I know that this has been asked before, but I couldn't find an answer ..
In the office, my 79XX phones (connected to dell / hp switches) are all
on their own separate network (i.e. we have data going through separate
switches). When they boot, they take ages on the configuring VLAN screen.
Hi,
I would like to test asterisk 1.4 development version , can
anyone send me a link to it . Thanks in advance.
Cheers,
boneyM
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I have an Asterisk box connected with a Panasonic KX-TEM824 and can not detect
HANGUP from this. Can anyone help me to get it work. Thanks!
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Hi,I've read many times multiple registration is an important missing feature in Asterisk.I'm not sure I've understood the reason(s) behind that.Could you explain ?Cheers
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THanks RR,
Am trying it right now.
But getting into all kinds of trouble!! ranging from SQL Alloc failed -
to seg faults.
I honestly donno anything abt odbc(n i seriously dont have the time to
rnd on that one, right now)
Can you or anyone else paste their config files for(related to voicemail
Hi Oliver,
just one advantage of multiple registrations : Imagine you are working
in two different departments with your time split 50/50. Now you have to
different offices. You have an office in department A but when working
for department B you are at a different one.
Now you want your personal
Hi All,
Im hoping someone can help resolve a problem that we
are having with new Cisco 7961 phones connected up to an Asterisk server.
The phones will work happily for a while, and then after a call is hungup, wont
make any further calls. If we use the Asterisk server to ping the phone,
Hi
a small question:
what is the best card for Asterisk for supply 2/4 BRI access to a old PABX ?
Thanks bye
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Can't this be solved with one extension simply ringing two different SIP devices?-brandonOn 9/15/06, Christian Mohrbacher
[EMAIL PROTECTED] wrote:Hi Oliver,just one advantage of multiple registrations : Imagine you are working
in two different departments with your time split 50/50. Now you have
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Douglas Garstang wrote:
The docs at that URL say that the dictionary has 'yes' in it... although I
don't understand how I can get replies like 'YOU HALF' if it doesn't exist in
the dictionary.
Did you read the Sphinx documentation?
Rather heavy
On 9/15/06, Noc Phibee [EMAIL PROTECTED] wrote:
a small question:
what is the best card for Asterisk for supply 2/4 BRI access to a old PABX ?
A good bri card is the quadbri of Junghanns/Beronet or Digium (haven't
tried the Digium one, but seems interesting because of the on-board
echo can..).
Hi,
I do not use queues but I have a lot of messages like that. I talked a
lot with Steve about this
It seems like Asterisk cannot agree with telco about which channels are
busy and which are not. Maybe a bug? I do not know...it seems too
strange Asterisk has a so big problem. There
In some cases : Yes.
But we have the following situation : We re using cisco 7960 phones in
each office (about 150 of them), but not every person has it's own
phone. Normally there are two employees in one office and they share one
phone, BUT have their own extension. Fortunately the Ciscos are
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Boneym wrote:
Hi,
I would like to test asterisk 1.4 development version , can anyone send me
a link to it . Thanks in advance.
This would be SVN trunk (http://www.asterisk.org/download):
Commands to check out code from our SVN repository:
#
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Matt Arnilo S. Baluyos (Mailing Lists) wrote:
On 9/13/06, Steve Totaro [EMAIL PROTECTED] wrote:
Matt Arnilo S. Baluyos (Mailing Lists) wrote:
Hello everyone,
What would be the best way to solve this error on ARI?
We are using ARI version
Hi list,
Any suggestions on how to deal correctly (socially and technically)
with users complaining about features/issues? For instance, users
complaining about echo; personally I ask the user(s) to give me all
the details when reporting echo (like; using handset/speaker,
internal/external call,
Hi,
I noticed sometimes I get the messages remote unix connection every
1or 2 seconds. I found that there is a second safe-asterisk process
which is probably trying to start/connect to asterisk.
Is there anybody who knows why (and maybe how to solve it)?
TIA
Giorgio Incantalupo
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi,
I do not use queues but I have a lot of messages like that. I talked a
lot with Steve about this
It seems like Asterisk cannot agree with telco about which channels are
busy and which are not. Maybe a bug? I do not know...it
On Thu, 14 Sep 2006, Faris Raouf wrote:
Incidentally I think there are people on this list who have no issues
with the TDM400p in the UK, but I have no idea how/why.
I have a small number of TDM400P's in the field - all with 1 FXO and 1 FXS
port, and it seems to just work, although once or
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
stoffell wrote:
Hi list,
Any suggestions on how to deal correctly (socially and technically)
with users complaining about features/issues? For instance, users
complaining about echo; personally I ask the user(s) to give me all
the details when
Hi,
I need to pass modem calls through a TDM400 card. Conecting the modem
to the FXS port (ZAP/1), it should be put through the FXO port (ZAP/4)
directly.
Even though Echo cancellation is disabled in both lines the call is
never successful. Modems speak for some time and then the line is hang
up
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I have searched this list and others, and see other pepole having this
issue. However, I have not seen how to fix it.
Sep 12 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Maximum
retries exceeded on transmission
[EMAIL
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I second this wish.
I third this wish :))
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
Hi,Who has ever programmed Shared Line Appearance option which is due with Asterisk 1.4 ?This feature should be in the trunk but I didn't dare to try it.Is it foreseeable to use it with Snom phones as this phones support SLA ?
Regards
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okk.. got it working.
the problem was that I had started out with Realtime, using Mysql.
Seems u can't use mysql and then put in odbc solely for voicemail
storage. res_odbc.conf entry decides that u r gonna use odbc for everything.
so had to replace mysql stuff with odbc in the conf files.
On Thu, Sep 14, 2006 at 10:37:59AM -0500, Rich Adamson wrote:
Try the above an see what the result is. If it does not address the
problem, at least one item has been removed from the list of
possibilities. ;)
OK, I can now replicate this without using outbound dialing at all, with a
tiny
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
No mention of Shared Line Appearance in the v1.4 new release. Anyone know
if they still plan to include it or not? Digium has been kind of quiet on
their work on that feature.
With their new Asterisk appliance running v1.4 I certainly
I'm interested in AOC (Advice of Charge) messages in Asterisk.
As far as I know, * does get AOC messages, but it's unable to do anything with
them. What I would like to know is:
- what is current status of AOC in Asterisk?
- is there any work going on AOC in Asterisk?
- is there anything I could
On Fri, Sep 15, 2006 at 09:52:02AM +0200, Giorgio Incantalupo wrote:
Hi,
I noticed sometimes I get the messages remote unix connection every
1or 2 seconds. I found that there is a second safe-asterisk process
which is probably trying to start/connect to asterisk.
Is there anybody who knows
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I know that this has been asked before, but I couldn't find an answer ..
In the office, my 79XX phones (connected to dell / hp switches) are all
on their own separate network (i.e. we have data going through separate
switches). When
On Thu, Sep 14, 2006 at 10:23:09AM -0500, Eric ManxPower Wieling wrote:
exten = _X.,1,Playback(pbx-invalid)
exten = _X.,2,Goto(s,1)
The problem with this is that all extensions now take 3 seconds longer to
answer. For example, with this extensions.conf:
[internal]
exten = 611,1,Answer()
exten
Hi,
ive tried to setup a svn trunk version of asterisk to test
voicemail with imap support and i am so far without success.
Is there _anyone_ running voicemail with IMAP Support who can
answer some basic questions?
regards,
Arnd
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In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
In some cases : Yes.
But we have the following situation : We re using cisco 7960 phones in
each office (about 150 of them)
Do Cisco phones support paging/intercom? If yes, please send me link to some
useful pages.
Now we want to give
Hi Tzafrir,
safe_asterisk use was encouraged on wiki pagesit has never given me
crash problems or something similarthe problem I have is to have two
safe_asterisk processes which causes a lot of messages inside logs.
How did you replace safe_asterisk?
Giorgio.
Tzafrir Cohen wrote:
Hi Benjamin,
Am trying to build a system, wherein users can access their profiles,
and hence voicemails thru a browser.
I am using Apache and am running it on another box and asterisk on
another. Am keeping them seperate to not have http traffic on the same
box as asterisk.
Now, my qs:
Is
On Fri, Sep 15, 2006 at 04:10:30PM +1000, Boneym wrote:
I would like to test asterisk 1.4 development version , can anyone
send me a link to it . Thanks in advance.
Try this:
(1) Open your web browser
(2) Enter www.asterisk.org
(3) Click on the link marked downloads, at the top of the
Hi
I am trying to understand why, if you don't use realtime users caching, the NAT MWI doesn't work with realtime sipfriends.
Is it because the code responsible for that doesnt work with realtime yet, or is there another thing I am missing?
thanks
Arne
BoneyM - your best bet is to download Asterisk from the SVN repository
(also known as trunk) - details are on the downloads page at
www.asterisk.org
later,
PaulH
AsteriskIT
On Fri, 2006-09-15 at 16:10 +1000, Boneym wrote:
Hi,
I would like to test asterisk 1.4 development version , can
Hello,
Why 491 pending though asterisk send INVITE to ser
proxy ?
How may I setup this configuration ?
here Is my Problem:
I want asterisk to sent none local URI to SER
My config asterisk svn-trunk:
UA===SER=ASTERISK===SER===sip URI
---INVITEINVITE-INVITE---
Giorgio Incantalupo wrote:
Hi Tzafrir,
safe_asterisk use was encouraged on wiki pagesit has never given me
crash problems or something similarthe problem I have is to have two
safe_asterisk processes which causes a lot of messages inside logs.
This is because you already have an
Hi,
ive just installed a svn trunk (r42858) and i am having problems
getting app_voicemail to even try to connect to a imap server.
Ive added the following to voicemail.conf
--
; new IMAP Stuff
imapserver=mydom.com
imapport=143
expungeonhangup=no
[..]
[default]
; Office Accounts
7709810 = 1234,
You can use Asterisk along with Ser. Asterisk for advanced features like
Voicemail and gateway, and Ser for routing SIP messages, Registrar, acc,
etc. Take a look at:
http://www.voip-info.org/wiki-Asterisk+at+large
It works!!
Regards,
Ricardo.
Tomislav Parčina wrote:
In article [EMAIL
On Thu, Sep 14, 2006 at 06:14:52PM +0530, Benjamin Jacob wrote:
Hello ppl,
Am trying to build a system, wherein users can access their profiles,
and hence voicemails thru a browser.
I am using Apache and am running it on another box and asterisk on
another. Am keeping them seperate to not
On Thu, Sep 14, 2006 at 08:33:43PM -0500, Eric ManxPower Wieling wrote:
Turn off relaxdtmf in zapata.conf if that does not help play with the
rxgain, if that does not help, play with the txgain. If the volume is
too loud or too soft on zap channels, Asterisk can sometimes miss or see
On Fri, Sep 15, 2006 at 11:02:37AM +0200, Giorgio Incantalupo wrote:
Hi Tzafrir,
safe_asterisk use was encouraged on wiki pagesit has never given me
crash problems or something similarthe problem I have is to have two
safe_asterisk processes which causes a lot of messages inside
Hi,
I need to pass modem calls through a TDM400 card. Conecting the modem
to the FXS port (ZAP/1), it should be put through the FXO port (ZAP/4)
directly.
Even though Echo cancellation is disabled in both lines the call is
never successful. Modems speak for some time and then the line is hang
up
Hi,
scenario:
Call comes in via ISDN BRI on Asterisk A. Callerid (set by zapata) is let's say
0151123456789. In the incoming context I prepend a 0 to that callerid. My snom
correctly displays 00151123456789. The call is also forwarted to Asterisk B.
On the incoming context of Asterisk B I
Hi Julian,
I know I have two process.the problem is I launch only one but
sometimes (I do not when or why) another process is launched.
Tzafrir Cohen told me to avoid safe_asterisk..I'll think about it and
then create my own launch script.
Giorgio Incantalupo
Julian Lyndon-Smith
Sorry for replying to my own post: I just switch the connection from Asterisk A
to Asterisk B from SIP to IAX without changing anything else (dialplans on both
system are the same). Now the correct callerID is logged. The behaviour changed
from 1.2.9 to 1.2.10 I suppose since this worked
Hi.
I'm using zaptel-1.2.9.1/libpri-1.2.3/asterisk-1.2.12.1 all patched with
bristuff-0.3.0-PRE1s.
What could be the problem when I get this compiler error:
-- cut ---
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g3 -Iinclude
is it a single s0 card?
how do you ring the 3 phones?
no problems with the installation of mISDN so far.
it is as easy as on Bristuff
regards
KAI
Henrik Woffinden schrieb:
Hi
Sorry... I haven't been specific enough...
I have several ISDN phones on my inside NT mode ISDN card, and I wan't
Search Daily Asterisk News for echo:
Yes, that's for the issue with echo, but I was more or less meaning
the social side, the communication with the users.. echo was an
example.. :) (bad choice maybe? :))
cheers
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prints print really to stdout?,
flushed the output?
$target = ;
print WAIT FOR DIGIT 5000\n;
$target .= STDIN;
print WAIT FOR DIGIT 5000\n;
$target .= STDIN;
print WAIT FOR DIGIT 5000\n;
$target .= STDIN;
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I've got a cisco 7960, with (amongst many others) the following in the
RINGLIST.DAT file
Foghorn foghorn.raw
I can manually select this for the ringtone. However, I was wanting to
use a normal ringtone, with foghorn being used if the call was coming in
from the
exten = 333,n,Authenticate(1234)
.
.
exten = 333,n+101,NoOp(Is this ok??)
Or i have to explicitly enumerate the priority? ... i'm searching for
doc about this.
as far as i know Auth( ) does not jump to n+101 if you dont use
Auth..(123,j)
enumrations are easier if you use somthing like
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
stoffell wrote:
Search Daily Asterisk News for echo:
Yes, that's for the issue with echo, but I was more or less meaning
the social side, the communication with the users.. echo was an
example.. :) (bad choice maybe? :))
:)
I kinda knew that
On Friday 15 September 2006 04:20, Brian Candler wrote:
it worse: I got a failure rate of about 50%, and in one case 66611 instead
of 611.
It's clear your system is possessed. Please contact your local clergyman for
help with these issues.
-A.
(seriously though, I've had this particular
Just tried it. When I run sip show channels it doesnt show any open
channels.
Thanks,
Frederik
On 14 Sep 2006, at 03:27, Bill Gibbs wrote:
Make those calls then check the CLI sip show channels and see if the
channels are stay up
-Original Message-
From: [EMAIL PROTECTED]
Ok... I got it. Someone changed the CDRs to reflect CALLERID(ANI) instead of
CALLERID(number) in 1.2.10. According to the release notes this was taken back
in 1.2.12. I do not know why this was not done for IAX as well so it would have
been consistent at least but well...
I am either going
One of the providers that I use already offers this feature via a macro
in the dail plan
http://connect.voicepulse.com/FlexRate.aspx
-Jason
On Fri, 2006-09-15 at 10:21 +0200, Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
No
I need to pass modem calls through a TDM400 card. Conecting the modem to
the FXS port (ZAP/1), it should be put through the FXO port (ZAP/4)
directly.
According to Digium, Fax calls (and modem calls) are not supported on
the TDM400 or TDM2400. They are designed for voice only. If you get it
to
I'm testing the use of static
SQL-config. Everything seems to work OK, exept these warnings:
Sep 15 15:06:20 WARNING[25374]: chan_sip.c:12829 reload_config: Section
'157217030' lacks typeSep 15 15:06:20 WARNING[25374]: chan_sip.c:12829
reload_config: Section 'TISP-157217030' lacks typeSep
Christian Mohrbacher wrote:
In some cases : Yes.
But we have the following situation : We re using cisco 7960 phones in
each office (about 150 of them), but not every person has it's own
phone. Normally there are two employees in one office and they share one
phone, BUT have their own extension.
I have gone to http://soft-switch.org/downloads/spandsp/
looking to app_rxfax and I dont see it? Where is it?
Parent Directory http://soft-switch.org/downloads/ -
[TXT] app_dtmftotext.c http://soft-switch.org/downloads/spandsp/app_dtmftotext.c17-Mar-2004
Fabian,
[EMAIL PROTECTED] escreveu:
I've try to use Astmanproxy with Asterisk TAPI line.
But login fails, astmanproxys error message:
Sep 13 20:06:26: [EMAIL PROTECTED] got: Response: Error
Sep 13 20:06:26: [EMAIL PROTECTED] got: Message: No variable specified
Sep 13 20:06:26: [EMAIL
Hello,
I am trying to write an AGI application that will transfer the caller to
a phone number on certain conditions. From what I understand (from the
astcc application and voip-info wiki), I should just be able to EXEC the
dial command. I'm having problems with this though. I send asterisk
CentOS 3.8, Asterisk 1.2.9.1, AMD Sempron(tm) Processor 3000+, mpg123-0.59r
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Why zaptel 1.2.5 and not the newer version?
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HUDLite has a very impressive interface. Works fine for me on Trixbox, but couldn't make it work on standalone Asterisk.
What kinds of bugs are there in it? Has anybody used its full version, how is it. I am thinking of upgrading one of my clients to fonality, just because of HUD's interface and
What sphinx documentation? All I could find was docs on the code, not on how to
USE the software.
-Original Message-
From: Matt Riddell (IT) [mailto:[EMAIL PROTECTED]
Sent: Friday, September 15, 2006 1:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
If you want to have a safe asterisk I would recommend using svscan
from daemontools package, more wonderfull software of D.J. Bernstein.
http://cr.yp.to/daemontools/svscan.html
Regards
On 9/15/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi Julian,
I know I have two process.the
As for FOP, when clients come to meet you after seeing attractive interfaces from other proprietary systems, its just embarrassing to show them such an ugly interface like FOP.
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Dear Steve,The phone may be looking for it's specific configuration files (not phone1.cfg, but instead 0004Fcfg {or [mac].cfg}). In our past experience, if the phone was ever formatted (fully formatted), the phone will request this from the FTP server specified. Of course confirm your
Hi,
I'm hoping to find
the answer to this here, because I believe the admin manual doesn't give
it.
I'd like to change
the led behavior on my Polycom 501 for the message waiting indicator.
Basically, I want to manipulate it to make the led always
on.
I can't find the led
pattern
Dear asterisk users,
the asterisk projects and ser enabled me to learn SIP,
I could be insulting sometimes . I must begin my
business with communigate and a French company. I
consider it regrettable that asterisk and ser could
not do it. I do hope that these open projects will
help many
cat /proc/zaptel/1are you seeing any IRQ misses?are you also seeing any HDLC errors in your asterisk debug log?Mark.On 9/8/06, Xue Liangliang
[EMAIL PROTECTED] wrote:
Hi, I just installed a TE110P ina supermicro server with Intel 945Gchipset,the customer reportedthe system has random drop
Julian Lyndon-Smith wrote:
I've got a cisco 7960, with (amongst many others) the following in the
RINGLIST.DAT file
Foghorn foghorn.raw
I can manually select this for the ringtone. However, I was wanting to
use a normal ringtone, with foghorn being used if the call was coming in
Dear asterisk users,
the asterisk projects and ser enabled me to learn SIP,
I could be insulting sometimes . I must begin my
business with communigate and a French company. I
consider it regrettable that asterisk and ser could
not do it. I do hope that these open projects will
help many
I have 2 single BRI s0 cards.
-1 in TE mode for the outside line
-1 in NT mode for the inside phones
If I dial the group with Dial(Zap/g2/,60,t) then all MSN's on all
phones ring.
But how do I dial so only MSN 10,11,12 rings?
If I dial every number as Dial(Zap/g2/10Zap/g2/11Zap/g2/12,60,t)
[EMAIL PROTECTED] a écrit :
Dear asterisk users,
the asterisk projects and ser enabled me to learn SIP,
I could be insulting sometimes . I must begin my
business with communigate and a French company. I
consider it regrettable that asterisk and ser could
not do it. I do hope that these open
Hi,
Does anyone know of any 4-wire analogue interface cards that could be
made to work with Asterisk? (I'm not averse to hacking channel drivers)
They would be used to support an always-on form of conferencing.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
[EMAIL PROTECTED] a écrit :
Dear asterisk users,
the asterisk projects and ser enabled me to learn SIP,
I could be insulting sometimes . I must begin my
business with communigate and a French company. I
consider it regrettable that asterisk and ser could
not do it. I do hope that these open
Rich Adamson wrote:
Julian Lyndon-Smith wrote:
I've got a cisco 7960, with (amongst many others) the following in the
RINGLIST.DAT file
Foghorn foghorn.raw
I can manually select this for the ringtone. However, I was wanting to
use a normal ringtone, with foghorn being used if the
Quoting Tony Mountifield [EMAIL PROTECTED]:
Does anyone know of any 4-wire analogue interface cards that could be
made to work with Asterisk? (I'm not averse to hacking channel drivers)
A T1 card to a D4 bank with something like a 4WEM or 4WTO should do the trick.
--Shane
As far as I know there is ZERO documentation and it's still too buggy to
even test. I'm pretty sure the Snom will work with it when they do get it
further along.
-Original Message-
From: Olivier [mailto:[EMAIL PROTECTED]
Sent: Friday, September 15, 2006 1:19 AM
To: Asterisk Users
Help me please..
ZT_SPANCONFIG failed on span 1: No such device or
address (6)
how can i fixed this problem.
Thank you.
JmiguelY
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Hi all,
I am getting a handle_request_invite: Failed to
authenticate user error when I attempt to receive
calls from a GSM gateway (I can successfully call
through the device VoIP-GSM from asterisk).
I have looked for a solution to this error but most
point me to adding a register line which I've
For all of us using these devices, I have some good news. There is a
self installable firmware update available from Nokia here (requires
windows box to install):
http://www.nokia.co.uk/nokia/0,1522,,00.html?orig=/softwareupdate
This seems to radically improve the behavior of the SIP
Scenario:
Two Astlinux
servers, main office/branch office. Calls come in via PSTN (ZAP)or
SIP VoIP provider always at the main office. Inbound call will ring a
number of extensions at main office and one phone located at a branch office
site. Calls are routed to the branch office via IAX
Does anybody know how to enable CallerID name passing from a Cisco
gateway (with PRI that has name and number) to an * box via SIP?
Supposedly CID name is enabled, but we can't get it passed to * and I've
googled and I can't find what I need.
___
I am on a sip line, ext 11, checking my voicemail. While I am in the midst of it, an incoming call from a pstn line, dialing 11, caller hears the internal message: "Asterisk mail box,... password" Obviously, the caller is confused. Where should I look in the configuration to avoid this ?
Just wondering if anyone has had any luck getting the cisco 7935 working
with asterisk and if so, what is the best way to go about it? on the
wiki there is talk about new software images etc, but I'm thinking those
are for the 7940 60 phones. If someone could point me in the right
direction
I'd look into Remote-Party-ID headers to affect the type of call
screening you want. I use this for caller ID blocking to/from SER but my
carrier doesn't support the name in the caller ID .
-Steve
Peder @ NetworkOblivion wrote:
Does anybody know how to enable CallerID name passing from a
Hi all,
Some questions about Asterisk Voicemail adjustments I want to make:
- how can I limit the number of voicemail messages stored per user in
their voicemail folder?
(to expire voicemail after a specified number of days I know that there
is in /contrib/scripts one script to do that)
-
I stumbled upon this yesterday while reading my usual news sites, and added it to Digg.com -- so be sure to digg it for even more exposure --
http://digg.com/tech_news/University_Dumps_Cisco_VoIP_Moving_6_000_Students_to_AsteriskThis is a great example for Asterisk, since most folks remain quiet
I am getting this error when trying to use app_txfax.
Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread 1078266208 (LWP 28837)]
0x003873a0b0df in __read_nocancel () from /lib64/tls/libpthread.so.0
(gdb) where
#0 0x003873a0b0df in __read_nocancel () from
Miles Scruggs wrote:
Just wondering if anyone has had any luck getting the cisco 7935
working with asterisk and if so, what is the best way to go about it?
on the
My testing shows it was a wasted purchase. Using CHAN_SCCP I was able
to get it to work, but not stably (i.e. keys stopped
I'm banging my head on compiling bristuff modules for Suse 10.0 with kernel
:
Linux laps1 2.6.13-15.11-smp #1 SMP Mon Jul 17 09:43:01 UTC 2006 x86_64
x86_64 x86_64 GNU/Linux
and Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s.
I get this :
laps1:~/Voipy/Bristuff/bristuff-0.3.0-PRE-1s/zaptel #
Some one recommended Sangoma E1 card, they said it has less problem for interrup conflct? Is that true according to your guys' experience?-- Regards!Liangliang
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Can anyone help me with information on how to implement or use the Attended transfer and parking calls?I have tried the extension 700 getting a number for the parked call but I was never been able to retrieve the call (don't know how) by dialing the indicated extension number.Also, we need
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