Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Hi
On Wed, Dec 19, 2007 at 12:19:08AM -0500, dave cantera wrote:
ok, here is my $0.02... I created a script since I had to
install/update so often and for various reasons...
you can choose to compile automatically or manually...
modify the current release numbers, your repository, and source root...
all else is automated..
which is unloading zap driver, stopping a running asterisk, getting the
current release, untar'ng it and compiling it...
enjoy,
daveC
You can find my take on the subject at
http://updates.xorcom.com/astribank/bristuff/1.4/bristuff-current/
I improved the existing scripts from bristuff to be more potent, as
explained in
http://updates.xorcom.com/astribank/bristuff/1.4/INSTALL.html
The bristuff scripts have a little wrapper install.sh that calls
download.sh (downloads and patches. Kind of like rpmbuild -bp) and
compile.sh (builds and installs).
That separation can reduce some of the need for user interaction in your
script.
If you want to use them, I figure you should just remove the patching
commands and then you should be able to use those scripts mostly
unchanged.
#!/bin/sh
#
#get_latest_rel.sh
#
# Dave Cantera: [EMAIL PROTECTED]
#
#get the current asterisk release components, put them in our REPOSITORY
#and unpack them in SRC_ROOT
--- Change to suite between these lines --
VER_AST=1.4.16
VER_ZAPTEL=1.4.7.1
VER_LIBPRI=1.4.3
VER_ADDONS=1.4.5
REPOSITORY=/root/tarballs
SRC_ROOT=/usr/local/src
--- Change to suite between these lines --
HTTP_SITE=http://downloads.digium.com;
PUB_DIR=/pub
TARBALL_AST=/asterisk/releases/asterisk-${VER_AST}.tar.gz
TARBALL_LIBPRI=/libpri/releases/libpri-${VER_LIBPRI}.tar.gz
TARBALL_ZAPTEL=/zaptel/releases/zaptel-${VER_ZAPTEL}.tar.gz
TARBALL_ADDONS=/asterisk/releases/asterisk-addons-${VER_ADDONS}.tar.gz
COMPONENTS=${HTTP_SITE}${PUB_DIR}${TARBALL_AST}
${HTTP_SITE}${PUB_DIR}${TARBALL_ZAPTEL}
${HTTP_SITE}${PUB_DIR}${TARBALL_LIBPRI}
${HTTP_SITE}${PUB_DIR}${TARBALL_ADDONS}
echo
echo
echo we are prepared to get the complete current release
echo of asterisk, libpri, zaptel, and addons
echo the tarballs will be placed in our REPOSITORY and
echo then extracted to our SRC_ROOT
echo
echo --- Activity Recap
echo
echo TARBALL REPOSITORY: ${REPOSITORY}
echoSRC_ROOT: ${SRC_ROOT}
echoasterisk tarball: ${TARBALL_AST}
echo libpri tarball: ${TARBALL_LIBPRI}
echo zaptel tarball: ${TARBALL_ZAPTEL}
echo addons tarball: ${TARBALL_ADDONS}
echo
echo -n Are You Ready? Y to procced:
read ANSWER
if [ null${ANSWER} == nullY ]
# a matter of style:
case $ANSWER in Y* | y*) :;;
*) echo Aborted by user ;;
exit 0
esac
# and good bye to unneeded nesting.
then
echo
echo -
echo stopping asterisk
echo
echo choose your poison:
echo a) /usr/bin/asterisk -xr stop now
echo b) /etc/init.d/asterisk stop
echo
echo -n which one?
read STOPCMD
if [ null${STOPCMD} == nulla ]
then
/usr/bin/asterisk -r -x 'stop now'
fi
if [ null${STOPCMD} == nullb ]
then
/etc/init.d/asterisk stop
fi
echo
echo -
echo get the current asterisk component releases and put them in
our repository ${REPOSITORY}
# lets go to the repository directory
cd ${REPOSITORY}
for TARBALL in `echo ${COMPONENTS}`
do
echo getting component: ${TARBALL}
#wget ${TARBALL}
Err... one needs to uncomment that line, I guess.
I tend to like using 'wget -c' . Otherwise strange things may happen if
I press ctrl-C in the middle of the download.
Sadly, the current downloads.digium.com will make you re-download the
tarballs
done
TARFILES=
asterisk-${VER_AST}.tar.gz
libpri-${VER_LIBPRI}.tar.gz
zaptel-${VER_ZAPTEL}.tar.gz
asterisk-addons-${VER_ADDONS}.tar.gz
echo
echo -
echo unpack the current asterisk component tarballs into our
source root ${SRC_ROOT}
# lets go to the source root directory
cd ${SRC_ROOT}
for TARBALL in `echo ${TARFILES}`
do
echo untar'ng component: ${TARBALL}
#tar xzf ${TARBALL}
done
echo
echo -
echo unloading Zap drivers
# unload the zaptel drivers
ZAP_MODULES=`lsmod | grep zap | awk '{printf(%s,,$4)}' | sed 's/,/
/g'`
for MODULE in `echo ${ZAP_MODULES}`
do
echo unloading zap module: ${MODULE}
#modprobe -r ${MODULE}
done
echo
echo now you are ready to compile at ${SRC_ROOT}
echo
echo -n Shall I continue with the compile? Y?
read COMPILE
if [ null${COMPILE} == nullY ]
then
echo Compiling Zaptel
Re: [asterisk-users] AsteriskNOW release date???
Speaking of attacks that aren't fair. Trixbox != FreePBX. They're completely separate products. Tilghman Lesher wrote: FreePBX seems to be the most logical choice to me. Which is being leveraged to take away business to anyone who has not sworn allegiance to Fonality. Sorry, couldn't resist. ;-) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-gui] Asterisk GUI - Call Waiting
bkruse wrote: Is this with the latest version of the gui? (branches/asterisknow) (http://asteriskNOW.org/install-related) Tell me what revision, and paste the context of the user entry thats having a problem. -bk Will Tatam wrote: Has anyone tested disabling call waiting for a SIP extension via the GUI ? I have deselected call waiting for a user with a SNOM 360 and applied my changes but they still get calls waiting and are reporting that 80% of the time when they get the bleeping in their ear when the new call comes in and that it kills the current call before they get chance to respond in any way ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-gui mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-gui ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-gui mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-gui This is using asgterisk now beta 6 [8018] callwaiting=no cid_number=02380988018 context=numberplan-custom-1 [EMAIL PROTECTED] fullname=Andrew Cartlidge group= hasagent=no hasdirectory=yes hasiax=no hasmanager=no hassip=yes hasvoicemail=yes host=dynamic mailbox=8018 secret=14731473 threewaycalling=yes zapchan= registeriax=no registersip=yes canreinvite=yes nat=no dtmfmode=rfc2833 vmsecret=1473 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
tzafrir,
thanks for the note. btw, Great docs!
asciidocs looks cool too!
thanks!
daveC
Tzafrir Cohen wrote:
Hi
On Wed, Dec 19, 2007 at 12:19:08AM -0500, dave cantera wrote:
ok, here is my $0.02... I created a script since I had to
install/update so often and for various reasons...
you can choose to compile automatically or manually...
modify the current release numbers, your repository, and source root...
all else is automated..
which is unloading zap driver, stopping a running asterisk, getting the
current release, untar'ng it and compiling it...
enjoy,
daveC
You can find my take on the subject at
http://updates.xorcom.com/astribank/bristuff/1.4/bristuff-current/
I improved the existing scripts from bristuff to be more potent, as
explained in
http://updates.xorcom.com/astribank/bristuff/1.4/INSTALL.html
The bristuff scripts have a little wrapper install.sh that calls
download.sh (downloads and patches. Kind of like rpmbuild -bp) and
compile.sh (builds and installs).
That separation can reduce some of the need for user interaction in your
script.
If you want to use them, I figure you should just remove the patching
commands and then you should be able to use those scripts mostly
unchanged.
#!/bin/sh
#
#get_latest_rel.sh
#
# Dave Cantera: [EMAIL PROTECTED]
#
#get the current asterisk release components, put them in our REPOSITORY
#and unpack them in SRC_ROOT
--- Change to suite between these lines --
VER_AST="1.4.16"
VER_ZAPTEL="1.4.7.1"
VER_LIBPRI="1.4.3"
VER_ADDONS="1.4.5"
REPOSITORY="/root/tarballs"
SRC_ROOT="/usr/local/src"
--- Change to suite between these lines --
HTTP_SITE="http://downloads.digium.com"
PUB_DIR="/pub"
TARBALL_AST="/asterisk/releases/asterisk-${VER_AST}.tar.gz"
TARBALL_LIBPRI="/libpri/releases/libpri-${VER_LIBPRI}.tar.gz"
TARBALL_ZAPTEL="/zaptel/releases/zaptel-${VER_ZAPTEL}.tar.gz"
TARBALL_ADDONS="/asterisk/releases/asterisk-addons-${VER_ADDONS}.tar.gz"
COMPONENTS="${HTTP_SITE}${PUB_DIR}${TARBALL_AST}
${HTTP_SITE}${PUB_DIR}${TARBALL_ZAPTEL}
${HTTP_SITE}${PUB_DIR}${TARBALL_LIBPRI}
${HTTP_SITE}${PUB_DIR}${TARBALL_ADDONS} "
echo
echo
echo " we are prepared to get the complete current release "
echo " of asterisk, libpri, zaptel, and addons "
echo " the tarballs will be placed in our REPOSITORY and "
echo " then extracted to our SRC_ROOT "
echo
echo "--- Activity Recap "
echo
echo " TARBALL REPOSITORY: ${REPOSITORY}"
echo " SRC_ROOT: ${SRC_ROOT}"
echo " asterisk tarball: ${TARBALL_AST}"
echo " libpri tarball: ${TARBALL_LIBPRI}"
echo " zaptel tarball: ${TARBALL_ZAPTEL}"
echo " addons tarball: ${TARBALL_ADDONS}"
echo
echo -n " Are You Ready? Y to procced: "
read ANSWER
if [ "null${ANSWER}" == "nullY" ]
# a matter of style:
case "$ANSWER" in Y* | y*) :;;
*) echo " Aborted by user ";;
exit 0
esac
# and good bye to unneeded nesting.
then
echo
echo "-"
echo " stopping asterisk "
echo
echo " choose your poison: "
echo " a) /usr/bin/asterisk -xr stop now"
echo " b) /etc/init.d/asterisk stop "
echo
echo -n " which one? "
read STOPCMD
if [ "null${STOPCMD}" == "nulla" ]
then
/usr/bin/asterisk -r -x 'stop now'
fi
if [ "null${STOPCMD}" == "nullb" ]
then
/etc/init.d/asterisk stop
fi
echo
echo "-"
echo " get the current asterisk component releases and put them in
our repository ${REPOSITORY}"
# lets go to the repository directory
cd ${REPOSITORY}
for TARBALL in `echo ${COMPONENTS}`
do
echo "getting component: ${TARBALL} "
#wget ${TARBALL}
Err... one needs to uncomment that line, I guess.
I tend to like using 'wget -c' . Otherwise strange things may happen if
I press ctrl-C in the middle of the download.
Sadly, the current downloads.digium.com will make you re-download the
tarballs
done
TARFILES="
asterisk-${VER_AST}.tar.gz
libpri-${VER_LIBPRI}.tar.gz
zaptel-${VER_ZAPTEL}.tar.gz
asterisk-addons-${VER_ADDONS}.tar.gz "
echo
echo "-"
echo " unpack the current asterisk component tarballs into our
source root ${SRC_ROOT}"
# lets go to the source root directory
cd ${SRC_ROOT}
for TARBALL in `echo ${TARFILES}`
do
echo "untar'ng component: ${TARBALL} "
#tar xzf ${TARBALL}
done
echo
echo "-"
echo " unloading Zap drivers"
# unload the zaptel drivers
ZAP_MODULES=`lsmod | grep zap | awk '{printf("%s,",$4)}' | sed 's/,/
/g'`
for MODULE in `echo ${ZAP_MODULES}`
do
echo "unloading zap module: ${MODULE}"
#modprobe -r ${MODULE}
done
echo
echo " now you are ready to compile at ${SRC_ROOT} "
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Hi Steve, On Tue, 2007-12-18 at 19:43 -0800, Steve Edwards wrote: The old syntax was inconsistent -- show manager command vs sip show channels and just plain bad -- for example sip reload should have been reload sip. I agree. Reload sip would be the logical thing. [snip] Approach the command line like a noob. I want Asterisk to show me something so I'll start the command line with show. I'm not quite sure what I'm doing, so I'll press TAB to see what I can show. Oh, channel looks like what I want. Hmm, too much. Maybe I should have qualified what kind of channel I'm looking for BEFORE the word channel. That makes sense to me. It's also what I'm used to from working with other equipment. Here's a suggestion -- stop thinking like a parser and start thinking like a person :) Which makes more sense (at least in English)? 1) show black dogs -- show sip channels 2) black show dogs -- sip show channels 3) dogs black show -- channels sip show 4) show dogs black -- show channels sip 5) black dogs show -- sip channels show 6) dogs show black -- channels show sip Excellent example. I'll put my 0.2 cents on #1 :) Is it too late to fix this for 1.6? I sincerely hope not. Your example shows that the CLI could use some TLC. Let's hope the powers that be agree. +1 Regards, Patrick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Hi Olle, On Wed, 2007-12-19 at 08:20 +0100, Johansson Olle E wrote: [snip] The old way was a mess. We had two different systems, one like your old show and one syntax starting with the module name. We had to move forward with only one syntax and decided to go for modulename verb which is not human language-like, but we haven't really clamed that the CLI is a human language parser. Maybe we should go for an avatar approach... I have not followed this discussion but the decision is quite puzzling to me. Why would you make the human interface to Asterisk not human language-like? That's just not logical. Were the devs expecting that the majority of users would be HAL2000 clones instead of humans? :) [snip] I do understand the pain with changing the CLI though, I hate to switch from Asterisk 1.0 to 1.2 to 1.4 and trunk and have different commands. This is only an issue for developers and existing users who have (a combination of) 1.0, 1.2 and 1.4 boxes and upgrade to a version with an improved CLI. New users who get the latest major version of Asterisk (assuming that version has the improved human language-like CLI) don't have that issue. I don't mind the CLI differences because at some point I move all my boxes to the new major release so only have to deal with one version of the CLI at any time. Change usually means one needs to adopt and an improved CLI seems worth it to me. Regards, Patrick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
I think it should be core dogs show black. Seriously though, I think you make a good point. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Steve Edwards Enviado el: miercoles, 19 de diciembre de 2007 4:43 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old! On Sat, 15 Dec 2007, Johansson Olle E wrote: I wonder if there are any major obstacles for upgrading. How about the change from a bad command line interface to a really bad command line interface? I mean, Seriously? (in a Grey's Anatomy kind of way...) The old syntax was inconsistent -- show manager command vs sip show channels and just plain bad -- for example sip reload should have been reload sip. The new syntax continues down the noun-verb path instead of correcting itself and using verb-noun like most other applications (MySQL, GDB, Oracle, etc.) Then, just to make it worse, now I have to learn which commands somebody (arbitrarily) decided are core and which are not -- for what benefit? Certainly doesn't make MY job easier! Approach the command line like a noob. I want Asterisk to show me something so I'll start the command line with show. I'm not quite sure what I'm doing, so I'll press TAB to see what I can show. Oh, channel looks like what I want. Hmm, too much. Maybe I should have qualified what kind of channel I'm looking for BEFORE the word channel. Here's a suggestion -- stop thinking like a parser and start thinking like a person :) Which makes more sense (at least in English)? 1) show black dogs -- show sip channels 2) black show dogs -- sip show channels 3) dogs black show -- channels sip show 4) show dogs black -- show channels sip 5) black dogs show -- sip channels show 6) dogs show black -- channels show sip Is it too late to fix this for 1.6? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Download Asterisk
Hey everybody, I just went to download and test the current version of Asterisk, go tired of the redirect impeded in the header and sat down for a few minutes and created the following shell script: Hope somebody finds it useful. I named it dget: [EMAIL PROTECTED] Download]# cat /usr/local/bin/dget #!/bin/sh eval filename=`echo $1|cut -f2 -d=` wget $filename ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Codecs: g729 and g723
Hi All; Does new asterisk version still requires g729 to be bought or it required for some features and not for others? Also, how can I use g723 with Asterisk? Regards Bilal Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Download Asterisk
Doug Lytle wrote: I just went to download and test the current version of Asterisk, go tired of the redirect That's why I created http://www.kempgen.net/asterisk/current/ Makes my life easier. :-) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
On Wed, Dec 19, 2007 at 02:40:21PM +0100, James Collier wrote: I think it should be core dogs show black. You should use color instead of black to make the comparison more valid. show dog color Doesn't sound right (Here's a colour for you, doggy. Fetch!). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX for asterisk to asterisk
Hi, I am not new to asterisk but this is the first time im using iax protocol. I have always used sip before, but i have heard its better to use iax if i want communication between 2 asterisk servers. I just registered asterisk server1 with asterisk server2 and tried to call server1 from 2 but the call does not pass thru. i dont see any messages on the recieving asterisk cli but the caller asterisk cli show the following messages Dec 19 11:51:57 WARNING[1987]: chan_iax2.c:7103 socket_read: Call rejected by *ip-address*: No authority found Dec 19 11:51:57 NOTICE[1987]: chan_iax2.c:1629 iax2_destroy: Avoiding IAX destroy deadlock -- Hungup 'IAX2/peer141-3' any clues? - Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Codecs: g729 and g723
On Wednesday 19 December 2007 08:01:16 bilal ghayyad wrote: Does new asterisk version still requires g729 to be bought or it required for some features and not for others? If you need to record audio, playback audio, or do anything which necessitates translating g729 to another codec (such as calling a channel using ulaw), you need to buy licenses. Also, how can I use g723 with Asterisk? The only way currently is to buy the transcoder card, as the patent holders will not license g.723.1 directly to Digium for a reasonable price. Note that this is only necessary for the next 7 years, as the related patents expire sometime in 2014. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk with alsa
I need to use asterisk with ALSA and CONSOLE/dsp. This was working fine standalone. However, I also need sound from another program on that same ALSA connection. It seems as though asterisk opens the sound port exclusively. I need it to share the sound card. How do I do that? Jerry ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsteriskNOW release date???
Well, thanks all of you who take the time to respond, although my question wasn't answered it shows me the big picture on this matter. Since the silence of Digium and the *NOW list I think this release will not arrive before new year! Thanks again... Raul -- Linux Counter #156439 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
On Wednesday 19 December 2007 07:40:21 James Collier wrote: I think it should be core dogs show black. No, that violates the pattern. dogs is not a verb. core show black dogs or dogs show black would be the correct form. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change sendmail return path
On Wednesday 19 December 2007 01:11:25 Johansson Olle E wrote: 19 dec 2007 kl. 01.07 skrev shadowym: Unfortunately that only changes the from field. So if you were to reply to the email that is the one Outlook would use. The receiving mail system looks at the return path in the header of the email to determine if it is valid. serveremail and fromstring do not change that. Again, the return path in the email is set to [EMAIL PROTECTED]. I can easily change mydomain.com in sendmail but cannot figure out how to change asterisk. Sendmail has a notion of trusted users that are allowed to change the envelope sender's address. Your Asterisk process userid propably does not belong to that group. Add it to the group in the wonderfully elegant and simple sendmail configuration and change the mailcommand in voicemail.conf so that you specify another sender. The line to do this is Tasterisk in sendmail.cf or in sendmail.mc: define(`confTRUSTED_USERS', `asterisk')dnl -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Tilghman Lesher wrote: On Wednesday 19 December 2007 07:40:21 James Collier wrote: I think it should be core dogs show black. No, that violates the pattern. dogs is not a verb. core show black dogs or dogs show black would be the correct form. Could this CLI syntax move over to the dev list, since it's mobing further away from the original question! /M ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
On Wed, 2007-12-19 at 08:33 -0600, Tilghman Lesher wrote: On Wednesday 19 December 2007 07:40:21 James Collier wrote: I think it should be core dogs show black. No, that violates the pattern. dogs is not a verb. core show black dogs or dogs show black would be the correct form. Sorry but I'm not a native English speaker and I don't get it. Why is dogs show black the correct form as opposed to the imho more correct (in spoken language) show black dogs? Regards, Patrick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Download Asterisk
Doug Lytle wrote: Hey everybody, I just went to download and test the current version of Asterisk, go tired of the redirect impeded in the header and sat down for a few Should have read got tired of the embedded redirect. This is why you shouldn't send to the list when you're on vicodin with an abscessed tooth. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk with alsa
On Wed, 19 Dec 2007, Jerry Geis wrote: It seems as though asterisk opens the sound port exclusively. I need it to share the sound card. As far as I know, the ability to mux audio streams is a function of whether the ALSA driver for the particular sound adaptor supports it. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Realtime SIP rtcachefriends
I haven't been able to find this on the wiki: If rtcachefriends=yes. When will a change to a realtime user/peer take effect? Next registration? Never? It's also not clear to me what the purposes of rtautoclear and ignoreregexpire are. The only info I have found is the comments in the sample config file. Sounds like rtautoclear will save memory if I have lots of peers. Is there any other reason to enable it? And why would I want to ignore the fact that a registration is expired? It seems if registration is expired and they don't reregister then the SIP device is probably not there anymore. Correct? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
On Wed, 19 Dec 2007, Patrick wrote: On Wed, 2007-12-19 at 08:33 -0600, Tilghman Lesher wrote: On Wednesday 19 December 2007 07:40:21 James Collier wrote: I think it should be core dogs show black. No, that violates the pattern. dogs is not a verb. core show black dogs or dogs show black would be the correct form. Sorry but I'm not a native English speaker and I don't get it. Why is dogs show black the correct form as opposed to the imho more correct (in spoken language) show black dogs? It's not. I think it was a humorous reply to a humorous reply. The core bit should die, die, die. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using MysqlPool Application 1.4
Atis Lezdins wrote: On 12/18/07, Cyril SCETBON [EMAIL PROTECTED] wrote: Hi, Since I've upgraded to Asterisk 1.4 I can't use a MySQL database anymore for select queries :-( I'm using dbquery from MysqlPool Application 1.4 and selecting something from a table returns nothing even if I try to do a query like SELECT 1; Is anyone in the same troubles ? Do you advice me another solution to connect to my database ? app_addon_sql_mysql from asterisk-addons - it works fine for me. I have to compile it cause it's not ported to ubuntu gutsy :-( It may be available in hardy... Regards, Atis -- Cyril SCETBON ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT] Re: Upgrade to Asterisk 1.4 - it's one year's old!
Steve Edwards wrote: On Wed, 19 Dec 2007, Patrick wrote: On Wed, 2007-12-19 at 08:33 -0600, Tilghman Lesher wrote: On Wednesday 19 December 2007 07:40:21 James Collier wrote: I think it should be core dogs show black. No, that violates the pattern. dogs is not a verb. core show black dogs or dogs show black would be the correct form. Sorry but I'm not a native English speaker and I don't get it. Why is dogs show black the correct form as opposed to the imho more correct (in spoken language) show black dogs? It's not. I think it was a humorous reply to a humorous reply. The core bit should die, die, die. die like in perl -e 'die; die; die; my $darling;' ? ;) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
On Wednesday 19 December 2007 09:31:02 Patrick wrote: On Wed, 2007-12-19 at 08:33 -0600, Tilghman Lesher wrote: On Wednesday 19 December 2007 07:40:21 James Collier wrote: I think it should be core dogs show black. No, that violates the pattern. dogs is not a verb. core show black dogs or dogs show black would be the correct form. Sorry but I'm not a native English speaker and I don't get it. Why is dogs show black the correct form as opposed to the imho more correct (in spoken language) show black dogs? Because the form is always section verb arguments, so dogs is the section, show is the verb, and black is the argument. I may not have come up with the convention, but I have faithfully enforced the convention, mainly for consistency. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX for asterisk to asterisk
On Wed, 2007-12-19 at 19:53 +0500, Rizwan Hisham wrote: Hi, I am not new to asterisk but this is the first time im using iax protocol. I have always used sip before, but i have heard its better to use iax if i want communication between 2 asterisk servers. A no authority found message found would indicate that you don't have an IAX user setup properly on the box being called. I just registered asterisk server1 with asterisk server2 and tried to call server1 from 2 but the call does not pass thru. Registration only tells server2 where to send calls to server1 -- it has absolutely nothing to do with server2 *accepting* calls from server1. For that to happen, you need a user section on server2 which server1 will authenticate against. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
On Wed, 19 Dec 2007, MatsK wrote: Steve Edwards wrote: On Wed, 19 Dec 2007, Patrick wrote: On Wed, 2007-12-19 at 08:33 -0600, Tilghman Lesher wrote: On Wednesday 19 December 2007 07:40:21 James Collier wrote: I think it should be core dogs show black. No, that violates the pattern. dogs is not a verb. core show black dogs or dogs show black would be the correct form. Sorry but I'm not a native English speaker and I don't get it. Why is dogs show black the correct form as opposed to the imho more correct (in spoken language) show black dogs? It's not. I think it was a humorous reply to a humorous reply. Please move this discussion away from this thread. Read Olles reply, that that has been discussed in the dev list so take it over there I disagree. The discussion has moved off-topic from O's question, so a new thread is appropriate, but I do think discussing what the user interface should look like belongs on the user list. We're not discussing code or the inner workings of Asterisk or even changing the functionality of Asterisk, just what the proper order of the words should be. Most of us users are people, not parsers. The developers? Well, that's why they're developers :) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] shoreline IP100 aka Polycom 500 boot problem
my client purchased a couple of shoreline ip-100 phones... I managed to get them to Not boot up... shows the polycom logo then goes blank... looks like the want mcgp... oh, mgcp... is there a solution for this? besides sending it back to polycom? daveC ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording on Hanup
What makes you decide wehter or not you want to keep the recorded file? Is the fact that the user hangup the call before the 30 seconds or the fact that he really talked? As far as I know current version of asterisk doesn't allow you to detect who end up the call. Of course you may use some tricks for these, I mean you may set the IVR to record a bit more than 30 seconds, then when the call hangs up when you reach the h extension in you diaplan you may check the answered time of the call. If your call has an answered time duration lower than 30 seconds, for sure was the caller who hangup the call. Hope it helps. On Dec 18, 2007 8:04 PM, Jamshed Zaidi [EMAIL PROTECTED] wrote: yes, the senario is this when user gets a call IVR starts playing and after hearing beep user starts recording message for 30 seconds(call duration is for 30 seconds). What i want is During 30 seconds if user does hangup his/her call then message should be recorded otherwise(after timeout) message is discarded. Is there any thing that will help me...??? currently I am doing the same thing on pressing 1 with php agi script and its working fine. On 12/19/07, Marco Mouta [EMAIL PROTECTED] wrote: What do you mean with record a call on hangup? If the calling party ends the call you want to keep recorded file? On Dec 18, 2007 6:27 PM, Jamshed Zaidi [EMAIL PROTECTED] wrote: Hello everyone out there, I am having a problem in call recording with php agi library. I have already recorded voice after playing an IVR, to accept the recording user need to press one. but I need to record a call on hangup, Is there any way to do it. Currently i am using record_file() function in php. Is there any way to record voice by using record_file() function with hangup. can anyone helps me in resolving this problem ??? -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] shoreline IP100 aka Polycom 500 boot problem
Load the sip on it and you good to go ... assuming the phones are ok ... _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dave cantera Sent: Wednesday, December 19, 2007 12:27 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] shoreline IP100 aka Polycom 500 boot problem my client purchased a couple of shoreline ip-100 phones... I managed to get them to Not boot up... shows the polycom logo then goes blank... looks like the want mcgp... oh, mgcp... is there a solution for this? besides sending it back to polycom? daveC ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Steve Edwards wrote: On Wed, 19 Dec 2007, Patrick wrote: On Wed, 2007-12-19 at 08:33 -0600, Tilghman Lesher wrote: On Wednesday 19 December 2007 07:40:21 James Collier wrote: I think it should be core dogs show black. No, that violates the pattern. dogs is not a verb. core show black dogs or dogs show black would be the correct form. Sorry but I'm not a native English speaker and I don't get it. Why is dogs show black the correct form as opposed to the imho more correct (in spoken language) show black dogs? It's not. I think it was a humorous reply to a humorous reply. The core bit should die, die, die. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Please move this discussion away from this thread. Read Olles reply, that that has been discussed in the dev list so take it over there ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk with alsa
/ It seems as though asterisk opens the sound port exclusively. // I need it to share the sound card. / As far as I know, the ability to mux audio streams is a function of whether the ALSA driver for the particular sound adaptor supports it. Well when I run aplay --nonblock file.wav - 2 at a time both wave files play. However when I run asterisk and then try aplay --nonblock file.wav it wont play at all. So asterisk is binding the port. How can I keep it from binding the port? Jerry ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
We're not discussing code or the inner workings of Asterisk or even changing the functionality of Asterisk, just what the proper order of the words should be. Most of us users are people, not parsers. The developers? Well, that's why they're developers :) Thanks in advance, We are discussing the inner workings of Asterisk as this is an Asterisk thread. With that in mind, we are also discussing the order that a program works best in parsing code. The real reason that programmers use languages (like C or perl) is that machines are less intelligent than humans. If we used English to program computers, the computer would have to read the slight nuances that exist in English and just like this thread, we would be asking mathematical machines to make assumptions about what each say. Who is to say what variant of the English language is to be used, because people may still not understand the syntax of language we use. That being said, ordering in a command structure should make sense to the application (less intelligent entity), not to the programmer (hopefully more intelligent). Anyone who has configured most applications would agree that they are more of a programming language than a conversational language. The Asterisk core program doesn't know what verbs each module, channel, res, or function contains. It must ask the code(noun), for a given verb (function) and then pass that function the options (adjectives). So if I use show black dogs, with dogs being the module, show being the verb, and black being the option, here is what would happen: Look for module show - doesn't exist Look for module black - doesn't exist Look for module dogs - Found, get reference Ask module dogs, for function show - found, get reference Send option black (remaining words from the parser) to function show in module black. In my opinion that makes Asterisk slow and introduces bugs if some programmer creates a new app_black which causes a video screen to go black, then we have a problem. In this example, we are left with fixing the position of the module as position 3 in the command stack. That also means that additional parameters (options) must limited to one (which doesn't work) or messes with the command structure by placing adjectives after the noun like: show black dogs dachshund That doesn't make any sense for humans again. So then for the computer, we are left with the following syntax that works: Module Function Option1, Option2, The module is fixed, position 1. The function is fixed, position 2. The options are everything that follows. In our example, that would be: dogs show black Is it English? No but it isn't Spanish, Italian, and whatever language I have left out. It is Asterisk and computers. It also means profitable employment for people willing to learn this language. We could fix the verbs that are used, but that means that every module would have to have the same core verbs and we could have no exceptions. That means that ZAP, SIP, and MeetMe could have no functions that adhere outside the standards OR that most modules would have huge amounts of unnecessary functions which do nothing but take up space and cause bugs. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/iaxclient IAX2 source port
Why not let the softphones register to the closest asterisk box and use dundi to route the calls to the box where the softphone is registered ? Not exactly sure how dundi would solve this issue. How does a softphone configured to connect to sitea.asterisk.server connect to siteb.asterisk.server automagically when it's at siteb? We can't just configure the softphones to connect to asterisk because they also need to work when the softphone is simply out in the world not at any site. In the end, the system we have works quite well and we're not really interested in the complexity of moving to dundi unless there's no other way or a very compelling reason to do so. (Glad it works well for you though) The NAT issue is a serious one for us that really does seem to be an oversight in the design/implementation of IAX2. Surely there's a way to tell asterisk to use an ephemeral source port for its outbound IAX2 connections... Chris ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording on Hanup
sure it will really help me thanx for responding. On 12/19/07, Marco Mouta [EMAIL PROTECTED] wrote: What makes you decide wehter or not you want to keep the recorded file? Is the fact that the user hangup the call before the 30 seconds or the fact that he really talked? As far as I know current version of asterisk doesn't allow you to detect who end up the call. Of course you may use some tricks for these, I mean you may set the IVR to record a bit more than 30 seconds, then when the call hangs up when you reach the h extension in you diaplan you may check the answered time of the call. If your call has an answered time duration lower than 30 seconds, for sure was the caller who hangup the call. Hope it helps. On Dec 18, 2007 8:04 PM, Jamshed Zaidi [EMAIL PROTECTED] wrote: yes, the senario is this when user gets a call IVR starts playing and after hearing beep user starts recording message for 30 seconds(call duration is for 30 seconds). What i want is During 30 seconds if user does hangup his/her call then message should be recorded otherwise(after timeout) message is discarded. Is there any thing that will help me...??? currently I am doing the same thing on pressing 1 with php agi script and its working fine. On 12/19/07, Marco Mouta [EMAIL PROTECTED] wrote: What do you mean with record a call on hangup? If the calling party ends the call you want to keep recorded file? On Dec 18, 2007 6:27 PM, Jamshed Zaidi [EMAIL PROTECTED] wrote: Hello everyone out there, I am having a problem in call recording with php agi library. I have already recorded voice after playing an IVR, to accept the recording user need to press one. but I need to record a call on hangup, Is there any way to do it. Currently i am using record_file() function in php. Is there any way to record voice by using record_file() function with hangup. can anyone helps me in resolving this problem ??? -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using * in extension name
I am trying to setup an extension of *7XXX that will allow me to dial *7 and then any extension and use the Pickup application to pickup a ringing phone. Ideally it will also check if the phone is ringing somehow and then either dial the extension or pick it up if it is ringing. But I can't get that far. If I use *7268 specially it works fine, but as soon as I introduce any wild card char (X, N, Z, !, .) it responds with 404 and nothing is logged. Can * be used in this manner in a dialplan? If so then any suggestions on what I can check to see why it is doing this? If not, does anybody have a better suggestion for me? I'd rather not use a regular digit as the begin code. Daniel Hazelbaker ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using * in extension name
Daniel Hazelbaker wrote: I am trying to setup an extension of *7XXX that will allow me to dial *7 and then any extension and use the Pickup application to pickup a ringing phone. Ideally it will also check if the phone is ringing somehow and then either dial the extension or pick it up if it is ringing. But I can't get that far. If I use *7268 specially it works fine, but as soon as I introduce any wild card char (X, N, Z, !, .) it responds with 404 and nothing is logged. Can * be used in this manner in a dialplan? If so then any suggestions on what I can check to see why it is doing this? If not, does anybody have a better suggestion for me? I'd rather not use a regular digit as the begin code. Make sure you have _*7XXX The leading underscore introduces a pattern. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using * in extension name
ringing. But I can't get that far. If I use *7268 specially it works fine, but as soon as I introduce any wild card char (X, N, Z, !, .) it responds with 404 and nothing is logged. Obvious suggestion, but did you prepend the extension with an underscore to tell asterisk you wanted a pattern match? To wit (assuming you're using extension.conf and not AEL, but AEL is similar): exten = *7268,1,NoOp will work for dialing *7268. But: exten = *76XX,1,NoOp won't ever match anything because you've not told asterisk you're asking for a pattern. But: exten = _*76XX,1,NoOp should do what you're after. Chris ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using * in extension name
Daniel Hazelbaker wrote: I am trying to setup an extension of *7XXX that will allow me to dial *7 and then any extension and use the Pickup application to pickup a ringing phone. Ideally it will also check if the phone is ringing somehow and then either dial the extension or pick it up if it is ringing. But I can't get that far. If I use *7268 specially it works fine, but as soon as I introduce any wild card char (X, N, Z, !, .) it responds with 404 and nothing is logged. Can * be used in this manner in a dialplan? If so then any suggestions on what I can check to see why it is doing this? If not, does anybody have a better suggestion for me? I'd rather not use a regular digit as the begin code. Daniel Hazelbaker You have to start with an underscore character, '_', to make wildcard matching work. So your extension should be _*7XXX Mike Clark ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using * in extension name
Daniel Hazelbaker wrote:
I am trying to setup an extension of *7XXX that will allow me to dial
*7 and then any extension and use the Pickup application to pickup a
ringing phone. Ideally it will also check if the phone is ringing
somehow and then either dial the extension or pick it up if it is
ringing.
I'm not sure if it's possible to do anything in the dialplan
after Pickup() even if it didn't work, but you might try
something like this:
_*7. = {
Pickup(${EXTEN:[EMAIL PROTECTED]);
Dial(Local/${EXTEN:2});
}
I find your idea a bit unusal anyway because when Pickup()'ing
a call I almost never want to talk to the original callee.
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? - http://www.das-asterisk-buch.de
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
___
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Re: [asterisk-users] Using * in extension name
(Hope you don't mind me replying to the list) Okay, time for me to feel stupid. Yes I was forgetting to start with an underscore. Somehow while I was looking at all the examples it never clicked that that should be there. :P Daniel On Dec 19, 2007, at 11:34 AM, Martin Smith wrote: If you could show us what you ARE using (the full line of the dialplan), I might be able to offer more suggestions. For example, with the full line, I might notice if you're starting the pattern with the underscore, which I believe is required to pattern match *at all*. Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Hazelbaker Sent: Wednesday, December 19, 2007 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Using * in extension name I am trying to setup an extension of *7XXX that will allow me to dial *7 and then any extension and use the Pickup application to pickup a ringing phone. Ideally it will also check if the phone is ringing somehow and then either dial the extension or pick it up if it is ringing. But I can't get that far. If I use *7268 specially it works fine, but as soon as I introduce any wild card char (X, N, Z, !, .) it responds with 404 and nothing is logged. Can * be used in this manner in a dialplan? If so then any suggestions on what I can check to see why it is doing this? If not, does anybody have a better suggestion for me? I'd rather not use a regular digit as the begin code. Daniel Hazelbaker ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problems chanskype on ubuntu gutsy
Hi all, I'm trying to install chanskype (http://www.chanskype.com) on an Ubuntu gutsy machine with 2.6.22-14-generic kernel. First I got some compile errors, include linux?capability.h in main.c fixed those errors. Now I'll try to load theier module, but got the following error: insmod: error inserting './ivcs.ko': -1 Unknown symbol in module In dmesg I find: [3123904.331682] ivcs: Unknown symbol malloc_sizes My searches on google don't give me any usefull result. Has someone any id or suggestions? I'm using the latest version of chanskype available on their website. there is only a bin foor ubunut 6.x. Thanks in advance, kr -Peter signature.asc Description: Digital signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.16.1
Oh, Asterisk 1.4.16.1 is available for download. :) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bulk Reverse Phone Lookup
Is anyone aware of a service where we can lookup phone numbers to determine a name and/or name + address available in bulk? We want to look up every number called to our call center, so it will be tens of thousands per day. Services that charge 3 to 5 cents per lookup will get way too expensive very quickly. Thus, I'm looking for a service that can either license a database or provide bulk lookups for maybe $300-$500/mo? Or even license a database for a few grand. Anyone know of something like this? -Norman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.16.1
On 22:00, Wed 19 Dec 07, Philipp Kempgen wrote: Oh, Asterisk 1.4.16.1 is available for download. :) 1.4.16.1 fixes a segfault in chan_iax which was introduced in the fix to the security issue in chan_iax which was the reason to release 1.4.16 in the first place :) Just to let people know why 1.4.16.1 was released. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] shoreline IP100 aka Polycom 500 boot problem
if that's all they do, then it doesn't sound good. There is a way to do a system format when they first load up, but the keys you hold down escapes me at this moment and I seem to remember it took some searching to find. I'll see if I can track it down. On Dec 19, 2007 11:56 AM, Robert Augustyn [EMAIL PROTECTED] wrote: Load the sip on it and you good to go ... assuming the phones are ok ... -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *dave cantera *Sent:* Wednesday, December 19, 2007 12:27 PM *To:* [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] shoreline IP100 aka Polycom 500 boot problem my client purchased a couple of shoreline ip-100 phones... I managed to get them to Not boot up... shows the polycom logo then goes blank... looks like the want mcgp... oh, mgcp... is there a solution for this? besides sending it back to polycom? daveC ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Since we're WAY OT anyway Tony Plack wrote: That being said, ordering in a command structure should make sense to the application (less intelligent entity), not to the programmer (hopefully more intelligent). If that were true then we really should be writing our dialplans in binary machine code, that is what that dumb computers REALLY understand. Fortunately, it's not true. We can take advantage of a GOOD programmer's skill to have the computer do the grunt work of converting something real people understand into machine code. We call the product of this process a High-Level Programming Language. A well-written application should attempt to minimize the amount of 'conversion' the user/programmer has to do. Therefore the command structure SHOULD be in a form that is natural for the user/programmer, NOT to the machine. Personally, I would vote for show dogs colour black but maybe I've spent too much time with Cisco's IOS! :-) regards, Drew PS. There does seem to be an assumption that programmers are intelligent, I'm not sure that this is a defensible position. ;-) -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] shoreline IP100 aka Polycom 500 boot problem
hold 468* or 135 when unit is booting... or not booting... hold those down when you first try to start the phone. then load the firmware/bootrom/etc On Dec 19, 2007 3:55 PM, Joe [EMAIL PROTECTED] wrote: if that's all they do, then it doesn't sound good. There is a way to do a system format when they first load up, but the keys you hold down escapes me at this moment and I seem to remember it took some searching to find. I'll see if I can track it down. On Dec 19, 2007 11:56 AM, Robert Augustyn [EMAIL PROTECTED] wrote: Load the sip on it and you good to go ... assuming the phones are ok ... -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *dave cantera *Sent:* Wednesday, December 19, 2007 12:27 PM *To:* [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] shoreline IP100 aka Polycom 500 boot problem my client purchased a couple of shoreline ip-100 phones... I managed to get them to Not boot up... shows the polycom logo then goes blank... looks like the want mcgp... oh, mgcp... is there a solution for this? besides sending it back to polycom? daveC ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime logic in Asterisk 1.4.16.1
Hello, I have configured one provider in Asterisk Realtime DB without username and password, only host=providers_IP and ipaddress=providers_IP Now when I'm trying to send call using this provider I'm using following string: Dial(SIP/[EMAIL PROTECTED]) In Asterisk 1.4.15 debug I see that Realtime engine is using query: [Dec 20 00:02:15] DEBUG[14634]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' to retrieve info about this device. And in Asterisk 1.4.16.1 I see: [Dec 20 00:04:12] DEBUG[25686]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' AND host = 'dynamic' Note: host = 'dynamic' Where this came from? In mine DB host=providers_IP, how Asterisk managed to visualize that it should be dynamic?! Offcourse I get: [Dec 20 00:05:58] WARNING[25686]: chan_sip.c:2898 create_addr: No such host: Provider [Dec 20 00:05:58] WARNING[25686]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) Because Realtime Engine is not able to find my Provider which is NOT DYNAMIC! No other settings changed. Same configuration files. res_config_mysql.so recompiled to 1.4.16.1. Please help or explain what's wrong! Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1
On 00:12, Thu 20 Dec 07, Mindaugas Kezys wrote: Hello, I have configured one provider in Asterisk Realtime DB without username and password, only host=providers_IP and ipaddress=providers_IP Now when I'm trying to send call using this provider I'm using following string: Dial(SIP/[EMAIL PROTECTED]) In Asterisk 1.4.15 debug I see that Realtime engine is using query: [Dec 20 00:02:15] DEBUG[14634]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' to retrieve info about this device. And in Asterisk 1.4.16.1 I see: [Dec 20 00:04:12] DEBUG[25686]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' AND host = 'dynamic' Note: host = 'dynamic' Where this came from? In mine DB host=providers_IP, how Asterisk managed to visualize that it should be dynamic?! Offcourse I get: [Dec 20 00:05:58] WARNING[25686]: chan_sip.c:2898 create_addr: No such host: Provider [Dec 20 00:05:58] WARNING[25686]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) Because Realtime Engine is not able to find my Provider which is NOT DYNAMIC! No other settings changed. Same configuration files. res_config_mysql.so recompiled to 1.4.16.1. Please help or explain what's wrong! Have a look at http://downloads.digium.com/pub/security/AST-2007-027.pdf That's why it's not working anymore -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT] black dogs (was: Re: Upgrade to Asterisk 1.4 - it's one year's old!)
Drew Gibson wrote: A well-written application should attempt to minimize the amount of 'conversion' the user/programmer has to do. Therefore the command structure SHOULD be in a form that is natural for the user/programmer, NOT to the machine. Personally, I would vote for show dogs colour black but maybe I've spent too much time with Cisco's IOS! :-) show me all the colored dogs now. hurry up! don't spent any time in those other threads. and while you're at it would you please fix that stupid mistake i made in the config file Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/iaxclient IAX2 source port
Michiel, I'm still confused as to how this would help me. Nevertheless, I'm curious: Why not let the softphones register to the closest asterisk box and use dundi to route the calls to the box where the softphone is registered ? How do you get around having to synchronize the softphone reg info between sites? Or is there some DUNDi magic that would allow a softphone from site-A to register on site-B's asterisk box without site-B having a local entry in iax.conf for that softphone? Chris ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] noun-verb vs verb-noun aka dogs black vs black dogs
Wow. I wasn't expecting such a voluminous reply -- some I agree with and some I don't. My apologies for an equally voluminous reply. On Wed, 19 Dec 2007, Tony Plack wrote: We're not discussing code or the inner workings of Asterisk or even changing the functionality of Asterisk, just what the proper order of the words should be. Most of us users are people, not parsers. The developers? Well, that's why they're developers :) We are discussing the inner workings of Asterisk as this is an Asterisk thread. With that in mind, we are also discussing the order that a program works best in parsing code. I disagree. I can discuss the placement of the controls in my car without the slightest interest in whatever is in front of the firewall. This is not the inner workings, just the external presentation. The real reason that programmers use languages (like C or perl) is that machines are less intelligent than humans. If we used English to program computers, the computer would have to read the slight nuances that exist in English and just like this thread, we would be asking mathematical machines to make assumptions about what each say. Who is to say what variant of the English language is to be used, because people may still not understand the syntax of language we use. Who is to say is the developers since it is their gift that we use. My only hope is to influence their decisions to what I think will suit me and the majority of whom I perceive to be the users. If I don't like it, I know my choices -- I can choose not to use their product, I can adapt to their decisions, I can suffer in silence, I can ask for consensus for change, I could even write a humongous patch to coerce their code to fit my view of the world. That being said, ordering in a command structure should make sense to the application (less intelligent entity), not to the programmer (hopefully more intelligent). Anyone who has configured most applications would agree that they are more of a programming language than a conversational language. I disagree. I think the command structure should make sense to me. I'm pretty sure the engineers that designed my car placed the air conditioner controls comfortably within my reach because users before me said that's where they wanted them to be. Otherwise, we might find the air conditioner control under the hood next to the compressor because it used less wire. I think the command structure should follow widely accepted patterns -- MySQL, GDB, Oracle, ad nauseum. Think how much fun it would be if Chevy engineers thought the brake should be under your right foot and Ford engineers thought the gas should be there. I'm sure both camps would have rational arguments. Now if we could just get the Brits on the right side of the road... The Asterisk core program doesn't know what verbs each module, channel, res, or function contains. It must ask the code(noun), for a given verb (function) and then pass that function the options (adjectives). You are mistaken. When a module is loaded it registers as many CLI objects as it wants to, so the core does know what commands are available. So if I use show black dogs, with dogs being the module, show being the verb, and black being the option, here is what would happen: Look for module show - doesn't exist Look for module black - doesn't exist Look for module dogs - Found, get reference Ask module dogs, for function show - found, get reference Send option black (remaining words from the parser) to function show in module black. This is only because of how you think it works. It doesn't work that way and I don't really care how it works. I just know how I want it to look. It would be perfectly reasonable for a module to register cli objects with a show command handler. Look at the code in chan_sip.c. The module registers multiple objects with sip and show as the first 2 words in the command. You are thinking way too rigidly. Asterisk does not need to know a noun from a verb, just what words can follow which words and how to complete the command (as in show channel TAB). In my opinion that makes Asterisk slow and introduces bugs if some programmer creates a new app_black which causes a video screen to go black, then we have a problem. And that is what it is -- an opinion. I don't see a problem here. A well defined, consistent command structure would ease the addition and adoption of new apps, not hinder it. In this example, we are left with fixing the position of the module as position 3 in the command stack. That also means that additional parameters (options) must limited to one (which doesn't work) or messes with the command structure by placing adjectives after the noun like: No. we don't need to fix anything in any position. The command parser only needs to know what words can follow the current word. show black dogs dachshund That doesn't make any sense for humans again. Your straw man
Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Thursday, December 20, 2007 12:22 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On 00:12, Thu 20 Dec 07, Mindaugas Kezys wrote: Hello, I have configured one provider in Asterisk Realtime DB without username and password, only host=providers_IP and ipaddress=providers_IP Now when I'm trying to send call using this provider I'm using following string: Dial(SIP/[EMAIL PROTECTED]) In Asterisk 1.4.15 debug I see that Realtime engine is using query: [Dec 20 00:02:15] DEBUG[14634]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' to retrieve info about this device. And in Asterisk 1.4.16.1 I see: [Dec 20 00:04:12] DEBUG[25686]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' AND host = 'dynamic' Note: host = 'dynamic' Where this came from? In mine DB host=providers_IP, how Asterisk managed to visualize that it should be dynamic?! Offcourse I get: [Dec 20 00:05:58] WARNING[25686]: chan_sip.c:2898 create_addr: No such host: Provider [Dec 20 00:05:58] WARNING[25686]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) Because Realtime Engine is not able to find my Provider which is NOT DYNAMIC! No other settings changed. Same configuration files. res_config_mysql.so recompiled to 1.4.16.1. Please help or explain what's wrong! Have a look at http://downloads.digium.com/pub/security/AST-2007-027.pdf That's why it's not working anymore -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? --- Thank you for pointing this, but I red this doc many times. It does not help. I tried to put username/password for my device - but it still is looking for dynamic. Does it mean I can't have anything else in host field for device except dynamic? Also this PDF states: An attacker may impersonate any user using host-based authentication without a secret, simply by guessing the username of that user. AFAIK host-based authentication is done by IP address. Username and password are not present. Following this I see no logic in above statements: host-based authentication without a secret - host-based auth. is always WITHOUT secret, and simply by guessing the username of that user - again - host-based auth. is always WITHOUT username If device (peer/user) has username/password - that's not HOST-BASED authentication. Correct me if I'm wrong. Question follows - how can I have host-based authentication in Realtime in Asterisk 1.4.16.1?? Maybe tommorow we will see Asterisk 1.4.16.2? Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsteriskNOW release date???
On Tue, 2007-12-18 at 11:16 -0400, Raúl Gómez C. wrote: Anyone knows about the date of the official (stable) release (v1.0) of AsteriskNOW??? It's supposed to be at the end of this year, which is very close now with no signs of it. I just got off the phone with the software product manager who is over AsteriskNOW, and have it on good authority that it will be released in early January. There were a few last-minute things that weren't quite ready, but as of today things appear to be back on track and almost ready for release. --- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] noun-verb vs verb-noun aka dogs black vs black dogs
On Wed, Dec 19, 2007 at 02:47:39PM -0800, Steve Edwards wrote: This only works because you are closed to the alternative. The alternative (verb-noun) works fine for the above referenced applications and many more. Do you want to tally the number of users of applications that use noun-verb instead of verb-noun? Is there a reason verb-noun works fine for them and not for us? OK, here's a small usability test to your idea: Here's a partial list of actions from asterisk 1.4. Which of them is supported by your hypothetical MGCP device? (no cheating, please) active add answer audit autoanswer boost clear convert del deltree dial dumphtml flash get hangup logoff mute put reload remove save send set show showkey transfer unmute -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1
On Wednesday 19 December 2007 16:12:27 Mindaugas Kezys wrote: [Dec 20 00:04:12] DEBUG[25686]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' AND host = 'dynamic' Note: host = 'dynamic' Correct, that's the FIRST lookup that is done. It then checks the IP address and does: SELECT * FROM devices WHERE name = 'Provider' AND host='23.45.67.89' where the IP address is what is sent in the SIP INVITE. If that fails, it does a lookup only on the name (old behavior). If that fails: SELECT * FROM devices WHERE host='23.45.67.89' AND port='5060' If that fails: SELECT * FROM devices WHERE ipaddr='23.45.67.89' AND port='5060' If that fails: SELECT * FROM devices WHERE host='23.45.67.89' and checks every match for insecure=yes If that fails: SELECT * FROM devices WHERE ipaddr='23.45.67.89' and checks every match for insecure=yes And if that fails, then it returns no match. So all of those queries had to run and fail for you to get no match. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using * in extension name
Just to finish off this thread for anybody else who wants the same
functionality:
As Philipp coded out, what I was originally doing was something akin
to the following:
_*7. = {
Pickup(${EXTEN:[EMAIL PROTECTED]);
Dial(Local/${EXTEN:2});
}
Somebody else wondered why I would do this as I wouldn't want to talk
to the original callee. Pickup() will never return if it successfully
picks up the call. So if it does return then we can assume (for now)
that the phone is not ringing and then proceed to dial the number.
As I said, this did work perfectly. However, I discovered that the
Grandstream phones we use automatically prepend ** to the monitored
extension. So if the phone is not ringing it will dial 268, if the
phone is ringing it will dial **268. So in the end I ended up with:
_**. = { Pickup(${EXTEN:[EMAIL PROTECTED]); }
And all works perfectly. Obviously for other phones I would have to
come up with something else like the above, but with the grandstreams
it seems to work great.
Daniel
On Dec 19, 2007, at 11:48 AM, Daniel Hazelbaker wrote:
(Hope you don't mind me replying to the list)
Okay, time for me to feel stupid. Yes I was forgetting to start
with an underscore. Somehow while I was looking at all the examples
it never clicked that that should be there. :P
Daniel
On Dec 19, 2007, at 11:34 AM, Martin Smith wrote:
If you could show us what you ARE using (the full line of the
dialplan),
I might be able to offer more suggestions. For example, with the full
line, I might notice if you're starting the pattern with the
underscore,
which I believe is required to pattern match *at all*.
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Daniel Hazelbaker
Sent: Wednesday, December 19, 2007 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Using * in extension name
I am trying to setup an extension of *7XXX that will allow me
to dial
*7 and then any extension and use the Pickup application to pickup a
ringing phone. Ideally it will also check if the phone is ringing
somehow and then either dial the extension or pick it up if it is
ringing. But I can't get that far. If I use *7268 specially
it works
fine, but as soon as I introduce any wild card char (X, N, Z,
!, .) it
responds with 404 and nothing is logged.
Can * be used in this manner in a dialplan? If so then any
suggestions on what I can check to see why it is doing this?
If not,
does anybody have a better suggestion for me? I'd rather not use a
regular digit as the begin code.
Daniel Hazelbaker
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Re: [asterisk-users] How to change sendmail return path
Thanks for the info Tilghman, I had high hopes for this solution for unfortunately it's not working. Did exactly as you specified but return path is still [EMAIL PROTECTED] even though [EMAIL PROTECTED] in voicemail.conf :( -Original Message- From: Tilghman Lesher [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 19, 2007 6:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to change sendmail return path On Wednesday 19 December 2007 01:11:25 Johansson Olle E wrote: 19 dec 2007 kl. 01.07 skrev shadowym: Unfortunately that only changes the from field. So if you were to reply to the email that is the one Outlook would use. The receiving mail system looks at the return path in the header of the email to determine if it is valid. serveremail and fromstring do not change that. Again, the return path in the email is set to [EMAIL PROTECTED]. I can easily change mydomain.com in sendmail but cannot figure out how to change asterisk. Sendmail has a notion of trusted users that are allowed to change the envelope sender's address. Your Asterisk process userid propably does not belong to that group. Add it to the group in the wonderfully elegant and simple sendmail configuration and change the mailcommand in voicemail.conf so that you specify another sender. The line to do this is Tasterisk in sendmail.cf or in sendmail.mc: define(`confTRUSTED_USERS', `asterisk')dnl -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] noun-verb vs verb-noun aka dogs black vs black dogs
On Wednesday December 19 2007 6:09 pm, Tzafrir Cohen wrote: On Wed, Dec 19, 2007 at 02:47:39PM -0800, Steve Edwards wrote: This only works because you are closed to the alternative. The alternative (verb-noun) works fine for the above referenced applications and many more. Do you want to tally the number of users of applications that use noun-verb instead of verb-noun? Is there a reason verb-noun works fine for them and not for us? OK, here's a small usability test to your idea: Here's a partial list of actions from asterisk 1.4. Which of them is supported by your hypothetical MGCP device? (no cheating, please) active add answer audit autoanswer boost clear convert del deltree dial dumphtml flash get hangup logoff mute put reload remove save send set show showkey transfer unmute Okay I have to put my 2 cents in now can't resist any longer even though it may only be worth 0.5 cents. In MY opinion, consistency is first and formost. I can learn almost any command struture IF i put my mind to it and I want to do so. What is hard for me is changing in mid stream. having said that I always liked a drill down structure. Big idea first, followed by category of idea, followed by.. and so on till you get the the exact single item that you are looking for. A US based example: show world north_america us state nh capitol Gives: Concord You could easily do : show world giving all the continents show world north_america giving all countries in North America and so on down the line. To ME and maybe only me, this make since, object world knows of continents, object continents knows of countries, object countries knows of state, object state knows of capitols. Easy for programmers, users and computers alike. again just my opinion. JohnM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Thursday, December 20, 2007 1:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Wednesday 19 December 2007 16:12:27 Mindaugas Kezys wrote: [Dec 20 00:04:12] DEBUG[25686]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' AND host = 'dynamic' Note: host = 'dynamic' Correct, that's the FIRST lookup that is done. It then checks the IP address and does: SELECT * FROM devices WHERE name = 'Provider' AND host='23.45.67.89' where the IP address is what is sent in the SIP INVITE. If that fails, it does a lookup only on the name (old behavior). If that fails: SELECT * FROM devices WHERE host='23.45.67.89' AND port='5060' If that fails: SELECT * FROM devices WHERE ipaddr='23.45.67.89' AND port='5060' If that fails: SELECT * FROM devices WHERE host='23.45.67.89' and checks every match for insecure=yes If that fails: SELECT * FROM devices WHERE ipaddr='23.45.67.89' and checks every match for insecure=yes And if that fails, then it returns no match. So all of those queries had to run and fail for you to get no match. -- Tilghman -- Thank you for explanation, but problem is that only this first query is executed: [Dec 20 00:04:12] DEBUG[25686]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' AND host = 'dynamic' [Dec 20 00:04:12] WARNING[25686]: chan_sip.c:2898 create_addr: No such host: Provider [Dec 20 00:04:12] WARNING[25686]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) That's it. No more queries. End of call. Why? Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] noun-verb vs verb-noun aka dogs black vs black dogs
On Thu, 20 Dec 2007, Tzafrir Cohen wrote: On Wed, Dec 19, 2007 at 02:47:39PM -0800, Steve Edwards wrote: This only works because you are closed to the alternative. The alternative (verb-noun) works fine for the above referenced applications and many more. Do you want to tally the number of users of applications that use noun-verb instead of verb-noun? Is there a reason verb-noun works fine for them and not for us? OK, here's a small usability test to your idea: Here's a partial list of actions from asterisk 1.4. Which of them is supported by your hypothetical MGCP device? (no cheating, please) active add answer audit autoanswer boost clear convert del deltree dial dumphtml flash get hangup logoff mute put reload remove save send set show showkey transfer unmute I don't have a hypothetical MGCP device. I don't even have a real MGCP device. I'm guessing MGCP is some sort of technology like IAX or SIP. These are not actions, but pieces of a jigsaw puzzle. The leaves only have meaning when they are attached to branches, Grasshopper. black dogs show doesn't mean anything, but show black dogs does. If cheating isn't allowed, I'd guess at assembling these words into commands like: show [active] mgcp channels hangup [mgcp] channel xyz set [mgcp] channel xyz boost [=] 5 But this doesn't prove or disprove anything except my ignorance of MGCP. A more meaningful exercise would be How would you phrase a command to show the configuration or status of a channel? or How would you phrase a command to set the transmit gain of a channel? I would do it like this: show [iax|mgcp|sip|zap] channel xyz [configuration|status] set channel xyz [receive|transmit] gain [=] 5 But I am open to alternatives :) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] noun-verb vs verb-noun aka dogs black vs black dogs
Tzafrir Cohen wrote: On Wed, Dec 19, 2007 at 02:47:39PM -0800, Steve Edwards wrote: This only works because you are closed to the alternative. The alternative (verb-noun) works fine for the above referenced applications and many more. Do you want to tally the number of users of applications that use noun-verb instead of verb-noun? Is there a reason verb-noun works fine for them and not for us? OK, here's a small usability test to your idea: Here's a partial list of actions from asterisk 1.4. Which of them is supported by your hypothetical MGCP device? (no cheating, please) active add answer audit autoanswer boost clear convert del deltree dial dumphtml flash get hangup logoff mute put reload remove save send set show showkey transfer unmute I agree that CURRENTLY, there's no way to make everything verb noun. However, I could completely see a requirement for all modules to register their syntax variations with a CLI interpreter, so that if you typed IN a verb, the CLI interpreter would know to which modules that verb applies because the modules have registered their verbs. That's generally the way it's done in interpreters which are V-N for everything. Of course, that might take a whole rewrite and be completely impractical. But hey it COULD be done N. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] noun-verb vs verb-noun aka dogs black vs black dogs
On Wed, 19 Dec 2007, John Millican wrote: On Wed, Dec 19, 2007 at 02:47:39PM -0800, Steve Edwards wrote: This only works because you are closed to the alternative. The alternative (verb-noun) works fine for the above referenced applications and many more. Do you want to tally the number of users of applications that use noun-verb instead of verb-noun? Is there a reason verb-noun works fine for them and not for us? In MY opinion, consistency is first and formost. I can learn almost any command struture IF i put my mind to it and I want to do so. What is hard for me is changing in mid stream. having said that I always liked a drill down structure. Big idea first, followed by category of idea, followed by.. and so on till you get the the exact single item that you are looking for. A US based example: show world north_america us state nh capitol Gives: Concord You could easily do : show world giving all the continents show world north_america giving all countries in North America and so on down the line. To ME and maybe only me, this make since, object world knows of continents, object continents knows of countries, object countries knows of state, object state knows of capitols. Easy for programmers, users and computers alike. again just my opinion. Pretty darn close. Sometimes the followed bys precede the category. So you could enter: show channel TAB and be presented with a list of all channels. Or you could enter: show sip channel TAB and be presented with a list of all SIP channels. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] noun-verb vs verb-noun aka dogs black vs black dogs
My $0.02: This definitely sounds the way to do it. I agree that the order that the words are needed is almost irrelevant, as long as the whole lot is consistent. Good example too. Cheers Daniel Cole (CCNA) Technical Support Ph: 1800 424 683 Fax: 03 5221 7659 e: [EMAIL PROTECTED] w: hugonet.com.au --- The information transmitted is the property of HugoNet and is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of the company. Any review, retransmission, dissemination and other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. P Please consider the environment before you print this e-mail or any attachments. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Millican Sent: Thursday, 20 December 2007 10:46 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] noun-verb vs verb-noun aka dogs black vs black dogs On Wednesday December 19 2007 6:09 pm, Tzafrir Cohen wrote: On Wed, Dec 19, 2007 at 02:47:39PM -0800, Steve Edwards wrote: This only works because you are closed to the alternative. The alternative (verb-noun) works fine for the above referenced applications and many more. Do you want to tally the number of users of applications that use noun-verb instead of verb-noun? Is there a reason verb-noun works fine for them and not for us? OK, here's a small usability test to your idea: Here's a partial list of actions from asterisk 1.4. Which of them is supported by your hypothetical MGCP device? (no cheating, please) active add answer audit autoanswer boost clear convert del deltree dial dumphtml flash get hangup logoff mute put reload remove save send set show showkey transfer unmute Okay I have to put my 2 cents in now can't resist any longer even though it may only be worth 0.5 cents. In MY opinion, consistency is first and formost. I can learn almost any command struture IF i put my mind to it and I want to do so. What is hard for me is changing in mid stream. having said that I always liked a drill down structure. Big idea first, followed by category of idea, followed by.. and so on till you get the the exact single item that you are looking for. A US based example: show world north_america us state nh capitol Gives: Concord You could easily do : show world giving all the continents show world north_america giving all countries in North America and so on down the line. To ME and maybe only me, this make since, object world knows of continents, object continents knows of countries, object countries knows of state, object state knows of capitols. Easy for programmers, users and computers alike. again just my opinion. JohnM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
- Original Message - From: Steve Edwards [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 19, 2007 5:43 AM Subject: Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old! On Sat, 15 Dec 2007, Johansson Olle E wrote: I wonder if there are any major obstacles for upgrading. How about the change from a bad command line interface to a really bad command line interface? I mean, Seriously? (in a Grey's Anatomy kind of way...) The old syntax was inconsistent -- show manager command vs sip show channels and just plain bad -- for example sip reload should have been reload sip. The new syntax continues down the noun-verb path instead of correcting itself and using verb-noun like most other applications (MySQL, GDB, Oracle, etc.) Then, just to make it worse, now I have to learn which commands somebody (arbitrarily) decided are core and which are not -- for what benefit? Certainly doesn't make MY job easier! Approach the command line like a noob. I want Asterisk to show me something so I'll start the command line with show. I'm not quite sure what I'm doing, so I'll press TAB to see what I can show. Oh, channel looks like what I want. Hmm, too much. Maybe I should have qualified what kind of channel I'm looking for BEFORE the word channel. Here's a suggestion -- stop thinking like a parser and start thinking like a person :) Which makes more sense (at least in English)? 1) show black dogs -- show sip channels 2) black show dogs -- sip show channels 3) dogs black show -- channels sip show 4) show dogs black -- show channels sip 5) black dogs show -- sip channels show 6) dogs show black -- channels show sip Is it too late to fix this for 1.6? Thanks in advance, I think as many people have pointed out they are used to a lot of commands out there so changing it yet again would make more people unhappy. But maybe asterisk can have both. Why not sip show channels for the old timers and show channels sip or show sip channels for the n00b's. Why shouldn't asterisk have both options ? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
I think it should stay here. Otherwise us users only (non devs) would have no input. I am personally not on the dev list. From the few times that I have posted questions on IRC I have been just taunted for my lack of knowledge on development issues. (This is a reason why I stay off the dev list). If I am just a laughing joke I don't feel like my opinion matters. (/action sniffles in corner) - Original Message - From: MatsK To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, December 19, 2007 4:59 PM Subject: Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old! Tilghman Lesher wrote: On Wednesday 19 December 2007 07:40:21 James Collier wrote: I think it should be core dogs show black. No, that violates the pattern. dogs is not a verb. core show black dogs or dogs show black would be the correct form. Could this CLI syntax move over to the dev list, since it's mobing further away from the original question! /M -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - testing
snip Our problem is that very few in the community test beta releases or development code. I want to send a big thank you to all that do, you are very important in this process. And for those of you who want to join, go to www.asterisk.org and find instructions on how to download development code for testing. Join the whoever tests this stuff group today :-) /snip Olle I would love to test but I do not know what I am looking for. I would say that I have a fairly good knowledge of Asterisk however I am not the best at tracing the root problems of issues. I have no problem of loading the bleeding edge version on a spate box, loading my current configs on it and seeing where it goes down. Maybe some info on what to look for when there are issues would help. Another thing is I have a limited amount of dial plan code that I use as well as time. Maybe people can contribute lines of dial plan that they use and we can put it on one central system. Other people can then test the functionality of what users post that their dial plan logic is supposed to do. If a test user has an issue with the way a call is supposed to work then some one else can look at where and why it is breaking. I think this is a good way for us to test each others dial plan logic, see errors etc. Of course there is a question of who will run the box, debug etc. I would have no problem donating a box for people to use in such a test scenario. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP service providers/PSTN termination points
Voipjet works for me. Also just Google whole sale VoIP, voip termination or VoIP reseller. - Original Message - From: Benjamin Jacob [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, December 17, 2007 6:45 AM Subject: [asterisk-users] VoIP service providers/PSTN termination points Hello ppl, Am looking at some PSTN termination providers in US. If this question has been repeated, please point me to the correct link, as I've tried searching the archives but have been unsuccesful so far. I have come across quite a few companies which provide the same, such as : Iconnecthere http://www.iconnecthere.com Vonage http://www.vonage.com Teliax http://www.teliax.com I found something known as Inphonex http://www.inphonex.com. These had the cheapest rates and quite a good coverage too. Anyone with experience on this one? I am looking at a combination of decent prices and good quality. Any other suggestions or ideas welcome too. TiA - Ben. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Codecs: g729 and g723
snip Note that this is only necessary for the next 7 years, as the related patents expire sometime in 2014. /snip Yay I can't wait. Maybe we can have a pool where people chip in for G723 so Digium can afford it. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any phone capable of displaying realtimequeuestatistics?
Have a name ? Contact info ? Are those guys on the list here ? - Original Message - From: Dean Collins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 13, 2007 2:27 PM Subject: Re: [asterisk-users] Any phone capable of displaying realtimequeuestatistics? I know one of the guys in the NY Asterisk meetup demonstrated that application on a polycom display. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: Thursday, 13 December 2007 1:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Any phone capable of displaying real timequeuestatistics? Queue Metrics - Original Message - From: Peter Pauly [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, December 11, 2007 9:06 PM Subject: [asterisk-users] Any phone capable of displaying real time queuestatistics? Are there any phones whose display can show queue statistics, ie: calls waiting, etc, on the phone itself without too much trouble with Asterisk? Especially while the phone is in use (on a call)? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any phone capable of displaying real timequeuestatistics?
My bad. I did not read the message correctly. I was half asleep when I responded (like I am now) - Original Message - From: Peter Pauly [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 13, 2007 2:17 PM Subject: Re: [asterisk-users] Any phone capable of displaying real timequeuestatistics? I don't see any evidence that queue metrics can push data to the phone. I'm really looking for a home-grown solution that pushes XML/HTML to a phone during a call, like the 7960's. On 12/13/07, Dovid B [EMAIL PROTECTED] wrote: Queue Metrics - Original Message - From: Peter Pauly [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, December 11, 2007 9:06 PM Subject: [asterisk-users] Any phone capable of displaying real time queuestatistics? Are there any phones whose display can show queue statistics, ie: calls waiting, etc, on the phone itself without too much trouble with Asterisk? Especially while the phone is in use (on a call)? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] turn off auto-seek extention - force use timeout
I have an application where a call-in user is prompted to enter an identification number for schedule information. That id number is setup as an extension, and if that extension doesn't exist, it tells them that they are not scheduled, then loops back to ask for the id number again. My problem is that asterisk pre-emptively goes to the i extension (invalid) too early depending on available extensions. For example, if I put in id number 4768, and there is only 4790 and 4732, it will push to the invalid extension on the 6, then the not scheduled playback message (a cepstral command) gets cancelled out from the DTMF push of the 8. So, if I put in 4768, I get prompted to enter an id number. What I would like to do is turn off this feature, so that the number input does not get evaluated until after the timeout (preferably configurable from the extensions.conf file). Thanks in advance -Justin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk not sending 200 OK
Is the phone behind NAT ? - Original Message - From: Rob Schall To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, December 12, 2007 4:00 PM Subject: Re: [asterisk-users] Asterisk not sending 200 OK Both boxes are on the outside of nats (public IPs for both). So I don't think that would be the case. Right? Rob C F wrote: nat On 12/11/07, Rob Schall [EMAIL PROTECTED] wrote: We're trying to get a SIP peer going between our asterisk box and our provider. It should then ring our phone. The call does come in and it does execute the extension in the dial plan. But the provider says they never get a 200 OK back and therefore they send another INVITE and then after a few seconds drop the call. Here's our setup: sip.conf [ngt-trunk] type=peer qualify=yes port=5060 context=from-trunk fromuser=603XXX host=onecps.onvoip.net registersip=no username=WebSolutions secret=603XXX dtmfmode=inband insecure=very extensions.conf [from-trunk] exten = _6035467131,1,Wait(1) exten = _6035467131,2,Dial(SIP/4610) ;exten = _6035467131,3,Playback(ws-ivr) ;exten = _6035467131,4,Hangup The debug looks like: -- Executing [EMAIL PROTECTED]:1] Wait(SIP/wsol-00820870, 1) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/wsol-00820870, SIP/4610) in new stack -- Called 4610 -- SIP/4610-00838160 is ringing [Dec 11 12:18:38] NOTICE[2624]: chan_sip.c:13753 handle_request_invite: Call from '' to extension '6035467131' rejected because extension not found. -- SIP/4610-00838160 answered SIP/wsol-00820870 Really destroying SIP dialog '[EMAIL PROTECTED]' Method: NOTIFY -- Executing [EMAIL PROTECTED]:1] Wait(SIP/wsol-00845fe0, 1) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/wsol-00845fe0, SIP/4610) in new stack -- Called 4610 -- SIP/4610-0084a310 is ringing [Dec 11 12:18:49] WARNING[2624]: chan_sip.c:1939 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 624360222 (Critical Response) [Dec 11 12:18:49] WARNING[2624]: chan_sip.c:1963 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. == Spawn extension (from-trunk, 6035467131, 2) exited non-zero on 'SIP/wsol-00820870' Really destroying SIP dialog '[EMAIL PROTECTED]' Method: INVITE Really destroying SIP dialog '[EMAIL PROTECTED]' Method: INVITE -- SIP/4610-0084a310 answered SIP/wsol-00845fe0 [Dec 11 12:18:51] NOTICE[2624]: chan_sip.c:13753 handle_request_invite: Call from '' to extension '603XXX' rejected because extension not found. == Spawn extension (from-trunk, 603XXX, 2) exited non-zero on 'SIP/wsol-00845fe0' Really destroying SIP dialog '[EMAIL PROTECTED]' Method: BYE [Dec 11 12:18:58] WARNING[2624]: chan_sip.c:1939 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 624363248 (Critical Response) Really destroying SIP dialog '[EMAIL PROTECTED]' Method: INVITE Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS [Dec 11 12:19:01] WARNING[2624]: chan_sip.c:1939 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 624366883 (Critical Response) Really destroying SIP dialog '[EMAIL PROTECTED]' Method: INVITE Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID in Cape Town South Africa
The biz list would be more of help for this. - Original Message - From: Eric Smith [EMAIL PROTECTED] To: Asterisk Users Mailing List asterisk-users@lists.digium.com Sent: Thursday, December 13, 2007 6:03 PM Subject: [asterisk-users] DID in Cape Town South Africa Are there any service providers offering Cape Town DID's? -- - Eric Smith ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to make multiple extensions simultaneous calls on a single telephone line
But he's talking about a single POTS line into a single FXO. Only one conversation can occur across this link at a time. The PSTN provider may send you special ring indications over that line to indicate which number was *dialed* to make it ring, but only one conversation at a time. Generally, when one of your FXS phones is calling out on the POTS line, the other FXS phone will fail in its Dial statement and move along in the dialplan, probably indicating Congestion if you've configured nothing else. Mojo robert boardman wrote: hi vincent, In the UK you can have multiple pots lines with the same telephone number. but you would need more fxo lines for this. Regards Robb Vincent Li wrote: Hi Lists, I have one box with two FXO and two FXS ports, it is running asterisk inside. I have one sinle POTS line connected to the one FXO and two phone sets connected to the FXS port. Extension 6003 is asigned to one fxs and 6004 is asigned to another fxs, the two extensions can call each other, they can both make/receive PSTN call, but they can't make PSTN call simultaneously. Is it achievble in Asterisk to let them make PSTN call simulataneously through one sinle POTS line? I don't know anything about traditional PBX system, it seems one shop can have one single phone number and mutiple extensions, then the extensionss can make/receive PSTN call simultaneously, is this the same senerio as the one single POTS line to FXO and multiple extensions on FXSs? Thanks for help. Vincent ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.15, Solaris and record command
I have installed Asterisk 1.4.15 on Solaris and got it all running seemingly fine. However, when I record a message or voicemail, it will not recognize the '#' key to stop recording. Hanging up is the only way to end the recording. DTMF seems to work fine elsewhere. Is there a common problem or issue that I am missing? I've tried Google, but have had no success. Thanks, Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] turn off auto-seek extention - force use timeout
So I'm guessing this is what you're doing:
--
[ids]
exten = s,1,playback(enter your id number)
exten = s,2,WaitExten(10)
exten = s,3,Goto(1)
exten = 4768,1,blahblahblah
exten = 4790,1,blahblahblah
exten = 4732,1,blahblahblah
exten = i,1,playback(error)
exten = i,2,goto(s,1)
--
So, maybe place the phones in a context that waits for a four-digit id
_before_ matching it to the context you were initially trying:
--
[getid]
exten = s,1,playback(enter your id number)
exten = s,2,WaitExten(10)
exten = s,3,Goto(1)
exten = _4XXX,1,goto(ids,${exten},1)
[ids]
exten = 4768,1,blahblahblah
exten = 4790,1,blahblahblah
exten = 4732,1,blahblahblah
exten = i,1,playback(error)
exten = i,2,goto(getid,s,1)
--
Untested: I wonder if one entered an extension that didn't exist, say
4555, when we tried to Goto(ids, 4555, 1) would we get directed to
extension i in the extensions context or would the call be dropped
completely?
Moj
Justin Killen wrote:
I have an application where a call-in user is prompted to enter an
identification number for schedule information. That id number is
setup as an extension, and if that extension doesn’t exist, it tells
them that they are not scheduled, then loops back to ask for the id
number again. My problem is that asterisk pre-emptively goes to the i
extension (invalid) too early depending on available extensions. For
example, if I put in id number 4768, and there is only 4790 and 4732,
it will push to the invalid extension on the 6, then the “not
scheduled” playback message (a cepstral command) gets cancelled out
from the DTMF push of the 8. So, if I put in 4768, I get prompted to
enter an id number. What I would like to do is turn off this feature,
so that the number input does not get evaluated until after the
timeout (preferably configurable from the extensions.conf file).
Thanks in advance
-Justin
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Re: [asterisk-users] turn off auto-seek extention - force use timeout
Creative use of the 'read' application? PaulH On Wed, 2007-12-19 at 17:12 -0800, Justin Killen wrote: I have an application where a call-in user is prompted to enter an identification number for schedule information. That id number is setup as an extension, and if that extension doesn’t exist, it tells them that they are not scheduled, then loops back to ask for the id number again. My problem is that asterisk pre-emptively goes to the i extension (invalid) too early depending on available extensions. For example, if I put in id number 4768, and there is only 4790 and 4732, it will push to the invalid extension on the 6, then the “not scheduled” playback message (a cepstral command) gets cancelled out from the DTMF push of the 8. So, if I put in 4768, I get prompted to enter an id number. What I would like to do is turn off this feature, so that the number input does not get evaluated until after the timeout (preferably configurable from the extensions.conf file). Thanks in advance -Justin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.15, Solaris and record command
Mike Clark wrote: I have installed Asterisk 1.4.15 on Solaris and got it all running seemingly fine. However, when I record a message or voicemail, it will not recognize the '#' key to stop recording. Hanging up is the only way to end the recording. DTMF seems to work fine elsewhere. Is there a common problem or issue that I am missing? I've tried Google, but have had no success. Thanks, Mike I did some testing and this appears to be some problem with my sip provider. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change sendmail return path
On Wednesday 19 December 2007 17:44:15 shadowym wrote: I had high hopes for this solution for unfortunately it's not working. Did exactly as you specified but return path is still [EMAIL PROTECTED] even though [EMAIL PROTECTED] in voicemail.conf :( Did you restart Sendmail? It doesn't pick up changes to its config file otherwise. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
On Wednesday 19 December 2007 07:22:18 Patrick wrote: On Tue, 2007-12-18 at 19:43 -0800, Steve Edwards wrote: The old syntax was inconsistent -- show manager command vs sip show channels and just plain bad -- for example sip reload should have been reload sip. I agree. Reload sip would be the logical thing. [snip] Approach the command line like a noob. I want Asterisk to show me something so I'll start the command line with show. I'm not quite sure what I'm doing, so I'll press TAB to see what I can show. Oh, channel looks like what I want. Hmm, too much. Maybe I should have qualified what kind of channel I'm looking for BEFORE the word channel. That makes sense to me. It's also what I'm used to from working with other equipment. Here's a suggestion -- stop thinking like a parser and start thinking like a person :) Which makes more sense (at least in English)? 1) show black dogs -- show sip channels 2) black show dogs -- sip show channels 3) dogs black show -- channels sip show 4) show dogs black -- show channels sip 5) black dogs show -- sip channels show 6) dogs show black -- channels show sip Excellent example. I'll put my 0.2 cents on #1 :) Is it too late to fix this for 1.6? I sincerely hope not. Your example shows that the CLI could use some TLC. Let's hope the powers that be agree. http://bugs.digium.com/view.php?id=11605 For everything that matches category verb arguments, this translation will work fine. For things which don't, well, they needed to be fixed anyway. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ip phone suggestion for Asia?
Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones. searching in the mail list, we found aastra is good, but they don't sale to asia. grandstream looks good also.there are many grandstream users in the list, can someone share any good or bad experience about grandstream today? if there are other good choice, please tell us!! thanks a lot for your help!! Regards, tbskyd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
dovid... while this seems like a good idea to have both sip show channels and show channels sip having two, three or even four ways to do the same thing would confuse/cripple the learning curve... * would turn into a microsoft mentality where there are dozens of ways to configure/reconfigure some of their products... word, for example, can be configured with or without the tool bars and then you can configure hot-keys... in fact, you can configure some products so that someone who learns it with a hacked config, could not possibly use the original stock config... sorry to go on about this but it is one of my hot buttons... daveC Dovid B wrote: - Original Message - From: Steve Edwards [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 19, 2007 5:43 AM Subject: Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old! On Sat, 15 Dec 2007, Johansson Olle E wrote: I wonder if there are any major obstacles for upgrading. How about the change from a bad command line interface to a really bad command line interface? I mean, Seriously? (in a Grey's Anatomy kind of way...) The old syntax was inconsistent -- show manager command vs sip show channels and just plain bad -- for example sip reload should have been reload sip. The new syntax continues down the noun-verb path instead of correcting itself and using verb-noun like most other applications (MySQL, GDB, Oracle, etc.) Then, just to make it worse, now I have to learn which commands somebody (arbitrarily) decided are core and which are not -- for what benefit? Certainly doesn't make MY job easier! Approach the command line like a noob. I want Asterisk to show me something so I'll start the command line with show. I'm not quite sure what I'm doing, so I'll press TAB to see what I can show. Oh, channel looks like what I want. Hmm, too much. Maybe I should have qualified what kind of channel I'm looking for BEFORE the word channel. Here's a suggestion -- stop thinking like a parser and start thinking like a person :) Which makes more sense (at least in English)? 1) show black dogs -- show sip channels 2) black show dogs -- sip show channels 3) dogs black show -- channels sip show 4) show dogs black -- show channels sip 5) black dogs show -- sip channels show 6) dogs show black -- channels show sip Is it too late to fix this for 1.6? Thanks in advance, I think as many people have pointed out they are used to a lot of commands out there so changing it yet again would make more people unhappy. But maybe asterisk can have both. Why not sip show channels for the old timers and show channels sip or show sip channels for the n00b's. Why shouldn't asterisk have both options ? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] turn off auto-seek extention - force use timeout
mojo,
nice suggestion.
daveC
Mojo with Horan Company, LLC wrote:
So I'm guessing this is what you're doing:
--
[ids]
exten = s,1,playback(enter your id number)
exten = s,2,WaitExten(10)
exten = s,3,Goto(1)
exten = 4768,1,blahblahblah
exten = 4790,1,blahblahblah
exten = 4732,1,blahblahblah
exten = i,1,playback(error)
exten = i,2,goto(s,1)
--
So, maybe place the phones in a context that waits for a four-digit id
_before_ matching it to the context you were initially trying:
--
[getid]
exten = s,1,playback(enter your id number)
exten = s,2,WaitExten(10)
exten = s,3,Goto(1)
exten = _4XXX,1,goto(ids,${exten},1)
[ids]
exten = 4768,1,blahblahblah
exten = 4790,1,blahblahblah
exten = 4732,1,blahblahblah
exten = i,1,playback(error)
exten = i,2,goto(getid,s,1)
--
Untested: I wonder if one entered an extension that didn't exist, say
4555, when we tried to Goto(ids, 4555, 1) would we get directed to
extension i in the extensions context or would the call be dropped
completely?
Moj
Justin Killen wrote:
I have an application where a call-in user is prompted to enter an
identification number for schedule information. That id number is
setup as an extension, and if that extension doesn’t exist, it tells
them that they are not scheduled, then loops back to ask for the id
number again. My problem is that asterisk pre-emptively goes to the i
extension (invalid) too early depending on available extensions. For
example, if I put in id number 4768, and there is only 4790 and 4732,
it will push to the invalid extension on the 6, then the “not
scheduled” playback message (a cepstral command) gets cancelled out
from the DTMF push of the 8. So, if I put in 4768, I get prompted to
enter an id number. What I would like to do is turn off this feature,
so that the number input does not get evaluated until after the
timeout (preferably configurable from the extensions.conf file).
Thanks in advance
-Justin
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--
My wife's sister is in California.
I should buy her a Videophone2008!
Truly, The Next Best Thing to Being There!
--
WorldWideVideoPhones.com
856.380.0894
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Re: [asterisk-users] ip phone suggestion for Asia?
Hi tbskyd, We have found that the Grandstream's are not that great a phone. One of our best sellers is the Snom range and I know that the Australian supplier spends half his time in Hong Kong so you shouldn't have any problems getting so over there. They are a little more expensive than the Grandstream's but cheaper than the Polycoms around that Aastra price range. Cheers, Joel. On Thu, 2007-12-20 at 12:33 +0800, d tbsky wrote: Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones. searching in the mail list, we found aastra is good, but they don't sale to asia. grandstream looks good also.there are many grandstream users in the list, can someone share any good or bad experience about grandstream today? if there are other good choice, please tell us!! thanks a lot for your help!! Regards, tbskyd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk.NET API --help required
Hello all, Here is the requirement from my side to use Asterisk.NET API to generate an automated call (outgoing) from asterisk and then link to one of the extensions which plays a sound file for the callee. For this i have worked out in the follwing way 1)modified manager.conf to facilitate this API to talk to asterisk 2)used the command Originate to call a Registered user under asterisk and when the user answers the phone it plays whatever i put against the extension.. But my exact requirement is like this 1)Call to the user 2)if answers connect him to the extension provided in the extensions.conf 3)if the user didnt lift the phone within the deault timeout period(30 sec) 4)if the user cancels the phone (Congestion case) 5)if the user not registerd to the(unreachable case) to trace the cases of 3, 4, 5 how should i follow the API I got confused with originate action,orginate sucess event , originate failure event can anybody give me a hint so that i can proceed further thanks in advance for the kind suggestions. regards srinivas antarvedi -- Srinivas Antarvedi ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
hi Joel: thanks a lot for your reply. i forgot snom :) i wrote a snom employ found in this email list, but got no reply. i saw there are huge complain about grandstream firmware this year. grandstream seems response and solve some of them. i wonder if their product is ok now. i don't know the situation of snom, will they response to user's request? thanks again for your kindly help!! Regards, tbskyd 2007/12/20, Joel Hill [EMAIL PROTECTED]: Hi tbskyd, We have found that the Grandstream's are not that great a phone. One of our best sellers is the Snom range and I know that the Australian supplier spends half his time in Hong Kong so you shouldn't have any problems getting so over there. They are a little more expensive than the Grandstream's but cheaper than the Polycoms around that Aastra price range. Cheers, Joel. On Thu, 2007-12-20 at 12:33 +0800, d tbsky wrote: Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones. searching in the mail list, we found aastra is good, but they don't sale to asia. grandstream looks good also.there are many grandstream users in the list, can someone share any good or bad experience about grandstream today? if there are other good choice, please tell us!! thanks a lot for your help!! Regards, tbskyd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
You get what you pay for. Snoms are good phones. Grandstreams are also good. I hear Snoms are easier to get around NAT and the seem like higher quality construction. Grandstreams are great for cheap and easy set-ups. I remember one guy telling me he buy up a case and if anything goes wrong with a unit he has a couple spares to go in its place. Over all... if you're looking for setting up office phones, Snom, Polycom, and Aastra look/feel nice. If you're looking to set up small offices or call centers on the cheap, Grandstreams are OK. I personally like Grandstream for home use, but I use Polycom for work, and I hear good things about Snom. On Dec 20, 2007 12:06 AM, d tbsky [EMAIL PROTECTED] wrote: hi Joel: thanks a lot for your reply. i forgot snom :) i wrote a snom employ found in this email list, but got no reply. i saw there are huge complain about grandstream firmware this year. grandstream seems response and solve some of them. i wonder if their product is ok now. i don't know the situation of snom, will they response to user's request? thanks again for your kindly help!! Regards, tbskyd 2007/12/20, Joel Hill [EMAIL PROTECTED]: Hi tbskyd, We have found that the Grandstream's are not that great a phone. One of our best sellers is the Snom range and I know that the Australian supplier spends half his time in Hong Kong so you shouldn't have any problems getting so over there. They are a little more expensive than the Grandstream's but cheaper than the Polycoms around that Aastra price range. Cheers, Joel. On Thu, 2007-12-20 at 12:33 +0800, d tbsky wrote: Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones. searching in the mail list, we found aastra is good, but they don't sale to asia. grandstream looks good also.there are many grandstream users in the list, can someone share any good or bad experience about grandstream today? if there are other good choice, please tell us!! thanks a lot for your help!! Regards, tbskyd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
hi Joe: thanks for your great opinion. we will buy spares of course. our new office needs about 150 phone.we use alcatel before. but want to change to asterisk now for better future. since aastra is not sale to asia (that's sad, aastra boss seems born at taiwan). i will choose snom or grandstream. polycom seems really expensive, i saw someone say their cheap model has no hardware transfer key. so i think i need to pay much more to get what i want in polycom. Regards, tbskyd 2007/12/20, Joe [EMAIL PROTECTED]: You get what you pay for. Snoms are good phones. Grandstreams are also good. I hear Snoms are easier to get around NAT and the seem like higher quality construction. Grandstreams are great for cheap and easy set-ups. I remember one guy telling me he buy up a case and if anything goes wrong with a unit he has a couple spares to go in its place. Over all... if you're looking for setting up office phones, Snom, Polycom, and Aastra look/feel nice. If you're looking to set up small offices or call centers on the cheap, Grandstreams are OK. I personally like Grandstream for home use, but I use Polycom for work, and I hear good things about Snom. On Dec 20, 2007 12:06 AM, d tbsky [EMAIL PROTECTED] wrote: hi Joel: thanks a lot for your reply. i forgot snom :) i wrote a snom employ found in this email list, but got no reply. i saw there are huge complain about grandstream firmware this year. grandstream seems response and solve some of them. i wonder if their product is ok now. i don't know the situation of snom, will they response to user's request? thanks again for your kindly help!! Regards, tbskyd 2007/12/20, Joel Hill [EMAIL PROTECTED]: Hi tbskyd, We have found that the Grandstream's are not that great a phone. One of our best sellers is the Snom range and I know that the Australian supplier spends half his time in Hong Kong so you shouldn't have any problems getting so over there. They are a little more expensive than the Grandstream's but cheaper than the Polycoms around that Aastra price range. Cheers, Joel. On Thu, 2007-12-20 at 12:33 +0800, d tbsky wrote: Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones. searching in the mail list, we found aastra is good, but they don't sale to asia. grandstream looks good also.there are many grandstream users in the list, can someone share any good or bad experience about grandstream today? if there are other good choice, please tell us!! thanks a lot for your help!! Regards, tbskyd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
