Hi All,
I can not install the asterisk-addons as it thinks there is no
mysqlclient installed. I have installed mysql, mysql-server and
mysql-devel and I am still unable to install the addons. I am running
CentOS 5.2 i386.
Please somebody help.
Hi,
fax gateway isn't just a packet bridging.
It does the mediation between T30 (voice) - T38 (fax over ip) protocols.
It does work for asterisk 1.4, asterisk 1.6, asterisk svn head.
If it doesn't please send me a bug report and I'm going to fix it.
Best regards
Daniel.
On Mon, Oct 6, 2008 at
Make sure they are not using double NAT. Many ISPs these days send
their subscribers a modem that in reality is a router.
Also if you can post the PAP2 configuration. I hope you are using
provisioning.. too bad Linksys makes it possible to obtain that
information.
On Mon, Oct 6, 2008 at 12:40
Klaverstyn, David C schrieb:
Hi All,
I can not install the asterisk-addons as it thinks there is no
mysqlclient installed. I have installed mysql, mysql-server and
mysql-devel and I am still unable to install the addons. I am running
CentOS 5.2 i386.
Please somebody help.
Yes, unfortunately, VOIP wiki did not mention about installing
mysql-client which it should have been.
Without mysql-client, you cannot change passwords, grants, etc.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stefan Schmidt
Sent:
Hi,
I agree with Gordon.
We are still using Asterisk 1.2 because we are waiting for Asterisk 1.4
features to work as for Asterisk 1.2 (it seems to us that parking and
queues have some problems... so not good enough for production).
Giorgio Incantalupo
Gordon Henderson wrote:
On Mon, 6 Oct
Hi!
I have a different approach. I wrote a small application which simply starts
an audio player. You can write a very small script to answer fast or just use
telnet like this:
telnet localhost 8642
At the moment everything is hardcoded, but can be changed in any case. I use
15s
2008/10/6 Steve Edwards [EMAIL PROTECTED]
On Mon, 6 Oct 2008, Alex Balashov wrote:
Giedrius Augys wrote:
What tools and programming (scripting) language do you use for FastAGI?
Whatever languages FastAGI APIs are available for. You are pretty much
limited to languages whose
Hi,
Currently I'm using Asterisk 1.2 and 1.4 in different setups. When a user wants
to pick up a call within his/her pickup group, *8 must be dialed (or whatever
you define in features.conf).
However, these users were used to another behavior when they had a commercial
PBX (Bosch). When a
Jim Boykin wrote:
I know about those packages. Questions is how do we use those packages
to build our own RPM. We use asterisk SVN trunk.
asterisk usually comes with asterisk.spec and make target rpm. With
some slight modifications on the spec file you can pretty much build
whatever you need
Atis Lezdins wrote:
On Tue, Oct 7, 2008 at 8:45 AM, Pavel Jezek [EMAIL PROTECTED] wrote:
Steve Murphy wrote:
On Mon, 2008-10-06 at 18:25 +0200, Pavel Jezek wrote:
Atis Lezdins wrote:
On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
Hi Ken,
we are quite satisfied with Linksys SRW248G4P. 48 port PoE, 4 GB uplinks
and 2 GBIC slots. VLAN, QoS and all the like is on board. Around US$600
I guess.
Only drawback in my opinion is that they are loud like a starting
airplane. You definately don't want them next to your desk. ;-)
Brendan Martens wrote:
Having thought some more about my issue I think I can perhaps ask my
question more succinctly: is it possible to get dynamic (or
realtime) data from ldap within the various .conf files?
If there is not a convenient function for getting this in the .conf
files,
Brendan Martens [EMAIL PROTECTED] writes:
On Oct 6, 2008, at 3:52 PM, Gordon Henderson wrote:
The answer you are looking for is that you should be using a
supported,
stable version, and right now, 1.4 is the only one that fits. If I
were
starting today, I'd go with 1.4.
1.6.0 has
Here, I've written a perl script that rewrites the actual sip.conf
itself (as well as generates a custom myexten.conf file, which is
included in the main extensions.conf file.)
I was hoping to keep it all native to asterisk, but I would be willing
to give that a try. Where can I get this
Hi, in tftp server I have the followings files:
apps11.1-1-3-15.sbn
cnu11.3-1-3-15.sbn
copstart.sh
cvm11sip.8-0-3-16.sbn
dsp11.1-1-3-15.sbn
jar11sip.8-0-3-16.sbn
load307
load369
SIP11.8-0-4SR1S.loads
term06.default.loads
term11.default.loads
..and on 7906g in status menu I have:
load file:
On Tue, Oct 7, 2008 at 2:20 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
On Tue, Oct 7, 2008 at 8:45 AM, Pavel Jezek [EMAIL PROTECTED] wrote:
Steve Murphy wrote:
On Mon, 2008-10-06 at 18:25 +0200, Pavel Jezek wrote:
Atis Lezdins wrote:
On Mon, Oct 6, 2008 at 5:21 PM,
Jim Boykin wrote:
I know about those packages. Questions is how do we use those
packages
to build our own RPM. We use asterisk SVN trunk.
asterisk usually comes with asterisk.spec and make target rpm. With
some slight modifications on the spec file you can pretty much build
whatever
Hi all,
i wrote a script agi, sking for a code, after that it sends an
email now i need to send a fax... any hints or tips for that?
Ty in advance.
--
Riccardo Cupardo
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Are you dialing a 1 before every number? That is required unless you make
another pattern match.
exten = _NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
Then it becomes 10-digit dialing without the need to dial a 1. If that
doesn't work open up the asterisk console and attempt to make a call and
Are includes supported in the file /etc/dahdi/system.conf
link you can include in say a sip.conf
What about in chan_dahdi.conf?
Thanks,
Jerry
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25
On Tue, Oct 07, 2008 at 10:08:25AM -0400, Jerry Geis wrote:
Are includes supported in the file /etc/dahdi/system.conf
no. Note that '#' begins a comment in system.conf .
link you can include in say a sip.conf
What about in chan_dahdi.conf?
Yes, just like any Asterisk configuration file.
SIP [EMAIL PROTECTED] writes:
The truth is there are plenty of email clients that CAN decode
Hotmail messages, and if you choose one that can't, you can't blame
anyone but yourself.
The truth is that there are no Netcom^WAOL^WHotmail users who write
anything worth reading. I had just
I have just confirmed that they may be having a problem with double NAT.
They have two ATAs, and they have two different DSL connections. One set-up
goes from the first DSL Modem (NAT Wirless are disabled on the DSL Modems)
to a Linksys WRT110 and then there is a WRT54G hooked in to the 110 that
On Tue, 2008-10-07 at 12:03 +0800, Nhadie wrote:
just wondering what's happening here:
i have a pap2 and an spa941. everytime i call my spa from my pap2 i can
see it being added dynamically on the regcontext:
[Oct 7 11:59:08] -- Saved useragent Linksys/SPA942-5.2.8 for peer
100100
On Sun, Oct 5, 2008 at 3:30 PM, Bill Michaelson [EMAIL PROTECTED] wrote:
The IP330 has a subminiature jack for headset/mic combos. Are there quality
headsets anyone would recommend for in-office use for heavy users with these
phones? Using any wiring path? I've tried a cell phone
I recently purchased a few SRW208P switches. They work fine. If you
run Windows. Granted a lot of people run windows instead of Mac or
Linux, but be aware (to those looking) that the SRW line of switches
REQUIRE Internet Explorer on Windows. The support site says it is
recommended, but
The voicemail command should be Voicemail([EMAIL PROTECTED]) so in
extensions.conf
exten = 101,n,Voicemail([EMAIL PROTECTED])
As for the console when you launch it add v's to set the debugging level
'asterisk -vr' you can also run 'core set debug X' X=debug level 0-10 I
believe. Just to
Brendan Martens wrote:
Jim Boykin wrote:
I know about those packages. Questions is how do we use those
packages
to build our own RPM. We use asterisk SVN trunk.
asterisk usually comes with asterisk.spec and make target rpm. With
some slight modifications on the spec file you can pretty
On Tue, Oct 7, 2008 at 8:45 AM, Pavel Jezek [EMAIL PROTECTED] wrote:
Steve Murphy wrote:
On Mon, 2008-10-06 at 18:25 +0200, Pavel Jezek wrote:
Atis Lezdins wrote:
On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
Hi, according to discussion on asterisk IRC, where
Julien,
I would love to see this solution so please upload the code.
Thank you very much.
robert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Julien Claassen
Sent: Tuesday, October 07, 2008 4:06 AM
To: Asterisk Users Mailing List -
I would like to make asterisk call center , I have a ET410P card
what I need to install like packages ?
where i can get a best documents to doing it ?
thank you for advance
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Here Is the situation. Both users can plug in their ATAs and I can watch
the server output, they register and then they can make calls and I can call
them. Some time later (usually within minutes) the ATAs show to be
unreachable and I can no longer call; however, they can still make calls.
Well, after very quickly making a test call it's not Vitelity. It could be
something with your account? Might want to try opening a support ticket. If
you want, create a sub account and e-mail me off list the username and
password and I'll test it with my box or vice versa.
On Tue, Oct 7, 2008 at
Hi All,
I need some help to about override callerid,
if i get blocked callerid and also having privacy=full.
i am trying to override callerid on that call, but the callerid is not
changed
The sip trace is given below
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
On Oct 7, 2008, at 4:19 AM, Chris Bagnall wrote:
I recently purchased a few SRW208P switches. They work fine. If you
run Windows. Granted a lot of people run windows instead of Mac or
Linux, but be aware (to those looking) that the SRW line of switches
REQUIRE Internet Explorer on Windows.
Am Dienstag, den 07.10.2008, 01:42 -0700 schrieb Vieri:
Hi,
Currently I'm using Asterisk 1.2 and 1.4 in different setups. When a user
wants to pick up a call
within his/her pickup group, *8 must be dialed (or whatever you define in
features.conf).
[...]
I was thinking of configuring some
Vieri wrote:
Hi,
Currently I'm using Asterisk 1.2 and 1.4 in different setups. When a user
wants to pick up a call within his/her pickup group, *8 must be dialed (or
whatever you define in features.conf).
However, these users were used to another behavior when they had a commercial
Are you sure you have set the 7960 to SIP?
By default they use SCCP, so you need to go through the process of
changing them over, which ideally would just be done with the edits you
have already in the load files but generally means going back to an
early version of the SIP code then working
Hi, I have a problem with Cisco 7906G and SIP protocol use with Asterisk
1.2.26.
I have uploaded in my tftp server the firmware
'cmterm-7911_7906-sip.8-0-4SR1' that use 'SIP11.8-0-4SR1S.loads' and in
SEPmacaddress.cnf.xml I have:
loadInformationSIP11.8-0-4SR1S/loadInformation
..but in tftp
--- On Tue, 10/7/08, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
regarding your combination of analog phones and ATAs I
would look for
the auto-dial functionality in the ATA. I am pretty sure I
saw it in one
web-interface or the other
Thanks!
I actually found the option. I'm using
Very new to Asterisk, on my console it says there are 47 bad
destinations...What is the best way to track these down and resolve
them
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25
Darren Severino wrote:
Well, after very quickly making a test call it's not Vitelity. It could
be something with your account? Might want to try opening a support
ticket. If you want, create a sub account and e-mail me off list the
username and password and I'll test it with my box or vice
Interesting, I've been using them since April and haven't had a problem. I
know they changed their server settings a while back but didn't notice
anything recently.
On Tue, Oct 7, 2008 at 11:47 AM, Roderick A. Anderson [EMAIL PROTECTED]wrote:
Darren Severino wrote:
Well, after very quickly
Hi!
I just uploaded a small tarball of my ast_picker application with a few
extras and an example_dialplan. You can find it here;
http://juliencoder.de/ast_picker-0.1.tar.bz2
As I said before: It's still early stage and not too customiseable, but you
can manage. If you need help, just tell
I don't mean to be a pain, but i could really do with a heads up on this...
does anyone have ANY ideas? I've trawled through google and come up with
nothing except for questions with no answers...
Cheers
Geraint
2008/10/6 Geraint Lee [EMAIL PROTECTED]
Hi all, I've done this a few times with
I posted this previously but didn't get a response. I have been working
through the tutorial in the Van Meggelen book and can't get a registered SIP
phone. I'm using 1.6, on Fedora 9 and a SPA941. I also tried an Engenius
Wi-Fi SIP phone. Both can be pinged from the Linux Computer but
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wilton Helm
Sent: Tuesday, October 07, 2008 12:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Setting up Asterisk
I posted this previously but didn't get a response. I have
run asterisk in verbosed mode
#asterisk -cg
and try to register your WiFi phone or Xlite softephone ... and watch log on
linux terminal
One more option this command will extract packect contain 5060 port...
#tcpdum -i eth0 port 5060
#if any issue in WiFi phone then try to
On Monday 06 October 2008 14:58:09 Karl Fife wrote:
In several places online, and in the Asterisk F.O.T. book, there is a
warning against using '_.' saying:
[it] should probably never be used.
However, the need often arises act on numeric extensions that begin with
*'s and #'s, and '+', and
I have a handful of cisco phones that has been working.
Today they started showing X's. looking at sip debug I see the 401
unauthorized.
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP IP:52110;branch=z9hG4bK29694d4a;received=IP
From: sip:[EMAIL PROTECTED];user=phone
To: sip:[EMAIL
Jerry Geis wrote:
I have a handful of cisco phones that has been working.
Today they started showing X's. looking at sip debug I see the 401
unauthorized.
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP IP:52110;branch=z9hG4bK29694d4a;received=IP
From: sip:[EMAIL PROTECTED];user=phone
To:
Doug,
I have your example working but how do I get this to work with a ring group?
One more problem I have is poor quality of sound when the call file is
played.
I do not have this problem when moh is played or when console/dsp is used
for live voice?
What could be the problem?
Do you know where I
Did the server reboot or lose communication? This happens with our 7970's
sometimes if there's been a hiccup, usually dialing voicemail registers them
back up - occasionally we've had to do the soft reboot from the screen.
401 unauth - looks like it may be md5secret issue, or nat traversal over
Hello Robert!
I don'texactly know, what you need for a ringing file. but if it is the
matter of just some announcement sound, I could make you one. It's easy.
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT
Since when is there a T.38 Gateway in Asterisk 1.4?
On Tue, Oct 7, 2008 at 3:01 AM, Daniel Ferenci
[EMAIL PROTECTED] wrote:
Hi,
fax gateway isn't just a packet bridging.
It does the mediation between T30 (voice) - T38 (fax over ip) protocols.
It does work for asterisk 1.4, asterisk 1.6,
Julien,
Thank you, I need a file which when played sounds like a phone ringing ...
:)
robert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Julien Claassen
Sent: Tuesday, October 07, 2008 3:51 PM
To: Asterisk Users Mailing List - Non-Commercial
Hi,
http://bugs.digium.com/view.php?id=13405 was posted on 30/08/2008.
I'm looking forward to seeing your feedback or bug report.
Thank you in advance.
Best regards
Daniel.
On Tue, Oct 7, 2008 at 10:01 PM, Andrew Joakimsen [EMAIL PROTECTED]wrote:
Since when is there a T.38 Gateway in
No problem... I'll whomp something up. I'll upload a tarball tomorrow or
thrusday morning at the latest.
Quality: desired samplingrate, bit-depth, channel number? Any particular
needs, or will CD quality just be fine for you?
Kindest regards
Julien
P.S.: Did you get to my
Hi,
I have a simple desire to be able to screen people before being onnected to
them. I`ve seen plenty of examples on the web and I`ve figured it out.
There is only one case in where it doesnt act as I want it to: if I hang up
the phone, I don`t want the caller to be disconnected but (for the
Mike wrote:
So, I guess my question is: how do I set a variable that ISN`T lost
when the call initiated using the Dial g option is hung up ?
You can use the internal database for that:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+db
Doug
--
Ben Franklin quote:
Those
Anything what can be played through the console/dsp will work for me.
Yes, I received your application and hope to play with it tonight or
tomorrow.
Thank you very much.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Julien Claassen
Sent:
Doug,
Thanks for the quick answer. How does that help me though, since this is a
per channel variable and not a global variable?
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Tuesday, October 07, 2008 16:54
To: Asterisk Users
Did the server reboot or lose communication? This happens with our 7970's
sometimes if there's been a hiccup, usually dialing voicemail registers them
back up - occasionally we've had to do the soft reboot from the screen.
401 unauth - looks like it may be md5secret issue, or nat traversal
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Tuesday, October 07, 2008 5:42 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] cisco phones getting SIP 401 unauthorized
Matt,
The phones are inside the LAN.
what is
THANK YOU!!!
This appears to have worked. I am assuming we can do the same thing on our
SPA-962s that we send to make sure they work with no problems.
Thank you to everyone here for your help. This is an excellent group to
have access to for questions. I hope to learn and be able to help
Mysql for CentOS 5.2 is the mysql client tools.
mysql.i386 : MySQL client programs and shared libraries.
Does anyone have any other suggestions?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee, John
(Sydney)
Sent: Tuesday, 7 October 2008 5:18 PM
To:
Mike wrote:
Doug,
Thanks for the quick answer. How does that help me though, since this is a
per channel variable and not a global variable?
Make sure your key in the database is specific to only that call. Time,
date, caller-id number or even a combination of all.
Can't you save your
Hi,
Got this weird problem that the caller does not hear a ring.
The issue is it's specific to the local telco:
Using telco 1 (mobile), calls in to my DID, caller hears a ring and gets
forwarded to voicemail if i did not answer.
Using telco 1 (landline), calls in to my DID, caller hears a
I'm using Teliax, and every incoming call has:
Executing [EMAIL PROTECTED]:2] Answer(IAX2/usrname-14376, ) in
new stack
-- Executing [EMAIL PROTECTED]:3] Dial(IAX2/usrname-14376,
DAHDI/1,60) in new stack
-- Called 1
-- DAHDI/1-1 is ringing
-- IAX2/usrname-14376 requested
I have a question about changing passwords.
When I change the secret field in sip.conf for a Grandstream phone, and
then use the browser to change the Authenticate ID field of the phone to
match what's in the sip.conf file, I can no longer make calls on the phone.
Any ideas?
Thanks for
Yes, try
perl -MCPAN -e install DBD::mysql
Then do a make clean, ./bootstrap, ./configure, make menuselect
Worked for me, not sure all the above is required but configure.
Thanks,
Steve Totaro
On Tue, Oct 7, 2008 at 7:19 PM, Klaverstyn, David C
[EMAIL PROTECTED] wrote:
Mysql for CentOS 5.2
Klaverstyn, David C schrieb:
Mysql for CentOS 5.2 is the mysql client tools.
mysql.i386 : MySQL client programs and shared libraries.
Does anyone have any other suggestions?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stefan
Philipp Kempgen schrieb:
Klaverstyn, David C schrieb:
Mysql for CentOS 5.2 is the mysql client tools.
mysql.i386 : MySQL client programs and shared libraries.
Does anyone have any other suggestions?
http://www.centos.org/modules/newbb/viewtopic.php?topic_id=13097
Or just download Debian
Ken Zarifes schrieb:
When I change the secret field in sip.conf for a Grandstream phone, and
then use the browser to change the Authenticate ID field of the phone to
match what's in the sip.conf file, I can no longer make calls on the phone.
Any ideas?
Go to the Asterisk CLI, core set
On Tue, Oct 7, 2008 at 8:30 PM, Ken Zarifes [EMAIL PROTECTED] wrote:
I have a question about changing passwords.
When I change the secret field in sip.conf for a Grandstream phone, and
then use the browser to change the Authenticate ID field of the phone to
match what's in the sip.conf
On Tue, Oct 7, 2008 at 8:04 PM, Nhadie [EMAIL PROTECTED] wrote:
Hi,
Got this weird problem that the caller does not hear a ring.
The issue is it's specific to the local telco:
Using telco 1 (mobile), calls in to my DID, caller hears a ring and gets
forwarded to voicemail if i did not
On Tue, Oct 7, 2008 at 4:36 PM, Mike [EMAIL PROTECTED] wrote:
Hi,
I have a simple desire to be able to screen people before being onnected to
them. I`ve seen plenty of examples on the web and I`ve figured it out.
There is only one case in where it doesn't act as I want it to: if I hang up
I recently did something similar using fax1.com. If you can send an
email you can send a fax that way.
On Tue, Oct 7, 2008 at 9:19 AM, Riccardo Cupardo [EMAIL PROTECTED] wrote:
Hi all,
i wrote a script agi, sking for a code, after that it sends an email now
i need to send a fax... any
I got this:
[Oct 7 18:26:17] NOTICE[6309] chan_sip.c: Registration from
'sip:[EMAIL PROTECTED];user=phone' failed for '192.168.163.134' -
Username/auth name mismatch
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
Kempgen
Sent: Tuesday, October
It has been a long time since I touched a GS phone but I think the field on
the browser is password, not Authenticate ID. Don't forget to reload
sip.conf or asterisk between changes.
You were absolutely right.
Thanks!
Ken
___
-- Bandwidth
The value is not Authenticate ID; From the config file:
# Authenticate ID
P36 = 8000
# Authenticate password
P34 =
If you look at the HTML source of the webconfig the form field you
need to edit will be marked P34.
On Tue, Oct 7, 2008 at 5:30 PM, Ken Zarifes [EMAIL PROTECTED] wrote:
I
Load the firmware of www.dd-wrt.com on that WRT54G and then put all
the VoIP devices directly behind it.
It MIGHT work to set the first NAT router to have the 2nd NAT router
in the 1st's DMZ... but I prefer to do things The Right Way.
On Tue, Oct 7, 2008 at 7:24 AM, Steve Anness [EMAIL
On Tue, Oct 7, 2008 at 6:00 PM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
Philipp Kempgen schrieb:
Klaverstyn, David C schrieb:
Mysql for CentOS 5.2 is the mysql client tools.
mysql.i386 : MySQL client programs and shared libraries.
Does anyone have any other suggestions?
Try making sure you use the r option in your dialstring. You should
*NOT* be answering a ringing channel, as Steve suggested, FWIW (if it
doesn't work any other way that is another story)
On Tue, Oct 7, 2008 at 5:04 PM, Nhadie [EMAIL PROTECTED] wrote:
Hi,
Got this weird problem that the caller
Leif and I discussed something like this at Astricon 2008, and we came up
with
this patch:
http://bugs.digium.com/view.php?id=13632
--
Tilghman
That's a great idea. Good work.
Also, nice work with the new CDR stuff in 1.6!
So that leaves only one question:
exten = ?
What
I am not sure if it is possible to somehow invoke a function to pick
up the call via dialplan, if it is a combination of that function and
DISA should do what you need.
On Tue, Oct 7, 2008 at 8:37 AM, Vieri [EMAIL PROTECTED] wrote:
--- On Tue, 10/7/08, Anselm Martin Hoffmeister [EMAIL
Andrew Joakimsen wrote:
Try making sure you use the r option in your dialstring. You should
*NOT* be answering a ringing channel, as Steve suggested, FWIW (if it
doesn't work any other way that is another story)
Thanks, tried the 'r' and it works. And even the voicemail worked after
that. I
Hi,
Is there a way to limit only one registration for each user at a time?
meaning if a user tries to register, but that user is already
registered. i will deny?
or is it possible to for a single user at the same time, and when
someone calls that user, it will ring both phones?
Just want
What do you do to get that message?
On Tue, Oct 7, 2008 at 8:45 AM, Mr surfit [EMAIL PROTECTED] wrote:
Very new to Asterisk, on my console it says there are 47 bad
destinations...What is the best way to track these down and resolve
them
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