[asterisk-users] parsing extensions
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 solution for the asterisk-extensions.conf
best regards
Dirk Rieger
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Re: [asterisk-users] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13 [ Virusgeprüft]
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 02:17:22: Hi, No IaxModem is only a modem simulator. Let asterisk do the difference, and send it to you iax extension... @++. Jean-Yves Avenard a écrit : Hi On 12/20/06, Lee Howard [EMAIL PROTECTED] wrote: This thread seems like an awfully crazy amount of work to get fax working when using IAXmodem and HylaFAX would do it without the headache, most likely. Does IAXmodem allows you to receive faxes with any extensions (auto-detecting incoming faxes). JY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Diese E-Mail und alle Anhänge enthalten vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese E-Mail und ihren Inhalt. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser E-Mail ist nicht gestattet. This e-mail and any attached files may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail by mistake) please notify the sender immediately and delete this e-mail. Any unauthorised duplication, disclosure or distribution of this e-mail and content is strictly forbidden. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13 [ Virusgeprüft]
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 with asterisk-1.2.9 but not with 1.2.13 - not tested 1.2.14 yet [EMAIL PROTECTED] schrieb am 18.12.2006 12:32:12: Hi On 12/18/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: Can you provide a backtrace of the crash? Sure. I've attached a backtrace for both 1.2.13 and 1.2.14 running the same version of spandsp and all other libraries. This is on a Fedora Core 6 machine (I can not attach the message as it makes the message over 40kB) http://www.avenard.org/asterisk/trace1-2-13.txt http://www.avenard.org/asterisk/trace1-2-14.txt Just saying it crashed doesn't really help. Well, the full backtrace was reported here last month, I was just pointing out that it was still happening with 1.2.14. Also: what libraries are involved? ldd /usr/lib/asterisk/modules/app_rxfax.so linked with spandsp 0.0.2 I get: linux-gate.so.1 = (0x00c6c000) libspandsp.so.0 = /usr/lib/libspandsp.so.0 (0x0071) libtiff.so.3 = /usr/lib/libtiff.so.3 (0x006b4000) libc.so.6 = /lib/libc.so.6 (0x001e7000) libm.so.6 = /lib/libm.so.6 (0x00e43000) libjpeg.so.62 = /usr/lib/libjpeg.so.62 (0x009c5000) libz.so.1 = /usr/lib/libz.so.1 (0x00d9e000) /lib/ld-linux.so.2 (0x00534000) I unfortunately can't try with spandsp 0.0.3 right now as I need a working asterisk ... linux-gate: spandsp: 0.0.3pre27 libtiff: 3.8.2 glibc: 2.5-3 libjpeg: 6b-37 and report what is the version and package of each library mentioned there. Any more automated way of doing this? This is standard Fedora Core 6. You can find last month, on this distribution list For the archive: http://lists.digium.com/pipermail/asterisk-users/2006-November/172652.html People mentioned this issue as well as where it was crashing. Hope that helps. Jean-Yves ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Diese E-Mail und alle Anhänge enthalten vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese E-Mail und ihren Inhalt. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser E-Mail ist nicht gestattet. This e-mail and any attached files may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail by mistake) please notify the sender immediately and delete this e-mail. Any unauthorised duplication, disclosure or distribution of this e-mail and content is strictly forbidden. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13
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
[http://www.soft-switch.org/downloads/snapshots/spandsp/
spandsp-20061116.tar.gz]
on a bristuffed asterisk (1.2.13)
[http://212.91.251.199/~junghanns.net/downloads/bristuff-0.3.0-
PRE-1v.tar.gz]
libtiff is at version 3.6.0
Running on: Linux router2 2.6.17-2-686 #1 SMP Wed Sep 13 16:34:10 UTC
2006 i686 GNU/Linux
Debian testing distro.
I've tried many combinations of bristuffed ast and spandsp versions,
but all fail at the same point. The last combination i got to work
was bristuffed 0.3.0-PRE-1i with spandsp-0.0.2-pre25 (on an earlier
kernel)
The app_rxfax.c in use is from:
[http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-
asterisk-1.2/app_rxfax.c]
On reception of a fax through RxFax, i get the exception. The
relevant part of the dialplan is
[macro-faxreceive]
; Receive a fax
exten = s,1,Set(FAXFILE=${FAXSPOOL}/${UNIQUEID}.tif) ; Save the
fax in a tif file
exten = s,2,RxFAX(${FAXFILE}) ; Receive it
exten = s,3,NoOp(Fax reception complete) ;
exten = s,4,Hangup
Running asterisk (with the above versions) through gdb and doing a
backtrace gives me:
#0 0xa7d45947 in raise () from /lib/tls/libc.so.6
#1 0xa7d470c9 in abort () from /lib/tls/libc.so.6
#2 0xa7d7afda in __fsetlocking () from /lib/tls/libc.so.6
#3 0xa7d8289f in mallopt () from /lib/tls/libc.so .6
#4 0xa7d82942 in free () from /lib/tls/libc.so.6
#5 0xa75efd68 in rxfax_exec (chan=0x818c5f8, data=0xa74a4798) at
app_rxfax.c:327
#6 0x08090088 in pbx_extension_helper (c=0x818c5f8, con=value
optimized out, context=value optimized out, exten=0x818c83c s,
priority=2,
label=0x0, callerid=0x0, action=1) at pbx.c:554
#7 0xa762cb05 in macro_exec (chan=0x818c5f8, data=0xa74aafe8) at
app_macro.c:221
#8 0x08090088 in pbx_extension_helper (c=0x818c5f8, con=value
optimized out, context=value optimized out, exten=0x818c83c s,
priority=1,
label=0x0, callerid=0x0, action=1) at pbx.c:554
#9 0x08091dee in __ast_pbx_run (c=0x818c5f8) at pbx.c:2231
#10 0x08092a1c in pbx_thread (data=0x818c5f8) at pbx.c:2518
#11 0xa7f0d0bd in start_thread () from /lib/tls/libpthread.so.0
#12 0xa7de892e in clone () from /lib/tls/libc.so.6
This seems to indicate that the ast_frfree(inf); at line 327 of
app_rxfax.c causes the problem chain?
I'm a bit lost on how to debug this further. Is this actually a
spandsp problem or is another package the cause?
Any tips?
marcel
--
Marcel van der Boom
HS-Development BV -- http://www.hsdev.com
So! webapplicatie framework -- http://make-it-so.info
[attachment smime.p7s deleted by Dirk Rieger/B-W]
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Re: [asterisk-users] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13
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: [http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps- asterisk-1.2/app_rxfax.c] This is the folder you mean, right? Yep Diese E-Mail und alle Anhänge enthalten vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese E-Mail und ihren Inhalt. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser E-Mail ist nicht gestattet. This e-mail and any attached files may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail by mistake) please notify the sender immediately and delete this e-mail. Any unauthorised duplication, disclosure or distribution of this e-mail and content is strictly forbidden. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spandsp and tif
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 spandsp. I have problems that spandsp do not add headers to fax page: LOCALHEADERINFO. Please help me. There is a bug in adding page header with spandsp-0.0.2pre26. I have fixed this in the development code, but I haven't yet put the fix into the 0.0.2prexx series. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have installed spandsp 0.0.3 , but I couldn't install rx_fax and tx_fax(from 0.0.2pre release) , because I've got error. I also have problem with tiff files, because I get error, if I have created tiff file from MS WORD (printing to tiff file) . Maybe you can say what parameters/atributes and programs I must choose, that avoid these erorrs (there is no problem with tiff fiiles created by rxfax :) ). Can you give me some advices how to solve these problems? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk?
[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 to use my modem for the fax stuff? Why would it not be a good idea to do things in software? Hi all, software-solution would be a good idea... eg. spandsp/iaxmodem hylafax but is where a app_rxfax/app_txfax planned/available for spandsp-0.0.3 ? this would myke things much easier as handling with hylafax, at least to receive fax. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.2.9.1 and spandsp and rxfax
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) schrieb am 18.07.2006 01:26:29: Hi, I have installed 1.2.9.1 on CentOS 4.3 successfully. I tried to install spandsp and app_rxfax.c and app_txfax.c and it crashes when it gets to the RxFax application. The spandsp-0.0.3 versions didn't work, because I could not find a version of app_rxfax.c and app_txfax.c that would compile with them. I had to use spandsp-0.0.2pre26. That compiled ok, and show applications rxfax works, but asterisk crashes completely when it gets to the RxFax application. There is no information at the *CLI prompt or the /var/log/asterisk/full other than it is entering the RxFax application. I checked by doing ldd /usr/lib/asterisk/modules/app_rxfax.so that I had the ldconfig stuff set up correctly so it could find the spandsp library. I am about to give up on this version of asterisk, and start trying older ones till I find one that works, but I thought I would ask for advice here first. --Rob -- http://rgr.freeshell.org/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] isdn-data over iax
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 (with kernel 2.6.13 asterisk1.2.9.1 zaptel/bri 1.2.6)...until disk-full ...TDMoE is automatically using T1 instead of the european E1 when using pri as signalling [EMAIL PROTECTED] schrieb am 29.06.2006 08:58:23: Hi, [EMAIL PROTECTED] wrote: 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... Any kind of Zaptel signalling should be fine. Check out http://www.voip-info.org/wiki/index.php?page=Asterisk+TDMoE Best regards, Florian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] isdn-data over iax
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 # hfc-pci 2 span=3,1,3,ccs,ami bchan=35-36 dchan=37 #TDMoE dynamic=eth,eth0/00:D0:09:E8:FA:EB,31,0 bchan=38-52 dchan=53 bchan=54-68 [EMAIL PROTECTED] schrieb am 27.06.2006 17:01:26: [EMAIL PROTECTED] wrote: 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 the remote end IAX is not suited for this. Maybe TDMoE is an option for you ? Florian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] isdn-data over iax
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 the remote end ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bristuff for * 1.2.6/zaptel 1.2.5
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... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bristuff for * 1.2.6/zaptel 1.2.5
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 list) that a new PRE bristuff was about to come out, supporting 1.2.6. For now I have switched to mISDN. as mISDN neighter support fax-protocol nor beeing able two work in NT-Mode it's no alternative for bristuff :( ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hfc-pci cards on ppc
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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] misdn problem
[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 installation script from beronet for the last
version ( I am using asterisk stable 1.2, so now is 1.2.5)
wget http://www.beronet.com/downloads/install-misdn-mqueue.tar.gz
tar -zxvf install-misdn-mqueue.tar.gz
cd /usr/src/install-misdn-mqueue
make
make install
everything OK
/etc/init.d/misdn-init scan
/etc/init.d/misdn-init config
/etc/init.d/misdn-init start
everything OK
then I modify the /etc/asterisk/misdn.conf, in a very standard way:
[general]
debug=0
method=standard
append_digits2exten=yes
bridging=yes
;tracefile=/var/log/asterisk/misdn.trace
[default]
immediate=yes
callgroup=1
pickupgroup=1
context=default
language=it
;nationalprefix=0
;internationalprefix=00
rxgain=0
txgain=0
dialplan=0
[TEports]
ports=1,2
context=from-pstn
msns=*
~
then:
chmod 755 /usr/lib/asterisk/modules
chown asterisk /dev/mISDN* -R
everything still OK
amportal start (I am using AMP )
OK.
when I try to access an external line, asterisk crashes with a
segmentation
fault;
the dial string is correct
...
-- Executing GotoIf(SIP/567-bb09, 1?20:21) in new stack
-- Goto (macro-dialout-trunk,s,20)
-- Executing SetVar(SIP/567-bb09, the_num=3481303063) in new
stack
-- Executing Dial(SIP/567-bb09, misdn/1/3481303063) in new stack
-- Called 1/3481303063
Ouch ... error while writing audio data: : Broken pipe
Segmentation fault (core dumped)
I am using Suse Linux 10, and I switched to default kernel (not SMP)
[EMAIL PROTECTED]:~ uname -r
2.6.13-15.8-default
Any help will be gratly appreciated.
by the way: I read it could be possible to use chan_capi insted of
chan_misdn, laying on misdn: is it correct: ?
And if it is, could anybody give me an advice on how ? I tried the 0.6.4
chan_capi version I succesfully installed on anothe box with Fritz!,
but in that case the capi driver for Fritz was present.
thank in advance,
Andrea
maybe try to dial via Dial(misdn/g:TEports/${EXTEN}) inside
extensions.conf
I encountered a few asterisk-crashes with mISDN as well
as it seems misdn doesn't like digital calls at all and is crashing in
this case...
yes, it's possible to use chan_capi via misdn/capi you just have to add
entries for the hfc-cards to your /etc/capi.conf
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Re: [Asterisk-Users] misdn problem
##
# mISDN (experimental) #
##
#avmfritz - - - - - -
#hfcpci - - - - - -
#hfcsusb- - - - - -
#hfcmulti - - - - - -
#sedlfax- - - - - -
#w6692pci - - - - - -
this is out of the gentoo capi.conf - simply uncomment the entry you need
I've never tried 2 hfc-cards - should be the last entry to be
changed/duplicated with 1/2 for the cardnumber as like your sample
[EMAIL PROTECTED] wrote on 15.03.2006 16:25:01:
Thank you for your answer.
I tried that syntax with misdn/g:TEports/${EXTEN}, but nothing changes.
what should I write in the /etc/capi.conf ?
If I had a Fritz, I would have
#SuSEconfig.isdn generated
# card fileproto io irq mem cardnr options
fcpci - - - - - 1
but having 2 billion ??? what to write ?
Andrea
[EMAIL PROTECTED]
de
Sent by: To
asterisk-users-bo Asterisk Users Mailing List -
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m.com asterisk-users@lists.digium.com
cc
15/03/2006 14.56 Subject
Re: [Asterisk-Users] misdn
problem
Please respond to
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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 installation script from beronet for the
last
version ( I am using asterisk stable 1.2, so now is 1.2.5)
wget http://www.beronet.com/downloads/install-misdn-mqueue.tar.gz
tar -zxvf install-misdn-mqueue.tar.gz
cd /usr/src/install-misdn-mqueue
make
make install
everything OK
/etc/init.d/misdn-init scan
/etc/init.d/misdn-init config
/etc/init.d/misdn-init start
everything OK
then I modify the /etc/asterisk/misdn.conf, in a very standard way:
[general]
debug=0
method=standard
append_digits2exten=yes
bridging=yes
;tracefile=/var/log/asterisk/misdn.trace
[default]
immediate=yes
callgroup=1
pickupgroup=1
context=default
language=it
;nationalprefix=0
;internationalprefix=00
rxgain=0
txgain=0
dialplan=0
[TEports]
ports=1,2
context=from-pstn
msns=*
~
then:
chmod 755 /usr/lib/asterisk/modules
chown asterisk /dev/mISDN* -R
everything still OK
amportal start (I am using AMP )
OK.
when I try to access an external line, asterisk crashes with a
segmentation
fault;
the dial string is correct
...
-- Executing GotoIf(SIP/567-bb09, 1?20:21) in new stack
-- Goto (macro-dialout-trunk,s,20)
-- Executing SetVar(SIP/567-bb09, the_num=3481303063) in new
stack
-- Executing Dial(SIP/567-bb09, misdn/1/3481303063) in new
stack
-- Called 1/3481303063
Ouch ... error while writing audio data: : Broken pipe
Segmentation fault (core dumped)
I am using Suse Linux 10, and I switched to default kernel (not SMP)
[EMAIL PROTECTED]:~ uname -r
2.6.13-15.8-default
Any help will be gratly appreciated.
by the way: I read it could be possible to use chan_capi insted of
chan_misdn, laying on misdn: is it correct: ?
And if it is, could anybody give me an advice on how ? I tried the
0.6.4
chan_capi version I succesfully installed on anothe box with Fritz!,
but in that case the capi driver for Fritz was present.
thank in advance,
Andrea
maybe try to dial via Dial(misdn/g:TEports/${EXTEN}) inside
extensions.conf
I encountered a few asterisk-crashes with mISDN as well
as it seems misdn doesn't like digital calls at all and is crashing in
this case...
yes, it's possible to use chan_capi via misdn/capi you just have to add
entries for the hfc-cards to your /etc/capi.conf
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[Asterisk-Users] misdn -- zap problem
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 audio from the misdn-device is very choppy. is where anything I can do about it ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: res_features pickupexten
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... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: res_features pickupexten
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 (zap/sip/sccp) - is anything else needed ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] res_features pickupexten
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 Current --- --- --- Pickup*8 *8 Blind Transfer# #1 Attended Transfer *2 One Touch Monitor *1 Disconnect Call * *0 the callgroup/pickupgroup settings are correct... dialing *8 or *8# on any client (zap/sip/sccp) results in unknown extension... using the automon-feature with *1 does work ...or is this feature only possible in the cvs-tree ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Server Wildcard TE110P [ Virusgeprüft]
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 the Wildcard TE110P to connect to our ISDN30. Has anyone any experience with this combination? Would the TE110P work in this server? I've listed the PCI slots the machine has: 2 ( 2 ) x PCI-X / 66 MHz - full-length ¦ 3 ( 3 ) x PCI - full-length Any response is appreciated. Phil. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interface card for Euro-ISDN (BRI) [ Virusgeprüft]
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 14:17:59: I'm looking for an interface card for termination of Euro-ISDN2 (BRI) lines. That is ISDN lines from the telco into my Asterisk box. Any recommendations, good/bad expiriences ? At present I'm looking at cards from BeroNet and Junghanns. How many lines do you want to terminate? Two to Four ISDN2 lines. That gives me a maximum of eight voice channels. /John ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco phone issue
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 booting correctly and I have dial tone. The other 12sp starts up, then I get a message on the display stating Requesting Load ID, then it reboots. I am not sure why this is occurring. The phone does was working on a CCM installation not long ago. Below is the skinny.conf file. Anyone seen this issue? ; ; Skinny Configuration for Asterisk ; [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 192.168.1.51 ; Address to bind to dateFormat = M-D-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 allow = all ; disallow = ; Typical config for 12SP+ [griffin] device=SEP0010EB002A64 model=12SP version=P00203010100 context=from-internal line = 1234 [emma] device=SEP00306409C932 model=12SP version=P00203010100 context=from-internal line = 1500 [scott] device=SEP0010EB0013DF model=30VIP version=P00203010100 context=from-internal line = 2000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] local exchange dialtone on ISDN/bristuff?
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
[external_tone]
exten = et,1,DigitTimeout(1)
exten = et,2,Playtones(dial)
exten = et,3,WaitExten(8)
exten = _X,1,DIAL(ZAP/g1/${EXTEN})
exten = _X.,1,DIAL(ZAP/g1/${EXTEN})
exten = _X,102,PLAYTONES(busy)
exten = _X.,102,PLAYTONES(busy)
[EMAIL PROTECTED] wrote on 04.01.2006 21:48:19:
How can I get external (telecom local exchange) dialtone on HFC ISDN BRI
with bristuff/zaphfc driver?
with capi, voip-info say that it should be something like:
Dial(CAPI/MSN:b)
But with zaphfc, if I try: Dial(ZAP/1/), I just get NOANSWER.
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Re: [Asterisk-Users] Problem with octobri and x100p clone
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 ok. but when I want to make a call outside, I get this error -- Executing Dial(SIP/203-7f0a, Zap/g1/69X65|30) in new stack Dec 19 12:33:51 NOTICE[3042]: app_dial.c:759 dial_exec: Unable to create channel of type 'Zap' the group 1 are 8 lines of the Octobri the group 2 are 2 x100p clones zapata.conf file- [EMAIL PROTECTED] asterisk]# vi zapata.conf ; ; Default context ; context=enlace signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=800 relaxdtmf=yes rxgain=4.0 txgain=4.0 busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived channel=25-26 group=2 switchtype = euroisdn signalling = bri_cpe ; p2mp NT mode (for connecting ISDN phones in point-to-multipoint mode) ;signalling = bri_net_ptmp ; p2p NT mode (for connecting an ISDN pbx in point-to-point mode) ;signalling = bri_net pridialplan = local prilocaldialplan = local nationalprefix = internationalprefix = 0 echocancel = yes context=incoming group = 1 ; S/T port 1 channel = 1-2 group = 1 ; S/T port 2 channel = 4-5 so on to latest Octobri port group = 1 ; S/T port 8 channel = 22-23 ---end of zapata.conf file zaptel.conf file- [EMAIL PROTECTED] etc]# vi zaptel.conf loadzone=es defaultzone=es # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami span=5,1,3,ccs,ami span=6,0,3,ccs,ami span=7,0,3,ccs,ami span=8,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 bchan=13,14 dchan=15 bchan=16,17 dchan=18 bchan=19,20 dchan=21 bchan=22,23 dchan=24 fxsks=25-26 --end of zaptel.conf file - someone could help me? thanks in advance Bye___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hint on Zap channels
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 Zap channels work for incoming calls but usually you use a group/span in your dialplan so it's possible to use both channels for any extension/msn but for outgoing calls both isdn-devices use any free channel of the isdn-line [EMAIL PROTECTED] wrote on 16.12.2005 09:13:33: [EMAIL PROTECTED] ha scritto: is it possible to use the cid of a isdn-phone as well to identify multiple devices behind one line ? I did not understand the question, what you mean? Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hint on Zap channels
[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 that hints could reliably work for them. thanks for the answer - I expected this, although I hoped something different ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hint on Zap channels
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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hint on Zap channels
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: [EMAIL PROTECTED] ha scritto: 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 Fixed in current CVS 1.2 and HEAD older versions have a case sensitivity issue so you have to write it in the right way this one works exten = 1, hint, Zap/1 this one does not work exten = 1, hint, ZAP/1 this one does not work exten = 1, hint, zap/1 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple HFC-PCI cards in mixed modes (TE+NT) won't work!...
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 happend that the cards 'got lost' - stop working without any known reason or by recalling ztcfg without florz patch it didn't work at all in cause of to many interrupts from the cards when using zaphfc check with 'dmesg' to see in witch mode the cards are initilized if this doesn't match what you need check/set the mode for the zaphfc-module inside /etc/modules.conf check that each zaphfc-card has it's own interrupt not sharing with an other device (boot with noapic-parameter to switch of apic to see the real bios-interrupts) check with zttest if you got results above 99,976% otherwise may give chan_misdn a chance - I got it to work but with similar problems like zaphfc [EMAIL PROTECTED] wrote on 22.11.2005 22:04:31: My asterisk server breaks as soon as I turn one of the two cards in to an NT card, which I suspect is an issue in the (Florz-patched) BriStuffed chan_zap.so module, but as my cards aren't original Junghanns', they obviously aren't supported by Junghanns... (ZapHFC btw shows both cards active, one as TE Master, the other as NT slave, so that seems to be OK! I therefore doubt it is the Florz patch!) Maybe it's worth a try, using chan_mISDN (experimental, but works!).. You can find the how-to pdf (for beronet, hfc, etc.. cards) on http://www.beronet.com/downloads/. There also is an install-script that helps you through the installation, I have gotten it to work with a junghanns card and 1x HFC pci card. Didn't have a 2nd hfc around to try back then.. I you have results (good/bad) keep the list (or me) posted :) Cheers. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Chan-sccp-users] Need help with hint and call group
...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 PROTECTED] speeddial = 104,104,[EMAIL PROTECTED] as the watchgroup is only used by the hint function it doesn't disturb your usual extensions... best regards Dirk Rieger [EMAIL PROTECTED] wrote on 10.10.2005 23:52:28: We have 4 employees and we’re running Cisco 7970 phones. Each phone has a unique SCCP line configured (in the autologin area of the sccp.conf file) for each employee. We have hints set up in the extension.conf file like the following: exten = 101,hint,SCCP/101 exten = 102,hint,SCCP/102 exten = 103,hint,SCCP/103 exten = 104,hint,SCCP/104 We have speeddial= lines set up for all other employees to assign the other employees to softkeys. All employee lines show up on all phones. This works fine. If employee A is on the phone, all other employees see his line as being in use. My problem is when we add an auto attendant to the mix. We want to add a new line to the phone called “Tech Support” and have that line ring if someone dials extension 400. We have lines SCCP/401-404 (configured in the sccp.conf file on the autologin= line as well) set up to display as the “Tech Support” line on each phone. We have the following in extensions.conf: exten = 400,1,Dial(SCCP/401SCCP/402SCCP/403SCCP/404,20) exten = 400,2,Voicemail([EMAIL PROTECTED]) exten = 400,3,Hangup exten = 400,102,Voicemail([EMAIL PROTECTED]) exten = 400,103,Hangup When someone calls extension 400, the “Tech Support” line rings on all phones. Great. Our problem: We want whoever answers the tech support line to show their personal line as being in use. We have tried adding the following (in addition to the other hints), but it doesn’t work: exten = 401,hint,SCCP/101 exten = 402,hint,SCCP/102 exten = 403,hint,SCCP/103 exten = 404,hint,SCCP/104 Theoretically, when a user picks up the “Tech Support” line, it should use extension 401, 402, 403, or 404 and therefore show busy next to the users primary extension (101, 102, 103 or 104). I don’t know if this is an issue with Asterisk (i.e. only allowing one hint per extension) or with the chan_sccp driver. Has anyone found a way to make this work? Jordan Bean Webcore Technologies, Inc. toll 800.584.9950 voice 512.320.7071 fax 512.320.7072 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote call pick-up
...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 this (seemingly simple) function to work. Check callgroups and pickupgroups in the channel configuration files. There are sample configurations in the sample configs. /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zttest - 100% ?
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 value if you run zttest this way: nice -n -20 zttest Carlos On 9/30/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: 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 that we've been messing around with all other possible parameters for weeks without any result. Until now we've got a setup that we can live with at least until we get different hardware. It's really worse calling someone and missing the name the called person said then picking up the phone in cause of echo-cancelling parameters or even think the line is dead, or if you've got massive echoes and it takes about 30 seconds to filter them out if at all. Dirk [EMAIL PROTECTED] wrote on 30.09.2005 16:34:18: [EMAIL PROTECTED] wrote: 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 to zttest 100% maybe could give some information if where's something special to run asterisk on such boards... otherwise I think there are production differences on the ibm-mainboards or the used chipsets we'll change hardware next... You don't have to have 100% on zttest. You probably won't get it. I get the same results on one of my servers and it runs perfectly. Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zttest - 100% ?
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 to zttest 100% maybe could give some information if where's something special to run asterisk on such boards... otherwise I think there are production differences on the ibm-mainboards or the used chipsets we'll change hardware next... [EMAIL PROTECTED] wrote on 29.09.2005 18:35:03: This might seem a silly question but, what is the true meaning of the numbers zttest spits out? On 9/29/05, Marco Supino [EMAIL PROTECTED] wrote: Hi, My TDM is on its own IRQ, and the x306 has only one full-size PCI slot.. so no playing with it, what results do you get from zttest ? what IRQ is the card on ? Marco. Damian Funnell wrote: Have you checked that the TDM400P isn't sharing an IRQ with anything else? Don't trust /proc/interrupts - run lspci -v to confirm this. We have * running on an x206 and found that the only way to stop the TDP400P sharing an IRQ with other devices was to juggle cards between slots. Hope this helps! Damian. Marco Supino wrote: Hi, I would like to know what type of configuration could get me closer to 100% hits in zttest, when testing a TDM400P with 4 FXO ports, I am currently running kernel 2.4.31, on a IBM Xseries 306, with 3gh CPU, HT is disabled, PCI latency was changed, i still cant get more then 99.975% in the zttest testings, Thanks for any info. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zttest - 100% ?
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 that we've been messing around with all other possible parameters for weeks without any result. Until now we've got a setup that we can live with at least until we get different hardware. It's really worse calling someone and missing the name the called person said then picking up the phone in cause of echo-cancelling parameters or even think the line is dead, or if you've got massive echoes and it takes about 30 seconds to filter them out if at all. Dirk [EMAIL PROTECTED] wrote on 30.09.2005 16:34:18: [EMAIL PROTECTED] wrote: 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 to zttest 100% maybe could give some information if where's something special to run asterisk on such boards... otherwise I think there are production differences on the ibm-mainboards or the used chipsets we'll change hardware next... You don't have to have 100% on zttest. You probably won't get it. I get the same results on one of my servers and it runs perfectly. Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SetCIDName question
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 *CLI database put cidname 111222 test user Updated database successfully *CLI database show cidname /cidname/111222 : test user so now when someone calls from 111.222., it will change the CID info to 'test user' On Tue, 2005-09-13 at 07:46, [EMAIL PROTECTED] wrote: 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 it works to use SetCallerID but not to change only the CIDName with SetCIDName. At least it's not displayed on my cisco7960 with chan_sccp any suggestions what I've could have done wrong ? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GotoIf Syntax to match first digits
take a look into the wiki... http://www.voip-info.org/wiki-Asterisk+variables ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SetCIDName question
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 it works to use SetCallerID but not to change only the CIDName with SetCIDName. At least it's not displayed on my cisco7960 with chan_sccp any suggestions what I've could have done wrong ? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SetCIDName question
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 put cidname 111222 test user Updated database successfully *CLI database show cidname /cidname/111222 : test user so now when someone calls from 111.222., it will change the CID info to 'test user' On Tue, 2005-09-13 at 07:46, [EMAIL PROTECTED] wrote: 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 it works to use SetCallerID but not to change only the CIDName with SetCIDName. At least it's not displayed on my cisco7960 with chan_sccp any suggestions what I've could have done wrong ? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zapata help needed howto configure nationalprefix for a single card
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 ? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zapata nationalprefix-problem [Virus checked]
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 when I set this prefixes inside zapata.conf my internal isdn-phones get this prefix twice... is it possible to unset the prefixes for one or more cards serving internal line ? I tried it without luck ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE205P installation problem - ZT_SPANCONFIG failed on span 1 [Virus checked]
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 should be wct2xxp - maybe wct4xxp will do this as well modules should be installed within /lib/modules/YOUR_KERNEL_VERSION/misc/ Wichtige Vorabinformation bw computer bezieht am 01.09.2005 neue Geschäftsräume. Unsere neue Adresse+Rufnummer: bw computer Fangdieckstr. 64 (1. Stock) 22547 Hamburg T: +49 40 / 49 296 - 0 F: +49 40 / 49 296 - 100 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE205P installation problem - ZT_SPANCONFIG failed on span 1 [Virus checked]
maybe check that ztdummy is NOT loaded - otherwise I don't know... - call digium Wichtige Vorabinformation bw computer bezieht am 01.09.2005 neue Geschäftsräume. Unsere neue Adresse+Rufnummer: bw computer Fangdieckstr. 64 (1. Stock) 22547 Hamburg T: +49 40 / 49 296 - 0 F: +49 40 / 49 296 - 100 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk cisco 7960 softkeys [Virus checked]
does anyone know howto set the softkeys of an Cisco 7960 running on an asterisk server via chan_sccp ? Mit besten Grüßen Dirk Rieger ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk cisco 7960 softkeys [Virus checked]
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 softkeys position changing sccp_protocol.h, but of course you can't add softkeys that are not supported right now (conference, pickup, etc.) Thanks for the information - maybe I should try to get the subscription accessing the cisco-server first :( ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
