Hi all,
is where a possibility for simply parsing and changing variables for bad
characters ?
eg. removing a '/' from a number dialed by a manager-connected application
changing 123/4567890to 1234567890
via bash you could simply use 'echo ${exten/\//}' but i couldn't find a
working
Hi,
sure in an small office you can use iaxmodem/hylafax to receive faxes - we
use it for sending faxes, but would you try to set up about 100 iaxmodems
inside hylafax if you can handle it directly inside asterisk with rx_fax
and a small script ?
[EMAIL PROTECTED] schrieb am 20.12.2006
a few weeks ago I encountered the same problem.
I found out that asterisk is crashing when app_rxfax.so is calling line
327 of app_rxfax.c 'ast_frfree(int);' out of the testing tree running
with actual spandsp-0.0.3
commenting this line out it doesn't crash *, but that's no solution
it do work
please check if the old spandsp-version is kompletly removed
do you use the rxfax/txfax version out of the soft-switch/snapshots-folder
??? if not - try them
[EMAIL PROTECTED] wrote on 16.11.2006 11:27:36 AM:
Hi,
I'm using spandsp-0.0.3
On 16 nov 2006, at 12:12, [EMAIL PROTECTED] wrote:
please check if the old spandsp-version is kompletly removed
It is.
do you use the rxfax/txfax version out of the soft-switch/snapshots-
folder
??? if not - try them
From my original msg:
The app_rxfax.c in use is from:
try the rx_fax
and tx_fax below the snapshot-tree
within test-apps-asterisk-1.x
http://www.soft-switch.org/downloads/snapshots/spandsp/
[EMAIL PROTECTED] schrieb am
04.10.2006 22:11:43:
2006/10/4, Steve Underwood [EMAIL PROTECTED]:
Giedrius Augys wrote:
Hi,
Now I'm testing faxes with
[EMAIL PROTECTED] wrote on 31.08.2006
05:41:52 PM:
Matthias Fechner wrote:
Hello Roger,
* Roger Schreiter [EMAIL PROTECTED] [31-08-06 14:19]:
did google for asterisk and fax show no results?
yes I found spandsp but it will do everything in software.
Is it not a good idea
try to remove manually all parts of old spandsp-installations below /usr/
and /usr/local/ and reinstall both spandsp app_rtxfax
it's likely that you have some parts of the spandsp-0.0.3 left from prior
install which is incompatible to the 0.0.2-versions
[EMAIL PROTECTED] (Robert G. Ristroph)
TDMoE doesn't seem to be a good alternative.
it doesn't make sense to use an eth-interface used for
intranet-traffic/sip/sccp as well
...to heavy load to get a reliable function. On my test-asterisk with just
activated ztd_eth-module and configured zaptel it filled up my log with
error-messages
is the following zaptel.conf configuration correct for TDMoE used for
pri-cpe signalling - is this possible at all ?
I couldn't find an example...
loadzone=nl
defaultzone=nl
# pri E1 card
span=1,1,3,ccs,hdb3
bchan=1-15
dchan=16
bchan=17-31
# hfc-pci 1
span=2,1,3,ccs,ami
bchan=32-33
dchan=34
#
is it possible to route an ISDN-Data channel over an iax-connection ?
the setup is
pc with isdn-card - (zaphfc) Asterisk Server1 (iax) - (iax) Asterisk
Server2 (E1)-connecting to an external isdn-dialin router
via the iax-line the call is transfered as speech which is not accepted at
has anyone seen a bristuff version compatible to the actual *1.2.6/zaptel
1.2.5 ?
the actual bristuff-patches version 0.3.0 PRE 1l doesn't apply correctly
anymore...
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Am Montag, 3. April 2006 11.35 schrieb [EMAIL PROTECTED]:
has anyone seen a bristuff version compatible to the actual
*1.2.6/zaptel
1.2.5 ?
the actual bristuff-patches version 0.3.0 PRE 1l doesn't apply
correctly
anymore...
No, but I had the same problem. Somebody told me (on this
is where anyone out there having hfc-pci cards running with asterisk on
ppc-platform ?
any information on working cards, drivers, kernel, asterisk versions
would be helpful
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[EMAIL PROTECTED] wrote on 15.03.2006 14:37:27:
I am trying to use misdn insted of zaphfc to drive two billion isdn
cards
zaphfc is ok, but the problem with cdr and the fact tha you always have
to
wait the bristuffed version of asterisk took me to
try another way.
so I downloaded the misdn
##
# mISDN (experimental) #
##
#avmfritz - - - - - -
#hfcpci - - - - - -
#hfcsusb- - - - - -
I've got a problem with chan-misdn
I'm using asterisk with a hfcsusb-adapter in nt-mode connected to an
isdn-telephone
making calls to other internal clients like sip or sccp are without
problems
if I call into (or receive a call from) the pstn via a zap-channel (Digium
E1-card) my outgoing
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
i can confirm that this bug exists in 1.2.4 as well. we've managed to
fudge
it by dialplan tricks and Pickup().
Please report the bug.
In 1.2.1 it works fine.
thank you for the information...
the callgroup/pickupgroup settings are correct...
dialing *8 or *8# on any client (zap/sip/sccp) results in unknown
extension...
To pick-up with SIP phone, it has to be defined in sip.conf. Same goes
for zap and iax2.
callgroup and pickupgoup is configured in the config-files
is where anyone who knows what is needed to get the pickupexten (*8)
running ?
gentoo asterisk-stable 1.2.4/zap1.2.4 with bristuff
I've activated it in features.conf (default *8) and also tested other
extensions
res_features.so is loaded
show features says:
Builtin Feature Default
We've had this combination 206-xSeries and TE110P , but the zttest results
were not in the range above 99,76%
as well we had lots of echo-problems
...we changed to an other hardware platform
[EMAIL PROTECTED] wrote on 03.02.2006 16:49:14:
Hi,
I have an IBM xSeries 206 and now looking at
with incoming lines only maybe are active capi dual/quad-port cards from
AVM an alternative - but I've no experience with them together with
asterisk/chan_capi
an other way with 4 isdn-lines is to think about to order an partial E1
line with 8 channels...
[EMAIL PROTECTED] wrote on 31.01.2006
have you tried to parse the traffic what phone is requesting from your
tftp-server ?
maybe you get a hint where
[EMAIL PROTECTED] wrote on 05.01.2006 03:21:07:
I am working on adding three older Cisco phones to *, two 12SPs and one
30VIP. One of the 12SPs
(griffin) and the 30VIP (scott) is
I don't know if it's possible, but I use a workaround to simulate the
external dialtone:
I use '0' to access external lines
exten - _0,1,ChanIsAvail(Zap/g1)
exten - _0,2,playtones(dial)
exten - _0,3,goto(external_tone|et)
...extensions if some dialed without waiting for dialtone
have you checked the order the modules are loaded and that this matches
the zaptel.conf ?
[EMAIL PROTECTED] wrote on 22.12.2005 09:48:55:
Hi everybody,
I have a problem with my *..
I have an Octobri Card working good..
and 2 x100p clones ..
the fact is that * modeprobes ok and load drivers
an isdn-line has two usable 64k channels and you can connect multiple
phones to an isdn-line
each phone is using it's own msn/cid
for calls towards the isdn-phones you can tell asterisk to use an
specified channel
eg.
exten-123,1,Dial(Zap/1/123)
exten-124,1,Dial(Zap/2/124)
this way hints for
[EMAIL PROTECTED] wrote on 16.12.2005 16:18:49:
[EMAIL PROTECTED] wrote:
an isdn-line has two usable 64k channels and you can connect multiple
phones to an isdn-line
each phone is using it's own msn/cid
Since Asterisk is not aware of these being individual devices, there is
no way
Hi all
has anyone an working example of a hint-entry with a Zap-Channel ?
I've got hint working with SIP and SCCP but Zap doesn't seem to work
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this does work, and is adding the hint to the channel on the isdn-card
and you can also add a watch to the second-isdn-channel of the card
is it possible to use the cid of a isdn-phone as well to identify multiple
devices behind one line ?
[EMAIL PROTECTED] wrote on 15.12.2005 14:31:09:
I've encountered a few issues with zaphfc-cards...
I think you meen simple isdn-cards with cologne-chipset
first - I had three cards working with mixed-modes 3nt, 1nt/2te, 2te/1nt
it does work but it was not possible to reinitilize the cards to change
te/nt mode without reboot
also sometimes it
...why don't you put the hints in an own area as like
[watchgroup]
exten = 1,hint,SCCP/101
exten = 2,hint,SCCP/102
exten = 3,hint,SCCP/103
exten = 4,hint,SCCP/104
and then inside sccp.conf:
speeddial = 101,101,[EMAIL PROTECTED]
speeddial = 102,102,[EMAIL PROTECTED]
speeddial = 103,103,[EMAIL
...or test the PickUpChan command coming with the bristuff-patch from
zapata
Damian Funnell wrote:
Hi,
Does anyone have remote call pick-up working on * (either via SIP or
otherwise)? If so then can you post your features.conf, sip.conf
and/or
zapata.conf?
We can't seem to get
do you think it would make any difference to change the process-priority
if zttest is the only running process except ssh-daemon and the
login-shells ?
[EMAIL PROTECTED] wrote on 30.09.2005 18:11:47:
Are you starting Asterisk with the -p option (high priority?)
Also, do you get a different
just as an (bad) example:
we are using an x206 and couldn't get the zttest above 99.975
equal what we were doing
single irq, w/o acpi, w/o apic, different kernels, w/o
hyperthreading, different slots, a.s.o.
for an Digium wildcard TE110P
so if someone got such a board
Digium itself is saying their cards may work not properly with zttest
results below 99,98
the card itself is working the way that we can call out and receive
calls, but we encountered massive echo-problems - sometimes more,
sometimes less even on lines within the same phone-provider and be sure
finally I did it - I put some of the vars in (double)quotes - this didn't
work
even if there's a space inside, the vars need not to be kept inside
(double)quotes...
You probably want to use 'database put' for changing incoming CID
http://voip-info.org/tiki-index.php?page=database%20put
take a look into the wiki...
http://www.voip-info.org/wiki-Asterisk+variables
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Hi all,
I tried to set the calleridname of an incoming call to get different
incoming labels displayed for different incoming numbers.
This does work for hidden number-calls so I can set the displayed CIDName
on my cisco7960 from CID withheld to abc CID withheld
If the incoming CID isn't hidden
I don't think using database is the solution for prepending a shortname to
the cidname based on the dialed incoming extension
The cidname isn't static...
You probably want to use 'database put' for changing incoming CID
http://voip-info.org/tiki-index.php?page=database%20put
*CLI database
is where anyone who can tell me how it's possible to set nationalprefix
internationalprefix for a single isdn-card and not for all installed cards
?
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has anyone an idea how to display incoming national/international
isdn-pstn-calls correctly to internal isdn AND sccp/sip-phones ?
without nationalprefix=0 and internationalprefix=00 I get incoming phone
numbers correctly on isdn-phones
but the leading zero's are stripped of for non-isdn phones
have you checked if the card is recognized by the kernel
...loaded the needed module for the card
to see which modules are actually loaded: lsmod
to see which pci-cards are recognized by the kernel: lspci
...the digium cards are usually detected as an unknown network device
the needed module
maybe check that ztdummy is NOT loaded - otherwise I don't know... - call
digium
Wichtige Vorabinformation
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does anyone know howto set the softkeys of an Cisco 7960 running on an
asterisk server via chan_sccp ?
Mit besten Grüßen
Dirk Rieger
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does anyone know howto set the softkeys of an Cisco 7960 running on an
asterisk server via chan_sccp ?
You can have localized softkeys (with german language) using the
SCCP-dictionary.xml.
You can download it from the cisco website. Look for local ip telephony.
At least you can move the
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