Generate $500 $2500 a month - Own Your Own Business
http://parkovani-u-letiste-praha.cz/httpwagregerw2.php?aforcamp=329
Well, this is it, Capet. kevon wingate
Wed, 16 May 2012 18:07:05
--
Not using the CDR for billing, but I do use it to see usage and to
know if it's cheaper to purchase a provider with unlimited incoming
and pay-per-minute outgoing. I disabled 'SIP Transformation' in the
SonicWall and so far so good (10/10 calls worked, more testing to be
had, stay tuned.)
On Sat,
Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode)
with a public IP address. We have our phone system setup as 172.16.2.x
that connect through the SonicWall to Asterisk. Incoming calls work
flawlessly and we no longer get one-way audio. We are only using SIP
(3 trunks now,
Ok, recompiling it now with a 1 instead of XMIT_CRITICAL. Will check
back to see if it worked. Would be nice if it did :)
Thanks,
Kurt
On Fri, Nov 7, 2008 at 3:38 PM, Doug [EMAIL PROTECTED] wrote:
At 14:15 11/7/2008, SIP wrote:
Kurt Knudsen wrote:
Specs: Asterisk 1.4.22 running behind
That seems to have sort of worked. It seems the phone decided to end
the call this time, instead of Asterisk and now the call is dangling
inside of 'sip show channels'.
So that solution didn't work :(
On Fri, Nov 7, 2008 at 4:28 PM, Kurt Knudsen [EMAIL PROTECTED] wrote:
Ok, recompiling it now
A previous issue has popped up and once again I'm out of ideas. During
the evenings it seems that the TDM channels will spike (dahdi_monitor)
and will refuse to listen for audio of any type, this includes DTMF.
The only resolution I know of is to stop Asterisk and restart the
dahdi service, but
Any updates? It still seems to happen, though not as often as it used to.
We're using Polycom 320 phones, if that makes a difference, though we did do
it with X-Lite as well.
On Sat, Oct 11, 2008 at 3:03 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Thanks, Steve,
That's what I am unsure of. I
them together.
Or some other math logic to check the result.
On incoming Set(DIALSTATUS=CHANUNAVAIL) and it'll ring busy to bandwidth(or
out of service, you can tweak this).
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt Knudsen
Sent: Monday, October 20, 2008 10
Bandwidth.com?
Thanks.
On Mon, Oct 20, 2008 at 8:30 PM, [EMAIL PROTECTED] wrote:
-- Kurt Knudsen wrote :
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and someone
tries to dial out, they cause another call to get one-way audio (the caller
hears us, we cannot hear them
:[EMAIL PROTECTED] On Behalf Of Kurt Knudsen
Sent: Monday, October 20, 2008 1:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent
one-way audio
The GotoIf works, because it does failover sometimes, just
Now that I have a new card and my echo problems are 'mostly' solved, I
have another major issue to deal with. After about an hour or so the
card will stop detecting DTMF tones on incoming calls. dahdi_monitor
shows the following:
[EMAIL PROTECTED] wctdm24xxp]# dahdi_monitor 1 -v
Visual Audio
Here's some freaky stuff coming from Areski CDR tool:
101. 2008-10-13 03:41:23 DAHDI/1... 000 unknown 000 BackGround
silence/5 s
ANSWERED 00:20
102. 2008-10-13 03:11:30 DAHDI/1... 000 unknown 000 BackGround
silence/5 s
ANSWERED 00:21
103. 2008-10-13 02:41:23 DAHDI/1...
I use the 'generic' file in Postfix to map an email address that is not in
use to someone's text messaging address. It'd be [EMAIL PROTECTED]
ie: [EMAIL PROTECTED] Then, any email that gets sent to
[EMAIL PROTECTED], will get automatically sent to that person's phone.
On Mon, Oct 13, 2008 at 3:14
since you are setting up two
separate trunks with Bandwidth, you need to limit each trunk to one call,
rather than two.
Thanks,
Steve Totaro
On Fri, Oct 10, 2008 at 9:47 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
externip messes up DTMF detection, and by messes up I mean it doesn't
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and someone
tries to dial out, they cause another call to get one-way audio (the caller
hears us, we cannot hear them). This happens 100% of the time and
Bandwidth.com doesn't offer any support. I don't see any setting that
channels are
in use, it tries to connect to the 2nd trunk and thus kills the audio.
Nothing strange came up in Wireshark or the firewall logs.
Thanks.
On Fri, Oct 10, 2008 at 5:40 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen [EMAIL PROTECTED]wrote
problems.
Thanks,
Steve Totaro
On Fri, Oct 10, 2008 at 6:32 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Hi Steve,
It's behind a NAT/Firewall but SIP translation is enabled and removing it
from behind the firewall did nothing, it still dropped calls. The calls
connect and everything works
account which
is able to handle multiple calls simultaneously?
Thanks in advance.
Kurt
_
Find a local pizza place, movie theater, and more
.then map the best route!
http://maps.live.com/?icid=hmtag1FORM=MGAC01
Hi list,
trixbox web-administration can be reached by host ip. since I am trying
trixbox on the machine where I host my website as well, can I move trixbox
main page to xxx.xxx.xxx.xxx/asterisk? which file I should move and should I
modify the file? Thanks.
Kurt
% of the time.
Any suggestion on what to look for.
I do have my reg time set for 180 seconds on the cisco ATA186.
[72459]
type=friend
username=XX
secret=X
host=dynamic
context=voice-mail
dtmfmode=rfc2833
;canreivet=yes
nat=yes
qualify=yes
Kurt
a
conversations. 50% of time I can not. Maybe I should complain to my
SIP service provider.
Kurt
---
if your connection is also used for web, email, and the worst, p2p, you
better to have qos on your router
A 488 can mean a codec miss match. Check that your Asterisk box is
configured for g729.
Kurt
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debug ccsip message
Kurt
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to the 2400 voicemail box. What I need to
understand is how to notify the other three phones that voicemail was
left on the 2400 extension.
The other three DIDs must be able to access the 2400 voicemail, and delete it.
Any ideas.
Kurt
___
--Bandwidth
Sounds like a timing issue or interop issue. Get rid of the NFAS (3rd t1 with all B channels) and make them all plain PRIs without D channel sharing.
Jason Walker [EMAIL PROTECTED] wrote:
I have looked through other postings to the user group for HDLC errors, went through what worked for other
the name.
Kurt
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Is it possible to have a bunch of people call a meetme room then have
that room call
into another conference off net. T
Kurt
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http
the phone. Does anyone have a
doc explaining how to get the phone to register to asterisk.
Thanks,
Kurt
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Matching the Correct Inbound POTS Dial Peer for DID
For DID to work correctly, make sure the incoming call matches the
correct POTS dial-peer where the command direct-inward-dial is
configured.
If your PRI has DIDs you need the command.
Kurt
I need to able to ring 30 phones at once on * plus another 10 that are
not on Asterisk.
I know I can use the
Dial(SIP/1SIP/2…SIP/30SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED]/109) but
this
seems cumbersome. Is there an easier way to do achieve this?
Kurt
Kurt
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Is it possible in * to set the Packetization period. For example: If
I want G711 to be at
10ms. Is that possible in *?
Thanks,
Kurt
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,Meetme
exten = _15551232432,3,Hangup
meetme.conf
[voice-mail]
conf = 100
Thanks
Kurt
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'conf-invalid' (language 'en')
Sep 19 13:51:23 WARNING[14066]: file.c:554 ast_readaudio_callback:
Failed to write frame
-- Playing 'conf-getconfno' (language 'en')
Any help is greatly appreciated.
Kurt
___
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I would like to know if any body is using the Polycom Soundstation IP
4000 SIP conference phone with Asterisk. I am thinking of purchasing
one.
Kurt
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Asterisk-Users
with redhat support that's why I wanted to try the switch.. anyways HI HO hi ho back to Deb I go!
know any doc put together for asterisk on debian sarge? especialy with h323 support?Michiel van Baak [EMAIL PROTECTED] wrote:
On 09:38, Mon 15 Aug 05, kurt turner wrote: ONLY ON MONDAY! Well it used to work
I had MCGP working to a ADIT 600 fine with debain sarge stable / asterisk stable - wanted to try red hat and got the below message - then I re-installed debian and am still getting the same message below - any comments are greatly appreciated - I did play with the config files with no prevail -
ONLY ON MONDAY!
Well it used to work - calls between my aaln's that is. I moved from debain to redhat (same conf. files for asterisk) and this is what I get.. looks like several errors. errors I never got before. Also asterisk isn't observing the digits as I dial them like it used to however it
A user has their unavailable message played and once that message
is over the Comedian
message is played right after. Is there any way to prevent the
Comedian message being
played if the user's unavailable/busy message is being played.
Thanks,
Kurt
Kurt
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for Asterisk.. I'm really a class 5 voice guy tryin to keep up!!
Thanks,
Kurt
__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
Asterisk-Users mailing
or if that file is in a non-standardlocation, maybe add that path to ld.so.conf and then run ldconfig againOn Wed, 2005-08-10 at 08:09, kurt turner wrote: Asterisk has been working fine for me for several weeks using MGCP to a Adit600 for intra office calling. I have recently loaded h323 and the following
appreciated.
Kurt
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] in the login
I also created a new symbolic link to point to local direct instead to default:
lrwxrwxrwx 1 root root 35 Jul 18 11:01 vm -
/var/spool/asterisk/voicemail/local
Am I missing something else.
Kurt
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this.
Kurt
#!/usr/bin/perl -w
use warnings;
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-answer(); #I tested with this command pounded and not pounded out.
my $val = $ARGV[0];
open(IN, /var/lib/asterisk/agi-bin/cme_db) or die $!;
my $ext = 0;
print STDERR $val
/voicemail/local.
Kurt
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I noticed when I call certain IVR systems, such as 1800calldhl, that
Asterisk will not
barge the prompt. Would this imply that Asterisk has an Early media
detection problem.
Is anyone else experiencing this problem. Is there a fix?
Kurt
___
Asterisk
to be able to
automatically route the call out the T1 card. Is this
possible in Asterisk. I have not seen any preference commands for Asterisk.
If not, is there a work around for this type of set up.
Kurt
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Asterisk-Users
= _940xx,3,Voicemail(u${EXTEN})
exten = _940xx,4,Hangup
exten = _940xx,103,Voicemail(b${EXTEN:1})
exten = _940xx,104,Hangup
Thanks,
Kurt
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That works. What I am tyring to do is have two separate DIDs. One is
4027 and the
other is 94207. Line 1 = DID 4027 and Line 2 = DID 94027. Dialing
4027 works to line
1 but dial 94027 gets a 486 busy.
Kurt
On 4/21/05, Henry Devito [EMAIL PROTECTED] wrote:
Don't you have to configure your
not work.
Any help is greatly appreciated.
Kurt
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.
The device is set up for DHCP.
Any suggestions.
Kurt
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(${CALLERIDNUM}=Unknown)
exten = s,5,Voicemail(u${ext})
exten = s,6,Hangup
Kurt
On Wed, 09 Mar 2005 07:34:24 +1100, Howard Lowndes [EMAIL PROTECTED] wrote:
On Wed, 2005-03-09 at 05:29, kurt x wrote:
I am trying to test how the GotoIf and $LEN functions work but am not
succeeding
To me it looks like the $LEN function is not working. When I do
verbose start to * I
see that it walks right through every step whether or not the ani is
10 digits or something else.
Would it be better to write an AGI script?
Kurt
On Wed, 09 Mar 2005 11:41:50 -0600, Chris Wade [EMAIL
I,ve gotten the GotoIf statement working now. I hard coded the value
10 in place of the ${DIGITS} varible. Worked like a charm.
Thanks to everyone who helped.
Kurt
On Wed, 09 Mar 2005 12:07:51 -0600, Chris Wade [EMAIL PROTECTED] wrote:
kurt x wrote:
[globals]
Setvar(DIGITS=10
the s,3,Gotoif does not work. It also goes through each line(
1,2,3,4,5,6,7)
Any help is greatly appreciated.
Thanks
Kurt
Asterisk CVS-HEAD-07/14/04-16:28:29 built by
[EMAIL PROTECTED] on a i686 running Linux
[globals]
${ext}=0
SetGlobalVar(DIGITS=10)
[vmail]
exten = s,1,Answer
exten = s,2
,Playback(pbx-invalid)
exten = s,2,Goto,attendant|xxx2400|3
On Tue, 08 Mar 2005 12:36:38 -0600, Dennis Webb [EMAIL PROTECTED] wrote:
Can you post your dialplan for that extension. Also, NoOp works great for
debugging these issues.
On Tue, 2005-03-08 at 12:29, kurt x wrote:
I am
Can someone explain in greater detail the following two Control
frames. The IAX2
draft document had extremely brief explanations.
LAGRQ = Lag request
POKE = Poke request.
Thanks,
Kurt
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I would like to know if the following lines represent the RTP traffic
going across,
the CODEC being used is G711ulaw, or both. The complete trap is below
the dotted lines
Thanks
Kurt
asterick*CLI
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 004 Type: VOICE Subclass: 4
Timestamp: 02570ms
.
Kurt Fankhauser
WaveLincwww.wavelinc.com114 S. Walnut
St.Bucyrus, OH 44820419-562-6405
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 266.5.0 - Release Date: 2/25/2005
___
Asterisk-Users
-7800H?
this is the link:
http://www.senao.com.tw/english/product/product_wireless01_outdoor_1.asp
?pgtl=Wirelesstp1id=02tp2id=06proid=000131
On Mon, 21 Feb 2005 23:42:30 -0600, Kristian Kielhofner [EMAIL PROTECTED]
wrote:
Kurt Fankhauser wrote:
Sounds like I'm going to have to wait and hope some
tos=lowdelay
[master]
type=friend
secret=4435
context=home
defaultip=192.168.1.2
qualify=yes
My Box B extension.conf
[home]
exten = _24xx,1,Dial(IAX2/slave:[EMAIL PROTECTED]/[EMAIL PROTECTED])
Thanks in advance
Kurt
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Title: Message
Does anyone know of
any sip wifi phones? Only one i can find that is redily availiable is the zyxel
prestige 2000w and from what i hear it is flaky.
Kurt Fankhauser
WaveLincwww.wavelinc.com114 S. Walnut
St.Bucyrus, OH 44820419-562-6405
--
No virus found in this outgoing
back on Monday and it was dead.
-Matthew
From: Kurt Fankhauser [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Mon, 21 Feb 2005 20:34:18 -0800
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] sip wifi
box and all the radio buttons under incoming calls are
greyed out. the greyed out thing seems to be my biggest problem right now, also
do you have to use a ip phone to record your greeting because this wav file
stuff isn't working.
Kurt Fankhauser
WaveLincwww.wavelinc.com114 S. Walnut
:
[Asterisk-Users] help with @home
Can you work through
a process of elimination if you record the file using an internal extension by
dialing *77 and seeing if that works?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kurt FankhauserSent
I think the box is answering calls but I don't think the digital
receptionist is working properly.
Kurt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Thompson
Sent: Sunday, February 20, 2005 3:05 PM
To: Asterisk Users Mailing List - Non
Of Kurt FankhauserSent: Sunday, February 20, 2005 9:18
PMTo: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: RE: [Asterisk-Users] help with
@home
I'll buy a IP phone
tomarrow so i can do that
-Original
Message-From:
[EMAIL PROTECTED
Title: Message
i have
got the softphone working, now i am trying to setup voicemail for my @home box,
under extension i have voicemail directory enabled but when i call that
extension it just keeps rining and never goes to voicemail
kurt
-Original Message-From:
[EMAIL
Title: Message
i got
it
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kurt
FankhauserSent: Sunday, February 20, 2005 8:01 PMTo:
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject:
RE: [Asterisk-Users] help with @home
--On February 18, 2005 11:16:11 -0600 [EMAIL PROTECTED] wrote:
On Fri, Feb 18, 2005 at 02:18:37PM +0100, Kurt Bauer wrote:
I just read thru the changelog.txt of the current CVS version and what
catched my eye was the following line: 'Adding Q.SIG switchtype option
to chan_zap
receptionist,
also when i try to setup digital receptionist via uploading wav file and
save, it says file uploaded successfully but when i go back in there nothing
is in the digital receptionist page.
Kurt Fankhauser
WaveLinc
www.wavelinc.com
114 S. Walnut St.
Bucyrus, OH 44820
419-562-6405
--
No virus
experience with * and Q.SIG and wants to share ??
Thanks a lot in advance,
best regards,
Kurt
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When I receive voicemail notification via e-mail I noticed that the
${VM_CALLERID) puts the IP address of the * box when callee info is
not present. Is there a way to have the field put Unkown caller in
instead of the IP address of the * box.
Kurt
When I access the Directory() and use it to call an extension, the
origination hears garbled or inconsistent ringing. The termination
side rings normally and the conversation is clean in both directions.
Kurt
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Asterisk
The Directory command is working properly but the ringing herd in
the origination phone is either garbled or herd infrequently. The
termination phone does ring with consistency. Any suggestion on what
might be happening.
Kurt
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[-1222644816]: sched.c:221 sched_settime:
Request to schedule in the past?!?!
[attendant]
;Main welcome message
exten = s,1,Wait(2)
exten = s,2,DigitTimeout,5
exten = s,3,ResponseTimeout,25
exten = s,4,Background(welcome_n2p1)
exten = s,5,Hangup
Thanks in advance for help,
Kurt
): No such file or directory
Kurt
On Mon, 24 Jan 2005 12:53:11 -0500, Roger Gulbranson
[EMAIL PROTECTED] wrote:
On Mon, 2005-01-24 at 12:36 -0500, kurt x wrote:
. Once the .gsm file is finished playing you can not select any of the
menu items. The
.gsm file is roughly 15 to 17 seconds long
Is there any way to encrypt the PIN numbers in voicemail.conf.
I looked at the Wiki page for voicemail.conf but it did not mention
anything about that topic.
I am not using MySQL or any other thrid party database.
Kurt
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Other then the standard sip debug is there any other
sip debug bugs like for errors, events, etc.
Kurt
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Brain,
I did what you suggested but instead of going to VoiceMailMain it
starts the begining of
my recorded message each time I press the * key.
[vmail]
exten = a,1,Voicemail(u${ext})
exten = a,2,Hangup
Kurt
On Wed, 19 Jan 2005 11:48:18 -0500, Brian Dingman [EMAIL PROTECTED] wrote:
If you
to access my message
without using the auto attendant.
Is this possible with Comedian?
The below page did help.
http://www.voip-info.org/wiki-Asterisk+cmd+VoiceMailMain
Kurt
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http
extension via SIP to an another IP PBX. SO the * does not
need to register to a server just blindly send a SIP invite to the ip
address in the SIP.CONF file: 192.168.1.1
Any help would be appricated
Kurt
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That was the ticket. The Extra ) was the problem.
Thanks Sean.
Kurt
On Tue, 18 Jan 2005 08:13:31 -0800, Sean Kennedy [EMAIL PROTECTED] wrote:
kurt x wrote:
What I am trying to do is the following: A call is sent to the * box
via a SIP invite. The * box answers via an IVR menu system
would be greatly appreciated.
Kurt
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Question: What is your reasoning for using Cisco Voice Mail instead
of Asterisk's voice mail.
IMHO it would make more sense to keep everything on Asterisk.
Kurt
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--On Thursday, November 04, 2004 04:41:53 PM +0100 Peter Svensson
[EMAIL PROTECTED] wrote:
On Thu, 4 Nov 2004, Kurt Bauer wrote:
Is your timing source set correctly? If you are connecting to the pstn
the pstn connection should be the primary timing source.
connection is to a Ericsson MD110
;-))
BTW, I see a lot of the following messages too:
!! Unknown IE 49 (cs5, Unknown Information Element)
!! Unknown IE 50 (cs5, Unknown Information Element)
If any further information is needed to narrow the problem down please let
me know.
Thanks a lot in advance,
best regards,
Kurt
--On Thursday, November 04, 2004 03:19:56 PM +0100 Peter Svensson
[EMAIL PROTECTED] wrote:
On Thu, 4 Nov 2004, Kurt Bauer wrote:
Hi list,
every now and then I get the following message in my * logs:
chan_zap.c:7379 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary
D-channel of span 1
You need to either download 12.3(11)T or 12.3(10)LD.
Kurt
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See if you have the below configure under your dial peers or voice
service voip.
If you do, then issue this command no signaling forward unconditional
signaling forward unconditional
Kurt
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and the
sip.conf.
If you have any hints, please let me know. Thanks in advance,
best regards,
Kurt
example sip.log
Sip read:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Max-Forwards: 10
Record-Route:
sip:[EMAIL PROTECTED];ftag=000cce3a7be800087fd8099f-62cc5396;lr=on
Via: SIP/2.0/UDP 83.136.32.160
label -
gotoif,$[${AGENTS_AVAIL}]?${Q}:${NO_Q)
Hope that helps and if there is an easier way of doing this please show me
how.
br,
Kurt
--On Tuesday, August 31, 2004 09:57:29 PM +0200 Robert Rozman
[EMAIL PROTECTED] wrote:
Hi,
I'd like to implement scenario to send user to operator's queue
/.
How can I change the WEB base interface to point to the voicemail
directory? I do not want to use
a symbolic link to do this.
Thanks,
Kurt
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Thanks for the hints, 'overlapdial=yes' did the trick.
br,
kurt
--On Tuesday, August 24, 2004 10:08:08 PM +0200 Peter Svensson
[EMAIL PROTECTED] wrote:
On Tue, 24 Aug 2004, Christian Victor wrote:
maybe I oversee somth. very obvious, but I'm a little puzzled about
the following 'error
if the Data network fails. In addition, 911
will always be going out the PSTN so I know I need at least one POTs
circuit. Calls inbound and outbound will always routed through the
data network.
Thanks,
Kurt
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if the Data network fails. In addition, 911
will always be going out the PSTN so I know I need at least one POTs
circuit. Calls inbound and outbound will always routed through the
data network.
Thanks,
Kurt
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' to 1.0RC2 today, but then same problem.
If any of you has any hints, please let me know.
Thanks a lot,
br,
Kurt
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, so that couldn't be the problem, or has anything
changed since April (that was the CVS version with which it worked just
fine).
Thanks,
best regards,
Kurt
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: message-summary
Content-Type: application/simple-message-summary
Content-Length: 37
Messages-Waiting: yes
Voicemail: 7/0
(no NAT) to 192.168.0
Kurt
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