Upgrade to kernel 2.6.9, there are supposed to be significant bugfixes for
CAPI support in 2.6.9.
All of my CAPI systems use FC2, 2.6.9. I tried to go 2.6.10 but had
problems.
Craig
- Original Message -
From: Kib Eki [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
Asterisk (at least stable v1.06) does not seem to support DNIS. - DNIS and
$EXTEN are not the same thing.
We have discovered this recently where we had a block of telephone numbers
in Adelaide diverted to a second block of numbers in Perth. $EXTEN in
asterisk showed the local Perth number rather
Australia could really use a cheap BRI card. As far as the ISDN market is
concerned Australia is a bit of a backwater. The incumbent telco ensures
that it is cheaper to buy multiple BRI's than it is to get a fractional E1.
In this little corner of the world we pay $400 for a single port Fritz!
Sangoma makes a product that is eqiuvalent to the digium PRI boards, which
is one type of product that digium sells. If everyone started buying
Sangomas product over digium its hardly going to cause digium to go under.
The digium pri boards themselves are based on GPL hardware designs, if the
I've got a couple of Fritz! chan_capi installs under my belt here in
Australia. I've elected to use the mISDN capi drivers over the AVM ones and
it works quite well except for broken DID support, and of course all the
limitations of using non Zaptel drivers.
Craig
- Original Message -
[rant]
I wish my local reseller would 'dump' the product, or at least offer it
cheaper without support. The digium PRI cards IMHO are way too expensive
for those of us who are familiar with them and are only interested in
warranty support. I will probably soon be buying another 5 of them and
As an initial troubleshoot, can you preserve the original .tiff file from
rxfax and see if it is being received correctly or corrupted to determine if
the issue is in related to asteriks or somewhere downstream in the fax
processing to email part.
Craig
- Original Message -
From: Chris
Yes the digium cards are relatively cheap compared to traditional telephony
cards. A four port Eicon BRI card costs as much as the digium 4 port E1 so
on a per channel basis (8 vs 120) the digium is very reasonable. Must think
in terms of bang for buck before opening mouth next time.
As for the
This would be a good solution but be aware that at this time the Fritz! may
not handle DID (specifically PTP mode). The AVM drivers will not support
DID. The mISDN drivers and fritz! cards do seem to handle DID but chan_capi
doesn't pass the call to Asterisk (although you can see the call coming
Yes,
And wrote it up in the wiki -
http://www.voip-info.org/tiki-index.php?page=Asterisk%20fax look under the
heafding 'Emailing a fax based on DID'. I used LDAP but it could just as
easily be made to work with odbcget or whatever else you wanted to use.
Craig
- Original Message -
I personally would suggest dumping FC3 and going with FC2 and using either
chan_capi coupled with mISDN and one of the mISDN supported cards or getting
a HFC based card and using bristuff. ISDN4Linux is commonly touted as the
least preferred option.
I'm in the middle of setting up a test config
Just to add my 2c, I'm got three production boxes all using FC2, two with
kernel 2.6.5 and one using 2.6.9, One is a 1RU HP DL320 with a TE410P, one
is Dell 2850 2RU with Eicon Diva Server 4-BRI, mISDN and chan_capi and the
last is SIP only running on an old desktop Celeron 900 (intel i810
Is that a CVS-head thing? I have setup 1.0.3 today running as non-root
asterisk user on FC2 no problems.
Craig
- Original Message -
From: Andrew McRory [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, January 12, 2005 9:10 PM
Subject: Re: [Asterisk-Users] What is
I am running an HP DL320 1RU on FC2 kernel 2.6.5 with no problems - Card is
the digium quad port T1/E1 (3.3volt). This chassis only has a single PCI
slot.
Craig
- Original Message -
From: Michael Swan [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, January 14, 2005
I've got the following situation where a UA is trying to call another UA via
Asterisk and SER according to UA1 - * - SER - UA2. Now in the event that
SER generates a 404 Not Found for UA2 I would like Asterisk to return or
relay or forward or whatever the 404 to UA1. Anyone know this might be
1.0.5.16 breaks the messages (Voicemail) button. I'm messing with 1.0.5.20
and it appears to be ok so far, however there doesn't appear to be any way
to downgrade from 1.0.5.20 as it is ignoring any earlier firmwares - at
least using tftp anyway.
Craig
- Original Message -
From: David
Theres a couple of ways -
Check to see if your bank really requires you to press pound. Mine says to
press it, but all pins are fixed length so it may time out after a second or
two.
Alternatively put a regex in your dialplan to recognise the phone banking
and bill payment numbers and call the
It sounds like you have multiple devices sharing the same physical lines? I
think you will continue to have problems until you can rearrange the setup
to avoid line sharing to allow Asterisk to have dedicated access. Might
have more luck with ISDN.
Craig
- Original Message -
From: Jon
That URL has been locked down for resellers and vendors only for a couple of
days now. Pity, one of the good things about the Grandstream was their
freely available firmwares. Oh well, time to find another phone - the
Sipura 841 is looking interesting.
Craig
- Original Message -
From:
: [Asterisk-Users] Asterisk with Grandstream ringback
On Monday 31 January 2005 14:28, Craig Guy wrote:
That URL has been locked down for resellers and vendors only for a
couple
of days now. Pity, one of the good things about the Grandstream was
their
freely available firmwares. Oh well, time
set 'DTMF_inband: 1' in your SIPDefault.cnf to have your voicemail work.
Craig
- Original Message -
From: Derek Conniffe [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, February 03, 2005 5:41 PM
Subject: Re:
I honestly can't understand what all the confusion is about.
There are two versions of Asterisk, CVS-Head and Stable. Head has no
version numbers, it seems to be delineated by date.
If you download cvs v1-0 then you will always get the current release of
stable whether it be 1.0.5, 1.0.6. or
I was browsing through the web config of a Sipura SPA-841 (Firmware 2.0.13)
and noticed a setting marked 'paging' under supplementary services on the
Phone settings page on the advanced admin login. Anyone know how it might
be used? Could it be like the Snom -
exten =
I might have a similar issue -
Using Fedora Core 2 with Kernel 2.6.9, Asterisk 1.0.3 Stable, chan_capi
0.3.5, AVM Fritz! PCI card v2.0 and mISDN
mISDN loads ok, CAPI loads ok, Asterisk sees the b channels as available but
is unable to place any outgoing calls. With capi debug enabled the error
Or you could go to a 2.6 kernel and use the mISDN drivers.
Craig
- Original Message -
From: Razza [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, February 20, 2005 8:00 PM
Subject: [Asterisk-Users] Mandrake CAPI
All,
I have been trying to get CAPI4Linux working
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy
Sent: 20 February 2005 23:57
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Mandrake CAPI
Or you could go to a 2.6 kernel and use the mISDN drivers.
Craig
- Original Message
If you use the mISDN Fritz! driver with CAPI you should be able to use up to
4 Fritz! cards. I have it working with one card but have not tried four.
Craig
- Original Message -
From: Brett, Gary [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
I set dtmfmode=inband for my 7960 in order for voicemail to work.
Craig
- Original Message -
From: Mark Johnson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, March 01, 2005 11:20 PM
Subject: Re:
I use a startup script with nothing in modprobe.conf:
#!/bin/bash
#
# System startup script for the isdn-capi subsystem
case $1 in
start)
echo -n Starting mISDN and CAPI
modprobe capi
modprobe mISDN_core
modprobe mISDN_l1
modprobe mISDN_l2
modprobe l3udss1
modprobe mISDN_capi
Maybe a long shot but if you run asterisk as non root user have you checked
the permissions on /dev/capi20 ? I have an eicon 4BRI card and every time I
reconfigure the card with divas_cfg I do a chown --recursive
asterisk:asterisk /dev/capi2*
Craig
- Original Message -
From: Junk Mail
I've been using 1.0.5.10 on 25 phones since August and I've only had to
reboot 2 phones the entire time.
Craig
- Original Message -
From: Vahan Yerkanian [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Monday, November 15, 2004 3:51
Running CVS Head 10 August on Kernel 2.6 (FC2) as non-root user with no
issues. I followed the instructions on the wiki for running as non root.
For startup and shutdown I've just copied the digium supplied scripts into
/etc/rc.d/init.d and created the appropriate links into /etc/rc.d/rc3.d as
About once an hour the phone displays '403' on the display for about 10
seconds or so with this firmware. There is no corresponding entry on the *
console. 'Spose it has something to do with registration. Apart from that
it looks ok so far and the web interface now looks much better.
Craig
Yes, 1.0.5.16 breaks the message button. 1.0.5.18 fixes the message button
but shows a 403 error 'bout once an hour.
Craig
- Original Message -
From: Gilad Ben-Yossef [EMAIL PROTECTED]
To: Michael Nolan [EMAIL PROTECTED]; Asterisk Users Mailing
List - Non-Commercial Discussion [EMAIL
I had exactly this problem when I did it. It would seem that either of the
perlscript that constructs the mime attachment or Outlook (Express or XP) is
not RFC compliant. I found the mime headers had to be modified for it to
work, Outlook wasn't reading, or was ignoring the filename part of the
Is anyone aware of a HFC-S card available in Australia with A-tick approval?
Current choices appear to be the traverse NetJet (ISDN4Linux only) and the
AVM Fritz (CAPI). However I would strongly prefer to use something ZAP
compatible with zaptel BRI and hence an HFC-S based card.
Craig
card for Australia?
Does the E100P (from digium) fit the bill, apart from the lack of A-Tick?
PaulH
-Original Message-
From: Craig Guy [mailto:[EMAIL PROTECTED]
Sent: Wednesday, 1 December 2004 2:59 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] HFC-S card for Australia
I second that. The current sccp implementations for Asterisk are highly
unstable and lacking in functionality. Use the SIP images instead. If you
need _must_ have a hardware operators panel/console thingy then get a SNOM.
Craig
- Original Message -
From: Walid Azab [EMAIL PROTECTED]
You need firmware 1.0.5.16 (Broken message button for voicemail) or 1.0.5.18
(Still in Beta, phone display '403' error about once per hour for 10 seconds
or so. In order to use attended transfer you place the caller on hold by
pressing the flash button and then dial the third person. Once you
transfer to work with the BT-100 on 1.0.5.16.
Where
did you get 1.0.5.18? It's not anywhere obvious on Grandstream's web site.
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy
Sent: Thursday, December 09, 2004 02:56
To: [EMAIL
Austech Partnerships (www.atp.org.au) I believe are the A-tick holders for
digium hardware in Australia. They have told me previously that digium
hardware not supplied by them is not approved for connection in Australia.
Even then the only approved hardware currently are the quad-port PRI cards.
If you are using an Athlon then you might have a VIA chipset and apparently
non-intel chipsets can have these sorts of interrupt problems (Via
especially). Try changing to an intel chipset motherboard.
Craig
- Original Message -
From: Michael Welter [EMAIL PROTECTED]
To: Asterisk Users
Telstra Onramp use a 9 digit callerid. I use the following macro to
'correct' incoming callerid so that numbers are in the proper format to
callback from the callerid memory of the phone:
exten = s,1,GotoIf($[${ARG1} = ]?100:2);Check for
null callerid and jump to 100 if so,
What you can do is retain your internal callerid information in sip.conf to
allow your voicemail to work correctly and have internal callerid show
extension rather than full external number. In your dialplan you then have
a line pre-pending your external prefix to outgoing calls so that callerid
Hi Jesse,
I would strongly recommend changing over to the SIP image and uisng
something like the Flash Operators Panel (www.asternic.org) instead of the
7914's. I experimented with chan_sccp2 a few weeks ago and decided that it
wasn't for me right now due to both the very limited support for the
Looked at one of these phones about a month ago (the ATA 323, haven't seen
the ATA 723), the base unit is very light and the rubber stoppers are crap
so the phone slides across the desk whenever you pickup the handset. There
was no visual MWI (message waiting indicator) on the phone and the
Sounds like you'll need a TE410p (Austel approved) or an E100p (non Austel
approved). Which provide 4 or 1 E1/T1 interfaces respectively. Depending
on your number of internal extensions and need for call queues etc one
server running Asterisk could handle everything. We currently have an
From what I have seen so far on this list if you are running a version of
CVS-Head prior to release of Asterisk 1.0 then you should keep it and not
try to change or upgrade it. It would appear that there are a lot of recent
changes that may break if you try to upgrade to current CVS-Head, and
The answer will be found in how you have setup your contexts. You will need
to specify a default context in your sip.conf or whatever the mysql table
equivalent is and then handle calls from this context appropriately in your
extensions.conf, eg (Very simplistic but you get the idea).
sip.conf
Depends on what you mean by firewall. If both ends are behind a NAT type
router then you shouldn't have any problems provided you configure 'nat=yes'
and 'qualify=yes' in Asterisk, and the appropriate STUN settings in the
phone. The qualify is important as it keeps a 'session' open in the NAT
There is currently no such feature on the BT100 although someone did post
two weeks or so ago that firmware 1.0.5.12 would have it. As yet, there is
no hint of this new firmware. Alternately I think there is a patch around
somewhere to do it within Asterisk, play detective and see if you can
If it has a spare PRI port then build up an * server with an E100 card and
connect using an ISDN crossover cable. If the Samsung can support analog
trunks you could stick in a TDM400 with a couple of FXO ports.
Craig
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Hi,
I had my PRI dropout this afternoon after logging the errors below - Can't
seem to find much info on what it might be or what caused it. Whilst
waiting for the telco tech to call I took the server down and restarted it
and it all came back good,but that may just have been coincidence so I
Don't be lazy, check the bug reports for this application - wander over to
https://sourceforge.net/tracker/index.php?func=detailaid=1049761group_id=106482atid=644546
It is a known issue with build 0.04
Craig
- Original Message -
From: Rana Dutt [EMAIL PROTECTED]
To: Asterisk Users List
I haven't really looked into ASTWind too much but I assume there would be
network access available to it? It might be useful for the purpose of
providing limited PBX services and acting as a gateway to trunk 4 or 5 SIP
phones via IAX across a WAN link to an ITSP, central office or some such.
PROTECTED] On Behalf Of Craig Guy
Sent: Monday, November 01, 2004 5:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Linux and Windows
I haven't really looked into ASTWind too much but I assume there would be
network access available
I've run up to 50 concurrent calls on the PE850 and PE860 using TE205p.
I also came across the te110p issue which manifests itself as popping and
crackling audio. It is rather insidious as zttest is fine, the problem does
not appear to be missed interrupts. In my case the Digium distributor
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall
Sent: Monday, 24 September 2007 6:15 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Anyone use the Linksys phones?
Is anyone out there using any
The Linksys SPA962 with SPA932 sidecar support both speed dial and BLF.
IMHO very good for the money and very easy to provision once you get a hold
of the proper provisioning guide. These things are designed for mass
deployment and remote provisioning. As other people have noted, you need to
I have successfully used the Grandstream ATA286 and Linksys PAP2NA. I would
recommend the Grandstream over the Linksys as there is less configuration to
do and it is IMHO more reliable for faxes. I have been able to get analog
data modem connect at 48k on the grandstream whilst cannot get
Yes, I have compiled and used rxfax succesfully on 1.0.9, 1.0.10 and 1.2.4
to receive from analog fax machines. I have never yet been able to get
rxfax working with txfax - my debugs when I try look like the logs in your
email.
Craig
- Original Message -
From: Jesse Guardiani
@lists.digium.com
Sent: Monday, February 20, 2006 12:20 PM
Subject: [Asterisk-Users] Re: spandsp 0.0.2pre25
Craig Guy cguy at bigpond.net.au writes:
Yes, I have compiled and used rxfax succesfully on 1.0.9, 1.0.10 and
1.2.4
to receive from analog fax machines. I have never yet been able to get
In the 1.0.x branch asterisk does not always send SIGHUP to agi scripts on
call hangup. In 1.2.x a SIGHUP is always sent, even using DEADAGI - From
the UPGRADE.txt in the source:
AGI:
* AGI scripts did not always get SIGHUP at the end, previously. That
behavior has been fixed. If you do
I have been involved with a BRI install using 3 x Draytek minivigor 128 BRI
adapters and chan_mISDN. The draytek units use the HFCS-USB chipset, are
USB and take power from the USB interface. Each adapter will support PTP,
PTMP, TE and I think NT mode with a maximum of 8 adapters (16
We didn't ask specifically for new ones. I believe the old ones went out of
stock a long time ago. We ordered four at once and they all came with the
HFC chipset.
Craig
- Original Message -
From: James Harper [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
I believe that IAXVAR in Asterisk 1.6 will do what you want. I have a
backport of this for Asterisk 1.2.14 or so floating around somewhere but it
hasn't been maintained or used for months, may not be compatible with the
1.6 implementation and I offer it with no support whatsoever.
Craig
On 4/25/08, Tobias Ahlander [EMAIL PROTECTED] wrote:
Date: Thu, 24 Apr 2008 06:54:27 -0700 (PDT)
From: Steve Edwards [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Forking in Dialplan
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
The Hylafax / Iaxmodem is a good, reliable combination. I have work with a
company that competes with eFax using the Hylafax / Iaxmodem combination for
termination and also soon for origination.
Craig
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Not necessarily - if you set your iaxmodems to only produce G4 encoded tiffs
you can then use something like c42pdf http://c42pdf.ffii.org/ which
essentially copies the tiff image data into a pdf container. Lightning
fast, quality is preserved, very little memory usage and very little cpu. I
Hi,
Was wondering if anyone had any tips or experience in getting a Nortel CS1K
and Asterisk 1.4.19 to talk to each other via NRS? So far I've gotten
asterisk to place calls to the CS1k via the NRS, however calls originated by
the CS1K get rejected by the NRS with a 404 Not Found message. If
Hi,
Does anyone know if it is possible to integrate Asterisk CDR's with
PhoneControl software? (www.phonecontrol.com). I think it should be
possible, but haven't been able to find any reference to it being done (or
even that it can't be done).
Craig
The FSV-4PFS as shipped will not switch Ethernet - it switches pins 1,2,4,5.
Craig
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of FailSafe Inc.
Sent: Tuesday, 2 September 2008 11:27 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] PRI Splitter
2008 7:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PRI Splitter
Hy Craig,
Can you elaborate on that? In our setup we have it doing just that and
it works without a glitch.
Regards,
Igor H.
Craig Guy wrote:
The FSV-4PFS as shipped
I got my dCAP by turning up to the exam at Astricon in Madrid a couple years
ago without doing any training. It may have changed since then but I found
that the practical exam would be difficult if not impossible to pass without
knowing what you were doing - either through real world experience
On Sep 21, 2008, at 9:15 AM, Tilghman Lesher wrote:
On Thursday 18 September 2008 20:56:58 Craig Guy wrote:
I felt at the time the written portion was heavily biased towards
people
who had done the training - in fact I would go so far as to say
that it was
designed specifically
That is not true regarding voice / fax detection with iaxmodem. If you are
running zaptel, then let it do the fax detection and have the iaxmodems
called from the fax context.
Craig
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mail-lists
Sent:
I haven't used the iaxmodem / hylafax combo for sending, only for receiving.
However my experience is that it is 99% reliable. I am using a Dell
PowerEdge 850 with a Pentium 2.8Ghz and 512mb ram. I think it is the
Pentium D but could be the dual core, not sure, whatever the base cpu was at
the
G729 and annex A differ in the perceived quality and cpu requirements. The
annex A version requires less CPU at the cost of loss of quality. The
bitstreams are compatible with each other in that a G729A codec can decode a
G729 stream and vice versa.
Craig
-Original Message-
From:
Which H.323 channel driver are you using, and could you post a log or debug
of a session.
Craig
- Original Message -
From: Andrei U [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, February 08, 2007 2:41 AM
Subject: [asterisk-users] H323 to SIP - One way voice
It's not that Digium don't want fax or t.38 support, it's just that it is
not very likely for Steve Underwood to provide it for Asterisk. I'm sure
that Digium are very keen for someone to write and contribute t.38 code for
Asterisk, it's just that there aren't very many people with the
Hi Richard,
there was a thread regarding this a while ago on the dev list which resulted
in a patch being made to allow variable passing via IAX2 channels. See
http://bugs.digium.com/view.php?id=7619 for the patch which I think is in
SVN or anyhow, is not in 1.2
I have recently backported
Hi,
Has anyone had any experience integrating Asterisk 1.4 with PhoneControl
call accounting software ( www.phonecontrol.com.au )
Apparently the s/w does SMDI on serial interface and IP collection. Looking
at SMDI in Asterisk I don't think that method will work for SIP calls.
Craig
The lcd in the current budgetone series cannot support alphnumeric display.
Craig
- Original Message -
From: Ricardo Carvalho [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, September 12, 2006 8:11 PM
spandsp supports 9600 rx and does not support ecm. If you want ecm, use
iaxmodem with hylafax - http://iaxmodem.sourceforge.net , currently hylafax
in conjunction with iaxmodem seems to be more reliable than rxfax and
spandsp by themselves.
Craig
- Original Message -
From: Artifex
as a very
bizarre way to me. :-)
bye,
Zsolt
On 9/13/06, Craig Guy [EMAIL PROTECTED] wrote:
spandsp supports 9600 rx and does not support ecm. If you want ecm, use
iaxmodem with hylafax - http://iaxmodem.sourceforge.net , currently
hylafax
in conjunction with iaxmodem seems to be more reliable than
Try this one:
http://www.soft-switch.org/downloads/snapshots/spandsp/
- Original Message -
From: Artifex Maximus [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 13, 2006 11:33 PM
Subject: Re:
I'm interested, too in how to accomplish this. I have tried earlier today
with a Snom360 to register it using its mac address as the authentication
username. I can't seem to get it to work (hopefully I'm just doing
something wrong).
My sip.conf (asterisk 1.2.12) looks something like:
] Re: Can you explain why multiple
registrationisan important (missing) feature ?
On 9/20/06, Craig Guy [EMAIL PROTECTED] wrote:
[9580]
type=peer
auth=000413242fff:[EMAIL PROTECTED]
It would be
[MAC ADDRESS]
type=peer
...etc..
Or at least, that's how I interpreted what Eric said. I think
I had an occasional PCI parity error on a TE405p on an HP DL320. Turned out
to be grease or some similar substance on the edge connector of the PCI
riser to the mainboard in the server presumably from the manufacturing
process. Had been bugging me for months until I finally tracked it down.
I have a similar or maybe the same issue. I have found no solution for it
but to just ignore the new hardware prompt. I saw someone say in an earlier
email that the pci subvendor id seems to change at random on reboot, causing
the card to be redetected. I also find that if the card is connected
I have found my PAP2-NA's to be picky about their DHCP server. My PAP2-NA
appears to lockup if it is set for DHCP and the server is my Netgear RP614
Websafe router. The fix for this is to unplug the ethernet from the unit,
plug in an analog handset, power it on, using the handset perform a
I'm sure its been brought up previously on the list but I personally don't
think that TTS is very practical due mostly to all the crap that gets
stuffed into emails these days. How do you handle:
RTF
HTML
Disclaimers
Signatures (inc ascii art sigs)
Virus scanner tags
I have about five Poweredge 750's and have used both TE110p's and new and
old firmware TE410p. I found the TE405p to be very crackly. My Poweredges
have both PCI slots populated - A DRAC4 card in the 5v slot and either
TE110p or TE410p in the 3.3v slot. The DRAC card consumes 3 interrupts all
No, though I have been looking for one for ages - if you find one let me
know.
Craig
- Original Message -
From: Stephen Allan (External account) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 25, 2005
obviously have has experience with chan_mISDN in AU and the Fritz.
Have you tried chan_capi? I am currently using a Fritz with chan_capi
in AU and am not entirely happy with it. Is chan_mISDN any better?
On 7/27/05, Craig Guy [EMAIL PROTECTED] wrote:
The mISDN Fritz! driver supports PTP mode
I've been using spandsp and rxfax to receive faxes for a while now and over
the past few days I've looked into the other side of things - txfax. I
can't seem to get it working properly. I've included debug logs below of
both the tx and rx side of things. I've tried three different servers,
Hi Bartek, I posted the exact same problem last week - I found that if I
connected two Asterisk systems together via a PRI crossover cable and talk
txfax to rxfax then you get a T4 state timeout. I tried connecting ports
one and two together on a TE410p and also connecting a TE410p to a
Hi Michael,
What phones are you using as this will affect your implementation. For
example do you want to dial zero, then hear a dialtone and dial the full
number or do you wish to dial the whole number with a preceeding zero in one
hit?
Craig
- Original Message -
From: Michael
Hi Tommy,
have you seen the Asterisk @ Home distribution? IMHO the easiest way to
install AMP.
Craig
- Original Message -
From: Tommy Denton [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, August 22, 2005 10:07 AM
Subject: [Asterisk-Users] perl-cpan
Dear
Hi rootlinux,
I'm in Australia where we also uses crc4 on the span line, could you also
show the relevant section of your zapata.conf? Looking at your
extensions.conf excerpt, it is customary to group the b channels eg
Dial(Zap/g1/12345678) and there should be an entry in zapata.conf under
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