Re: [Asterisk-Users] Playback() stops working.
On 5/1/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: I'm working on configuring asterisk 1.0.7 on Debian Sarge. The servers been tested a bit and seemed to working fine, but for some reason now, when I try and run the Playback() or Background() applications, or even try and goto voicemail asterisk refuses to play any sounds back to me. I have heard from 5 or so people about this problem. I run CVS STABLE (almost the same as 1.0.7) and had none of these issues. I wish I could help you, but I wanted to let you know that this is not a general problem. Well a quick reinstall of Asterisk solved the problem, but I hope it doesn't happen again :) Thanks for the heads up Eric. ~sm ___ 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] Meetme and a timing source
All, I seem to be confused :( Meetme won't work with the message That is not a valid conference number, please try again even with the simplest of configurations. Having trawled the list archives, wiki and harrased people on #asterisk I've come to a dead-end. I compiled ztdummy last night from source and it inserts without any errors. I also have a Digium TE110P card in the server although it currently isn't plugged into anything. Am I able to use this zaptel card for timing even though it isn't 'live'? I haven't configured a lot in /etc/zaptel.conf or /etc/asterisk/zapata.conf uname -r: 2.4.27-2-386 lon0asterisk01:~# lsmod Module Size Used byNot tainted ztdummy 1688 0 (unused) wcte11xp 20352 0 (unused) zaptel216608 2 [ztdummy wcte11xp] hisax 473648 0 (unused) isa-pnp25552 0 [hisax] isdn 112204 0 [hisax] slhc4144 0 [isdn] ehci-hcd 14764 0 (unused) usb-uhci 19504 0 [ztdummy] usbcore52268 1 [ehci-hcd usb-uhci] ide-scsi8272 0 piix7784 1 e1000 57676 1 ide-disk 12448 0 ide-detect 288 0 (unused) ide-cd 27072 0 cdrom 26212 0 [ide-cd] ide-core 91832 0 [ide-scsi piix ide-disk ide-detect ide-cd] rtc 5768 0 (autoclean) ext3 65388 3 (autoclean) jbd34628 3 (autoclean) [ext3] sd_mod 10764 8 (autoclean) megaraid2 28616 4 (autoclean) scsi_mod 86052 3 (autoclean) [ide-scsi sd_mod megaraid2] unix 12752 80 (autoclean) Any pointers greatfully received. ~sm ___ 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] Playback() stops working.
Hello, I'm working on configuring asterisk 1.0.7 on Debian Sarge. The servers been tested a bit and seemed to working fine, but for some reason now, when I try and run the Playback() or Background() applications, or even try and goto voicemail asterisk refuses to play any sounds back to me. How can I start to debug the cause of this? Thanks ~sm ___ 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] Some * scripts: Pull asterisk config from LDAP and authenticate() against voicemail passwords
Hello,
Wrote some Python scripts last night to scratch an itch I was having
with Asterisk.
http://www.beerandspeech.org/cgi-bin/blosxom.cgi/tech/linux/050429a.html
http://www.beerandspeech.org/images/050429/asterisk-config.py.txt
http://www.beerandspeech.org/images/050429/asterisk-passwd.py.txt
asterisk-config.py lets me store asterisk config data in LDAP and generate
config files from it
head -n 32 asterisk-config.py
#!/usr/bin/python
# This script connects to an LDAP server looking for attributes that it can use
# to build some asterisk config files
#
# You need to setup your LDAP server (this was written in mind to connect to AD)
# and populate the correct attributes - it will then go away and build sip.conf,
# extensions.conf, SIP$MAC.cnf, SEP$MAC.cnf and CTLSEP$MAC.cnf
#
# This script builds sip.conf and extensions.conf in 3 parts... it reads a file
# (by default /etc/asterisk/sip.conf.head ) then dynamically builds the
# extensions and then appends the contents of sip.conf.tail
#
# This means you can maintain the static content of sip.conf ( the [general]
# parameters and dynamically build the rest
#
# On my extensions.conf.head the last line is [default] so all the
# dynamically created extensions go into this context
#
# You can also elect to not include certain SIP lines, phones and extensions in
# the autoconfiguration process
#
# It also writes out a Cisco XML phone directory file and a HTML phone list
# style file. On these files you can expand your LDAP search to include
# external non-asterisk users - maybe business contacts or such-like
#
# BUGS: Currently doesn't do a lot of error checking so missing attributes may
# make the script barf.. also doesn't do anything to voicemail.conf - thats
coming
# next
#
# If you have any questions, comments or patches please email me at [EMAIL
PROTECTED]
and secondly asterisk-passwd.py allows me to run the Authenticate() command
using the same password
people have for their voicemail. Basically greps voicemail.conf and dumps their
password into separate file
head -n 18 asterisk-passwd.py
#!/usr/bin/python
# I needed to add a Asterisk Authenticate() command to a dialplan but
# I wanted to use the same passwords as those in voicemail.conf to avoid giving
# users multiple passwords - grepping voicemail.conf seemed a little complicated
# so this script runs every 5 minutes under cron and it refreshs the password
files
# in case a user changes their voicemail password (which happens like.
never!)
#
# To include in the dial plan I used this
# [callforwarding]
# exten = s,1,Authenticate(/etc/asterisk/passwords/${CALLERIDNAME})
#
# So the Authenticate command would read /etc/asterisk/passwords/683 for my
extension
#
# BUGS: Hmm, I guess it depends if your ${CALLEDIDNAME} equals your numeric
extension number - mine did :)
# If not play with the splitting of the line to get the correct index for the
# filename - look at the output of asterisk -rvv to see which filename it's
trying to read
# Any questions or comments welcome to [EMAIL PROTECTED]
Hope someone finds these useful - please let me know if you use them and it
works.
Rgds
~sm
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Re: [Asterisk-Users] One touch voicemail on Cisco 7940/60
On Thu, 2005-04-21 at 21:36 +0100, Ron Wellsted wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Morris, Simon wrote:
Hello,
I'd like to program my Cisco phones to authenticate themselves to
voicemail upon hitting the right button on my 7940/60's
Ideally the voicemail app will detect which extension the call is
coming
from and drop the user straight into the menu.
Is this possible?
Many thanks
~sm
Yes this is possible.
In your extensions.conf:
exten = _8501,1,Answer()
exten = _8501,2,VoicemailMain(s${CALLERIDNUM})
exten = _8501,3,Hangup()
then program the messages button to dial 8501 either via settings, SIP
Settings, Messages URI (set to 8501) or in the SIPDefault.cnf file
That method works perfectly - thanks to all that took the time to answer
~sm
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Re: [Asterisk-Users] One touch voicemail on Cisco 7940/60
On Fri, 2005-04-22 at 11:13 +0100, Simon Morris wrote:
On Thu, 2005-04-21 at 21:36 +0100, Ron Wellsted wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Morris, Simon wrote:
Hello,
I'd like to program my Cisco phones to authenticate themselves to
voicemail upon hitting the right button on my 7940/60's
Ideally the voicemail app will detect which extension the call is
coming
from and drop the user straight into the menu.
Is this possible?
Many thanks
~sm
Yes this is possible.
In your extensions.conf:
exten = _8501,1,Answer()
exten = _8501,2,VoicemailMain(s${CALLERIDNUM})
exten = _8501,3,Hangup()
then program the messages button to dial 8501 either via settings,
SIP
Settings, Messages URI (set to 8501) or in the SIPDefault.cnf file
Guys - sorry to change my original question.
That solution does exactly what I asked BUT! I'd like it to dial and
know which extension I'm coming from and then prompt for the password.
Just realised that the solution above allows people to wander around
listening to other peoples voicemail at the press of the button :-)
So.. how to bypass the Enter your mailbox number stage in voicemail
and go straight to the password prompt.
Thanks guys
~sm
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RE: [Asterisk-Users] Cisco 7940G SIP Conversion
On Thu, 2005-04-14 at 08:11 +0100, Michael West wrote: Boris, Thanks for sharing your results. I had to upgrade to image P0S3-06-3-00 first. Once I was at this level, I could then upgrade to image P0S3-07-4-00. I WAS aware of the differences in spelling in the OS79XX.TXT file and SIPDefault.cnf and KEPT the file names DIFFERENT according to a document on voip.info.org. Michael, Boris, Is it possible that you could send me a copy of your SIP.cnf file as I'm having trouble with mine... My 7960 keeps rebooting after TFTPing it down and I suspect it's too big. Seeing a working example would really help me out Thanks ~sm ___ 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] Who is willing to help an Asterisk newby?
On Wed, 2005-04-13 at 14:21 +0100, Wolf N. Paul wrote: As of last night, I have a working Asterisk system, courtesty of [EMAIL PROTECTED]. Now comes the need to iron out the wrinkles and fine-tune my setup. Who would be willing for me to shoot questions at him/her which would just annoy the list if I brought them here? Personally I think it is much more beneficial for you to ask the list questions - rather than disappear into a one-on-one session with someone. Not only for yourself who would get a wider range of advice and opinions from various people on-list, but for me :) who doesn't know all that much about * And of course for anyone reading the archives in time to come... ~sm ___ 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 on debian sarge doesn't start with CAPI module errors
Hello, Fresh install of Debian Sarge and asterisk from the debian archives. Asterisk doesn't start and dies with the following message. [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Apr 13 15:38:44 NOTICE[1580]: chan_capi.c:2635 load_module: CAPI not installed! Apr 13 15:38:44 WARNING[1580]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returning -1 Apr 13 15:38:44 WARNING[1580]: chan_capi.c:2811 unload_module: Unable to unregister from CAPI! == Unregistered channel type 'CAPI' Apr 13 15:38:44 WARNING[1580]: loader.c:440 load_modules: Loading module chan_capi.so failed! I have an ISDN card I'm going to install later, but I want to get Asterisk up and running with SIP first. lon0asterisk01:~# dpkg -l | grep asterisk ii asterisk 1.0.5-2open source Private Branch Exchange (PBX) ii asterisk-app-d 0.0.20050203-2 Text entry application for Asterisk ii asterisk-app-f 0.0.20050203-2 Softfax application for Asterisk ii asterisk-chan- 0.3.5-11 Common ISDN API 2.0 implementation for Aster ii asterisk-confi 1.0.5-2config files for asterisk ii asterisk-dev 1.0.5-2development files for asterisk ii asterisk-doc 1.0.5-2documentation for asterisk ii asterisk-gtk-c 1.0.5-2gtk based console for asterisk ii asterisk-h323 1.0.5-2asterisk H.323 VoIP channel ii asterisk-promp 1.0-1 German prompts for the Asterisk PBX ii asterisk-promp 0.0.20040928-1 French voice prompts for Asterisk ii asterisk-promp 0.8-2 Swedish voice prompts for Asterisk ii asterisk-sound 1.0.5-2sound files for asterisk ii asterisk-web-v 1.0.5-2web based (GCI) voice mail interface for ast lon0asterisk01:~# dpkg -l | grep capi ii capisuite 0.4.5-2easy fax and voice box solution for ISDN/CAP ii libcapi20-23.6.2005-01-03 libraries for CAPI support Any ideas? Thanks ~sm ___ 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
