The auto-dial dial plan command works on the ATA's as well (At least I got
it working on the Linksys PAP2-NA) based on the Sipura ATA user guide.
Insert a dialplan entry in the phone of s0:123 which causes the phone /
ATA to dial 123 as soon as it goes offhook.
Craig
- Original Message -
Asterisk (or at least Asterisk - Stable) will not present DNIS to the CLI.
If the number originally dialled is redirected then you will see the
redirected number presented rather than the original one. The original
number appears to be present in libpri but doesn't seem to make it into
asterisk
Sendmail isn't really that hard to configure for simple stuff like this.
Most Linux distros have /etc/mail/sendmail.mc, so set your smart relay host
and the appropriate masquerading options - the options for these are spelt
out in the sendmail.mc file. If you want to receive bounces then also set
tried sending thru a 15 page fax. All 15 pages were received in the tiff
image, but every 2 or 3 pages, it would seem as if the image skipped an
inch. So instead of being 8.5 x 11, it was 8.5 x 10 (or 9).
-Matthew
From: Craig Guy [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non
I agree with Steve on this, am piloting Spandsp 0.0.2pre15 on asterisk 1.0.7
with a TE405p, euroisdn. Fedora Core 2, kernel 2.6.9. Running on an old
Dell Optiplex desktop PIII 450mhz with 256mb ram. Takes on average 350
faxes / day with just under 1% failed faxes. I define a failed fax as one
There is a bug with safe_asterisk and FC2, you must edit the script to
remove 'daemon' from the the startup command and then it will auto restart.
Craig
- Original Message -
From: David Phelan [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
You could try http://www.inter7.com/?page=astfax - I haven't used it yet
myself but it looks like it'll work.
Craig
- Original Message -
From: Anton Krall [EMAIL PROTECTED]
To: 'Justin Newman' [EMAIL PROTECTED]; 'Asterisk Users Mailing
List - Non-Commercial Discussion'
In your asterisk script in init.d that calls safe_asterisk change this:
start() {
# Start daemons.
echo -n $Starting asterisk:
if [ -f $SAFE_ASTERISK ] ; then
DAEMON=$SAFE_ASTERISK
fi
if [ $AST_USER ] ; then
ASTARGS=-U
I haven't seen any of that behaviour on my system. If it makes any
difference I run asterisk as a non root user.
Craig
- Original Message -
From: Guido Hecken [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday,
Can anyone with a TE110p confirm that it will fit and work in both a 3.3 and
5 volt pci slot? From photos it looks to be a universal card but the digium
literature makes no mention of voltage requirements.
Craig
___
Asterisk-Users mailing list
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
[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
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 -
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
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
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:
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 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
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:
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
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
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
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 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
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:
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
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
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
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
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
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
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
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,
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
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
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]
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
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
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
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
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
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
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
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]
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
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
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
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
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
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
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
Just some more information regarding the 7914 addon for the 7960 phone. The
7914 requires upgraded firmware to be able to work with a 7960 of firmware
5.x or above. Do not upgrade the firmware of your 7960 above 5.x until you
have done your 7914 first as you cannot downgrade the Cisco to a pre
Do you have early dial enabled at all?
Craig
- Original Message -
From: Eric Jacksch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Monday, September 06, 2004 5:16 AM
Subject: RE: [Asterisk-Users] Wildcards and variable number of
chan_skinny refers to support for the cisco Skinny Client Control Protocol
which is used by Cisco IP phones in a Cisco Call manager environment. It
sounds like the PIX is forwarding stuff to port 2000 on your asterisk box.
If you are not using SCCP then you can prevent the module loading by
From my observation, if a call cannot be successfully placed then execution
goes to n+101. So for example if a phone is busy then the call can't be
placed (channel can't be created) and you jump tp n+101 which is typically
voicemail busy. In the case of a phone being offline then the call cannot
I have now today also configured a 7960 to work with asterisk via chan_sccp.
I have only used SCCP firmware 5.0 (5) with the 7960 and I gotta say that I
much prefer the SIP 7.2 firmware. The real reason at this stage for going
SCCP is for support of the 7914 expansion module. This phone will be
We are using a Netgear FSM7326P to PoE a 7960 (with 7914 attached).
Craig
- Original Message -
From: Michael Welter [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, August 29, 2004 2:13 AM
Subject: Re: [Asterisk-Users] POE
I'm in the process of doing this over the next few days. I'm currently
running Asterisk and about 25 Grandstreams but the receptionist has just
gone and bought a Cisco 7960 with a 7914 operators console and so I thought
I'd have a crack at running it using chan_sccp2 and hoping that 7914 support
Hi Steven,
We have just built an Asterisk 1U server using a HP DL320 and a TE410p card.
Is working well, however we were caught out when it arrived without the
combo floppy/cdrom which is an expensive 'option'. We ended up installing
FC2 via PXE. Is very very noisy, even with fans set to 'low'.
California London England
www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy
Sent: Tuesday, August 24, 2004 7:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dell PowerEdge 750
Hi, I was wondering how other people might handle this situation. We have a
10 channel fractional E1 and have so far only allocated about 25 numbers out
of a 100 number block (9500 - 9599). If an external caller dials an invalid
extension then we would like for them to be put through to our main
Hi James,
This is a feature that needs to be enabled on both the phones and on
Asterisk. So after enabling on your BT100 you need to add
'cancallforward=yes' to each extension in sip.conf you would like to add
this feature to as in :-
[9500]
context=internal
type=friend
username=9500
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Craig Guy
Sent: Friday, August 20, 2004 12:36 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
Hi James,
This is a feature that needs to be enabled on both the phones and on
Asterisk. So after
Hi Will,
That won't work (I tried it originally). What happens is that an invalid
extension will match the _123495XX pattern and asterisk will try to dial it.
My understanding is that the i extension will only activate if no pattern
match can be found, eg if somehow 9[0-4,6-9]XX gets into the
I have had success with this using both the X100p (wcfxs and wcfxo) and
TE410p (wct4xxp) under Redhat FC2 2.6.5. The instructions are on the wiki,
do the following:
ln -s /lib/modules/2.6.5-1.358/build linux-2.6
cd zaptel
make clean
make linux26
make install
Having said that I have found the
everything
else has been bedded in.
Craig
- Original Message -
From: Craig Guy [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, August 14, 2004 10:21 AM
Subject: Re: [Asterisk-Users] te410p and Telstra Onramp 10
Thankyou very much for that :) I assume you would be using DNIS to match
Hi,
now that these cards have approval in Australia, has anyone had any luck in
connecting them to a Telstra Onramp 10 service? Are these configured as a
PRI, bchan=1-10, dchan=16 in zaptel.conf and switchtype=euroisdn in
zapata.conf? Better yet anyone have any of the above .conf files they
, August 14, 2004 6:04 AM
Subject: Re: [Asterisk-Users] te410p and Telstra Onramp 10
On Friday 13 August 2004 22:06, Craig Guy wrote:
Hi,
Is an onramp 10 what is referred to as a 'channel bank'?
A channel bank is a device that would take the onramp 10 in one side a
present
10 separate PSTN
14, 2004 9:36 AM
Subject: Re: [Asterisk-Users] te410p and Telstra Onramp 10
On Fri, 2004-08-13 at 22:06, Craig Guy wrote:
Hi,
now that these cards have approval in Australia, has anyone had any luck
in
connecting them to a Telstra Onramp 10 service? Are these configured as
a
PRI, bchan
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