, Bradley [EMAIL PROTECTED] wrote:
cluster (or clusters, in the case of one site). So there is no NAT, and it
is an Asterisk-only solution (at least insofar as telephony software is
concerned).
I'm just barging in.. This all looks 'very' promising stuff, I'm
looking forward
On 3/2/06, mustardman29 [EMAIL PROTECTED] wrote:
Flashybrid looks interesting. I remember reading about the Astlinux
development environment but have not heard much about it lately. Could not
find any links to it anywhere. Now that I have a link I will have to check
that out as well.
Also
hello,
Has anyone got some real life experience with a good software
videophone for windows? (SIP)
On looking in the wiki I think the best looks to be the eyeBeam phone
(commercial), any other suggestions? (free/commercial)
cheers
___
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On 3/1/06, mustardman29 [EMAIL PROTECTED] wrote:
I am aware of Astlinux and the other embedded Asterisk solutions out there?
Astlinux is nice but the problem is that when I hit a snag and need to
incorporate a patch and what not I cannot do that with Astlinux because I
cannot compile my own
On 3/1/06, Kristian Kielhofner [EMAIL PROTECTED] wrote:
Did you try the development environment:
http://mirror.astlinux.org/astlinux-devel.tar.bz2
Kristian, this means I could create a bristuff'ed version of astlinux
by using this one? Cool!
cheers
On 2/26/06, mustardman29 [EMAIL PROTECTED] wrote:
Since I am using two completely different phones it must be my Asterisk
configuration.
I don't know about the aastra, but on the GXP-2000 this is a bug. (do
you run latest firmware? maybe it's fixed in that one)
cheers
On 2/25/06, Chris Bagnall [EMAIL PROTECTED] wrote:
It's a fascinating thread, this.
So, for all the criticism, I'll continue using cheap switches, recycled
Chris, I mostly agree.. In Europe a 'small' business often only counts
2 - 5 persons. When the budget doesn't allow it, the only way one
On 2/22/06, Clint Sharp [EMAIL PROTECTED] wrote:
I had to drop 1.0.1.12 because it has a serious handset volume issue that
seems to cut the handset volume in half. Fix one bug, cause another.
True, but the latest (beta, okay, but does that matter?) firmware
fixes bot and some other. Please
Hi there,
Is it possible with the new aastra firmware to have distinctive ring
support? (the wiki says: There doesn't seem to be any way to have the
server request a distinctive ring.)
The rest of the features make this sound like a good phone. (price/quality)
cheers
On 2/17/06, Gareth Owen [EMAIL PROTECTED] wrote:
No, distinctive ring isn't supported in 1.3.1. You only have the option
of setting the ring-tone on a per-line basis.
hm, okay. is it a feature that will be built-in in the future? or can
you say for sure it will not?
thanks,
cheers
I'm going to propose to my boss the buying 15 Grandstream GXP-2000 phones.
- Is it a good choice (budget limit of 100 Euro/phone is mandatory)?
- Can be a profitable business the direct buying of 50 phones (to save other
money) or is it a risk?
if you've never tried a phone, it's always a
On 2/8/06, Morgan Gilroy [EMAIL PROTECTED] wrote:
As far as I know there will be no difference.
32bit runs natively on AMD64 chips.
The only advantage of 64bit is the extra address space and huge integers
I agree, but.. I have recently installed debian with an em64t kernel
(it was a Xeon 3.0),
On 2/6/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
linksys 942 doesnt look very competetive anyway.
(2 10mbit ethernet ports? who is linksys kidding?)
Ouch, indeed, that's really working against the other features this
phone has. So the advantage over a 941 is also gone now.. :(
Thanks for
On 2/6/06, Conrad Wood [EMAIL PROTECTED] wrote:
Please note the spelling of uniqueid. I find the spelling in
res_features.c - but only once I patched it with bristuff patches.
Does anyone know whether that is a known problem with bristuff? If so is
it fixed in a later version?
What version of
hi there,
I saw a page on voip-info about the thomson ST2030 phone. There is not
so much info on there, that's why I would like to raise a question
here.
Has anyone got hands-on experience with this phone? (with or without
extension module)
I am interested if it can be used (as SIP phone) in a
On 2/3/06, John Jensen [EMAIL PROTECTED] wrote:
Can a normal server with
Pentium 4 3.6 Ghz CPU
Most likely. It'll do 40-50 concurrent 711 to 729 transcodings.
Hm, interesting. In the case that you do PRI (or BRI) to G729. How do
you calculate this number (40-50) ? Or do you write this number
On 2/4/06, Technical Support [EMAIL PROTECTED] wrote:
Is there a web portal available for users to:
destar configures you asterisk, but also has a user-login to change
some user-settings.
http://destar.berlios.de/
cheers
___
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On 2/5/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
We have 10 people on our network and each person will have a SIP phone
connected to our Asterisk server. All phones, Asterisk, other servers and
users workstations will be using the same network. The question is: would
I need a QOS device
On 1/31/06, John Jensen [EMAIL PROTECTED] wrote:
Any recommendations, good/bad expiriences ?
At present I'm looking at cards from BeroNet and Junghanns.
Only have experience with junghanns cards, but they are the same..
beronet doesn't use bristuff.. but you can also use junghanns cards
the
On 1/31/06, Giordano Grandis [EMAIL PROTECTED] wrote:
[chan_sccp.so]Jan 31 10:31:15 WARNING[19727]: loader.c:258
ast_load_resource: /usr/lib/asterisk/modules/chan_sccp.so: undefined symbol:
ast_park_call
Jan 31 10:31:15 WARNING[19727]: loader.c:391 load_modules: Loading module
On 1/30/06, Strain Jer [EMAIL PROTECTED] wrote:
I was curious which one is best suited for asterisks. Thanks
The 'best' depends on your personal flavor I guess.
However I'm impressed by voiceone (http://www.voiceone.it/), didn't
use it yet, but will surely look into it sooner or later.
cheers
On 10/27/05, Erick Baum [EMAIL PROTECTED] wrote:
We're having a rather serious echo problemusing the Grandstream GXP-2000's with Asterisk 1.0.9. I'm wondering if there is something I'm overlooking that might be an easy fix. The echo seems to be worst on internal SIP to SIP calls but you do get it
On 10/24/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Yes I did notice it immediately.I intend to tweak more, but for themoment it seems like echo is minimized to zero.
I also encountered some echo problems and used (uncommented :)) following parameters in zconfig.h:
#define ECHO_CAN_MARK3
On 10/15/05, Lars Dybdahl [EMAIL PROTECTED] wrote:
It seems that the bristuff source from junghanns.com wasn't written
for gcc 4, which is the one included in FC4, and I have seen somedescriptions on making zaphfc compile, but there are more problemsthan just that one. Also, RPMs would reduce the
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