On 04.01.2012 07:25, Matt Darnell wrote:
We are looking to roll a solution that will have the following network layout:
ISDN-PRI -- Asterisk -- T.38 -- ATA -- Fax
Does version 1.8 with the Digium fax driver have this capability? I
like 1.8 because it is a long term support version.
Hi,
i'm trying to periodically playback a sound to an existing conference
with ConfBridge on Asterisk 10.0.0-rc3
Previously with MeetMe I generated a callout file and had an matching
local dialplan entry.
But this does not work... The local channel gets joined to the
conference, is stuck there
) solution?
am i the only one with this problem (haven't found anything about this
on the mailing list)
thanks
frank sautter
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Alex Ongena wrote:
I have a running asterisk 1.2.1 (bristuffed) with a TDM400 Board. Connected
to 1 port is am ordanary Fax Machine. Everything 'seems' to work,
however receiving faxes is very unreliable.
hello tomislav,
Tomislav Parcina wrote:
Does Asterisk support Advice of Charge? I was told that my Telco sends
me billing signalization that way, and I wonder can I use it?
I have found out that this is part of EURO ISDN standard. q.956 - Advice
Of Charge. Does anybody know how to implement
hello bastian,
you could use the patch i made http://bugs.digium.com/view.php?id=5014
frank
Bastian Schern schrieb:
I'm using the snom Phones together with Asterisk and I already able to
see which Peer is used via hint priority. Then a LED on the snom phone
is blinking. But I don't see who
Jason Pyeron schrieb:
On Wed, 9 Nov 2005, Olle E. Johansson wrote:
That is not supported yet. There is a patch in the issue tracker that
does this, but it's a proof-of-concept code. It will burden your
asterisk quite a lot if you put it to use in larger production sites.
Which issue are you
Olle E. Johansson wrote:
We really need test input of the latest patch in this issue report. And
we need them today. If you are interested in device state notification
in SIP - stop whatever you are doing and give us feedback NOW!
Thank you for your assistance!
Paul Brock schrieb:
Trying to set up these two buttons on a snom 360. The message waiting
key seems to send a call to it's own number, which is obviously engaged
and where you are prompted to leave another message to yourself, and the
conference key seems to do nothing.
this should no longer
Christian Wengel schrieb:
But now I have another problem.
The LEDs on the snoms are blinking now, if the extension is ringing. But
I can't pickup the call by hitting the blinking button.
this problem is not solved by only applying patch #3644.
but as you are not the first one asking for
hi christian,
Christian Wengel schrieb:
But the latest patch sipsubscribe-20050812.rev806v2.txt from
http://bugs.digium.com/view.php?id=3644 didn't worked,
maybe you like to try the latest patch i created a view hours ago...
sipsubscribe-20050823.rev813.txt on
hi,
after stumbling over the compile time flag in zaptel and after reading
the new features of the 2nd generation firmware of the TE405P/TE410P, i
was wondering if the cards are capable of upgrading the firmware in field?
regards
frank
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Matthew Boehm wrote:
Since you are using ODBC, this seems more likely to be an ODBC issue. If
you are concerned, you should just use the native MySQL RealTime driver.
It does not exibit the behavior you mentioned.
Frank Sautter wrote:
our asterisk is configured to retrieve sippeers
our asterisk is configured to retrieve sippeers and iaxpeers via odbc
from a mysql database. after each call show processlist; within the
mysql console shows 2 more persistent connections which are showing no
further activity and will not go away even after restaring asterisk.
is anybody else
Maik Schmitt schrieb:
one of our customers which wants a soft transfer between his old pbx to
asterisk and sip. the setup is as follows:
telco ---pri--- asterisk ---pri--- legacy pbx
everything is fine exept that when dialling from the legacy pbx it takes
about 3 seconds before the asterisk
hello everybody,
one of our customers which wants a soft transfer between his old pbx to
asterisk and sip. the setup is as follows:
telco ---pri--- asterisk ---pri--- legacy pbx
everything is fine exept that when dialling from the legacy pbx it takes
about 3 seconds before the asterisk
Tom Hayden wrote:
I've been using the extension lights on my polycoms before that patch,
so I'm not sure what it fixed, but I've only seen the lights work on
Polycoms and Snoms. Try using the hint priority and see if it works
for your gxp2000, be sure to post your results!
this gives you
hi listmembers,
please test my new patch to chan_sip.c which is to make call pickup on
the snom phones (and maybe other phones that support 'INVITE/Replaces')
work and make comments in the bugtracker
http://bugs.digium.com/view.php?id=3644 so it can make its way into the cvs.
this patch
hi,
is there anybody who was able to setup call pickup with a snom phone?
searching through the web brought up this:
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+snom
section call pickup
but this doesen't seem to work with current releases of the snom
firmware (and looking
know of a pager base station with an EuroISDN interface?
what's your general advice on those paging systems?
regards
frank sautter
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signalling protocoll will be used on the T1 side? is asterisk
translating this correctly?
- btw. where is the different bitrate coming from? is it 7bit T1 and
8bit E1 or 7kHz and 8kHz sample rate?
regards
frank sautter
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hi patrick,
Patrick schrieb:
Did you try contacting the vendor of the base stations to see if they
have a EuroISDN firmware update? My Eicon Diva Server BRI card supports
the 1TR6 protocol. The firmware can be found here:
ftp://ftp.isdn4linux.org/pub/isdn4linux/utils/eicon/firmware/
Perhaps AVM
, so I need to
learn more here. Maybe some else can inlight me here...
chan_capi currently supports receiving and sending of faxes utilizing
the onboard DSPs of the eicon cards.
please look for the neccessary patches at:
http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2
regards
frank
klaus-peter he has restarted to improve
chan_capi (i thought he lost interest in chan_capi and concentrated only
on his bri cards).
i hope klaus-peter will include the fax support into chan_capi-0.4.0!
So the CAPI on kernel 2.6 problem is on top now...
fine.
freundliche grüße
frank sautter
as if this are interesting features.
maybe you could take a look on the patches of carl sempla and cedrik
hans (faxing with eicon cards) and mine (transfer capability, limitation
of MSNs, cvs-head) both available using:
http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2
grüße
frank sautter
hello jairo,
Jairo Buendia wrote:
-- B-channel 0/1 succesfully restarted on span 1
unused b-channels are reset by asterisk every hour (default).
you can set the interval to another value in your
/etc/asterisk/zapata.conf
resetinterval=86400 ; e.g. reset every 24hours
or even longer.
i think
cable is
only a few meters and does not have any other devices on the bus.
regards
frank sautter
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hi gary,
Brett, Gary wrote:
(can I just
use a single cat5 straight through cable between them ?? and cant the Digium
e1 cards operate ok in both modes?)
you need a crossover cable (not the same as a ethernet x-over)
take a look at: http://www.voip-info.org/wiki-crossover+T1+cable
frank
TE110P) but I am not sure what to buy.
if you want to splice asterisk between a pbx and an S0 (ISDN-BRI) from
your telco then you will need a ISDN Card that support NT mode e.g.
cards with HFC chipset like those from www.junghanns.net or www.beronet.com.
regards
frank sautter
: span=1,1,0,esf,b8zs
what are the effects you experience (besides there is no d-channel on
one line)?
regards
frank sautter
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frank sautter
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hi,
since my latest libpri update i get these messages:
!! Unable to handle ROSE operation 36
!! Unable to handle ROSE operation 30
i searched through ITU X.219 and X.229 but can't find any values for the
Remote Operations Service Elements.
are these AOC-E messages?
regards
frank
hi,
i made a patch which allows the forwarding and the setting of the Bearer
Capability ID during the ISDN SETUP phase.
this solves several problems (primarily faxing) with SIN (german:
Dienstekennung) and asterisk.
http://bugs.digium.com/bug_view_page.php?bug_id=0003547
Frank Sautter wrote:
i
hi,
i have the problem that i'm not able to set and receive the Service
Indication (SIN) from our E1-PRI and from our ericsson BP250.
The problem is, that the Bearer Capability (BC) together with the High
Level Compatibility (HLC) and Low Level Compatibility (LLC) forms the
Service Indicator
Peter Svensson wrote:
This is rather weird?
this are also my thoughts...
What network do you receive this from?
the calling party has an E1-PRI from the Deutsche Telekom (germany's
former monopolist) and our E1-PRI is from Arcor which is on of the new
telco companies founded after the
Kevin P. Fleming wrote:
Frank Sautter wrote:
our customer uses this feature to show the callerid of the original
caller when redirecting a call to a mobile phone.
That is RDNIS, it shows the redirected number. In other words, it's
not CLID (Calling Line ID).
Check the RDNIS channel variable
hi,
how can the charge info from a E1-PRI be received and be forwarded to a
classic PBX?
regards
frank
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hi,
a feature of euroisdn is, that you dail a number e.g. 0732194490 (where
0 is the extension of the call dispatcher) and the phone is forwarded to
someone with an extension of 26.
our ericsson showed after the call was picked up 07321944926 and no
longer the dialled 0732194490.
another
Frank Sautter wrote:
on our incoming E1-PRI from german telco Arcor the leading 0 for the
(area access code in europe) and the 00 (country accescode in europe)
are missing on incoming callerids.
only prepending a single 0 is not the solution as suggested by some
writers on this list, because
Peter Svensson wrote:
On Fri, 4 Feb 2005, Frank Sautter wrote:
RDNIS is empty.
So the operator sets an incomplete callerid? Sounds like a
misconfiguration at the operators end.
Do a pri intense debug span XXX on one of the calls and post the log
of the SETUP to CONNECT_ACK messages.
Protocol
is appreciated.
regards
frank sautter
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hi,
Frank Sautter wrote:
on our incoming E1-PRI from german telco Arcor the leading 0 for the
(area access code in europe) and the 00 (country accescode in europe)
are missing on incoming callerids.
after peter svensson gave me some hints on where to look after, i made a
small patch to current
hi jay,
Jay Milk wrote:
The result can be found here:
http://www.muware.com/asterisk/
it seems as if your webserver tries to execute the .php file instead of
making them available for download...
regards
frank
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hi,
on our incoming E1-PRI from german telco Arcor the leading 0 for the
(area access code in europe) and the 00 (country accescode in europe)
are missing on incoming callerids.
only prepending a single 0 is not the solution as suggested by some
writers on this list, because there is no way to
Frank Sautter schrieb:
* i can't signal Busy to the calling party.
asterisk receives busy from the ericsson PBX but does not forward
this to the external caller. i tried with exten = _.,102,Busy() with
no effect. this is the part of the extensions.conf i'm using:
peter svensson gave me
hi,
thanks to peter i solved my problems with the asterisk server spliced
between the telco and our ericsson BP250.
the problem was solved by setting 'overlapdial=yes'
Peter Svensson wrote:
Am Dienstag, den 25.01.2005, 22:39 +0100 schrieb Frank Sautter:
the setup desired with asterisk spliced
conferencing on 35, with 0 conference users
Set option AUDIO MODE, value: OFF(0) on Zap/35-1
disabled echo cancellation on channel 35
-- Hungup 'Zap/35-1'
regards
frank sautter
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http
on span 2
-- B-channel 0/27 successfully restarted on span 2
-- B-channel 0/28 successfully restarted on span 2
-- B-channel 0/29 successfully restarted on span 2
-- B-channel 0/30 successfully restarted on span 2
-- B-channel 0/31 successfully restarted on span 2
regards
frank sautter
or only after several tries.
regards
frank sautter
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hi,
i'm having problems getting asterisk spliced between an E1 PRI (german
Telco Arcor) and an Ericsson Business Phone 250 digital PBX.
The Asterisk Server has a TE405P with it's port 1 connected to the E1
PRI provided by our telecommunications provider Arcor and port 2
connected to the E1 PRI
hi,
is there a possibility to provide german dialtones on an IAXy S100IPWRD?
'language=de' sets only the messages to german (voicemail, etc.)
is there something like 'loadzone' as in /etc/zaptel.conf
regards
frank
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hi vincent,
Vincent Guidoux schrieb:
I have a problem for install chan_capi
My pc: Suse 9.1, with asterisk current stable en cvs
And patch the chan_capi
chan_capi.c:1076: error: structure has no member named cid
as you are writing and apparent to the error message you are posting,
you are using
hi,
i have a problem with distorted voicemail sound on our asterisk test
machine.
i'm using cvs-head (2004-01-17) and chan_capi 0.3.5 (with my patches to
make chan_capi compile with asterisk cvs-head) and a diva quad-bri isdn
card.
other things work well with my setup (dial in, dial out,
Vincent Guidoux schrieb:
Now i have a un new prob
Executing Dial(SIP/2500-0bbb, CAPI/@4202270:0796273153|30|r) in new
stack
Jan 17 13:14:39 NOTICE[4146]: chan_capi.c:1173 capi_request: didn't find
capi device with outgoing msn = 4202270. you should check your config
well the error message says it
the parameter for the hint command into the
'appdata' column.
my other problem is: how can includes of other contexts be accomplished?
regards
frank sautter
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.bin
Firmware: http://www.snom.com/download/snom220-3.52-beta-SIP.bin
he functions keys are configured:
fkey5!: dest sip:[EMAIL PROTECTED];user=phone
how can the blinking state of the leds be achieved?
is this a firmware version issue of the 3.52 i'm using?
regards
frank sautter
hi john,
John Williams schrieb:
i made a patch that allows the compilation of chan_capi-0.3.5 against
current CVS-HEAD of asterisk.
If I remove the -2.95 from the CC declaration I get a very large number
of errors, the same ones I get when trying to compile without the patch.
ok, i forgot to
hi,
i made a patch that allows the compilation of chan_capi-0.3.5 against
current CVS-HEAD of asterisk.
it also incorporates the capiAnswerFax patch
the patch can be downloaded at
http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2
regards
Frank Sautter
hi,
can someone give me any hints if the old german ISDN protocol '1TR6' is
supported by asterisk.
we have a potential customer who has an existing conventional PBX which
has to be extended by an asterisk server. unfortunately this existing
PBX speaks 1TR6 on it's ISDN ports.
regards
frank
with Asterisk. To activate 1TR6 all I would have to do is
upload the proper firmware to the card. Maybe the AVM Fritz! cards
support 1TR6 too. Worth checking out. The Eicon cards are expensive
while the AVM Fritz! is much cheaper.
On Thu, 2004-11-11 at 11:38 +0100, Frank Sautter wrote:
can someone
hello,
i just wanted to inform you, that i made some patches to say.c so * can
speak numbers and dates in a correct german syntax.
the patches are available through
http://bugs.digium.com/bug_view_page.php?bug_id=0002780
a compatible (but not complete) set of german sounds can be found on
hi,
our asterisk server is currently connected via 4 isdn trunks to our main
pbx using it as a voip gateway for homeworkers.
currently this is the dial command for outgoing calls
exten = _., 1, Dial,CAPI/141:${EXTEN}
what i like to do, is giving each sip-user a different outgoing msn (the
hi,
i just wanted to ask if there is a german localization for the audio
files of the mailbox available on the net.
regards
frank sautter
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