Hi to all again,
Thanks for your quick response
as you siad "set callerid via extensions.conf by using apps available. Look at
"show applications" via the asterisk CLI."
can i write context for incoming ,
like
(orignal)
[ukincomming]
include => defaults
exten => s,1,Wait,1
Insert a Wait(2) before Answer
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes
Sent: Friday, 28 January 2005 17:30
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Caller ID in AU
Is anyone in AU successfully gett
ftp://ftp.linuxsys.com/pub/releases/FC1/asterisk-CVS/
This release includes the file permission corrections. feedback requested,
pls. write me directly thanks.
--
Andrew McRory - President/CTO
Linux Systems Engineers, Inc. - http://www.linuxsys.com
Located in beautiful Tallahassee, Florida
Of
Is anyone in AU successfully getting Caller ID from the analogue PSTN
service?
If so, what settings?
--
Howard.
http://www.voip-info.org/tiki-index.php?page=Asterisk+and+Australian+Caller+ID
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Title: Message
Hi,
Can
anyone help me with this:
I have
downloaded latest stable version of Asterisk using the asterisk-update.sh
script.
Compilation and installation passed
well.
When I
start Asterisk I get the following error:
[pbx_realtime.so]Jan 28 09:35:08 WARNING[3253]:
loader
does anyone know when current HEAD is scheduled to stabilise? Is
there a plan, or is it still "some time in the future"?
I believe I saw an announcement recently that it will start
stabilizing in February, with the goal of releasing 1.1 on the
six-month anniversary of the 1.0 release.
When was t
Does Asterisk support Instant Messaging?
How should I configure Asterisk for working as im proxy?
Thanks,
Paolo
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Hello,
I try to connect VoIP phones to Asterisk on private network,
And use Asterisk as outbound proxy via his public IP.
But the localnet and externip with nat=yes, just is not working,
I believe it might only rewrite SIP headers but does not touch
The rtp stream. Am I right ?
R.
-Origi
Hi to all again,
Thanks for your quick response
as you siad "set callerid via extensions.conf by using apps available. Look at
"show applications" via the asterisk CLI."
can i write context for incoming ,
like. can some one of you will guide me.
(orignal)
[ukincomming]
include => defaults
exte
Does any one know if attended call transfer has been added into the STABLE
release of asterisk yet?
Any news? I am also looking for #-Transfers for asterisk-stable.
Thomas
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Dear All,
All these days I was ahpppily using Asterisk with TDM11B, but from
today all of a sudden asterisk has started acting strange.
The telephone device connected to channel 1 rings continously,
following info is displayed on console
-- Starting simple switch on 'Zap/4-1'
Jan 28
I have a client that experienced quality problems and he said the
resolution turned out to be the QoS option for the nic card (even
though their backbone didn't support QoS). Try the softphones
with and without QoS to hear the difference.
> Anyone got any tips on improving
> It seems on my phone, which is hooked up to a large pbx network powered by
> an asterisk server, that it will randomly start ringing with a callerid#
> of 2013 which is its username for that phone. I have looked and been
> watching on the asterisk command line with the -vvr switch and not
Frank Sautter schrieb:
* i can't signal Busy to the calling party.
asterisk receives busy from the ericsson PBX but does not forward
this to the external caller. i tried with exten => _.,102,Busy() with
no effect. this is the part of the extensions.conf i'm using:
peter svensson gave me the hi
We have several agents on queues, and want to indicate to them that they
are logged in or logged out. We have tried several different ways, from
changing the screen to presenting different service menus, but cannot
get anything to be "in their face" (their words, not mine).
One of our team has
When I use an analog phone connected to a Sipura SPA-2000 it takes about
3-4 seconds before the number is actually dialled.
Very annoying especially if you are connecting an intercom to it.
Can I change this behaviour and do I need to look at * config or the
config of the SPA-2000?
Thanks!
Randy Johnson <> wrote on Thursday, 27 January 2005 9:02 PM:
> Hello,
>
> Here is the dial command:
>
> exten =>
> 790,2,Dial(SIP/[EMAIL PROTECTED]|60|M(screen^${CALLERIDNUM}))
>
>
> Here is the macro
>
> [macro-screen]
> exten => s,1,Wait(0.2)
> exten => s,2,say number ${ARG1}
Perhaps you
Hi!
> Call is then connected as follows.
>
> PSTN -> Provider -> Head Office -> Provider -> Remote
>
> But after it is transferred, I want the resulting route to be:
>
> PSTN -> Provider -> Remote
>
> Otherwise Head office has 2 times the bandwidth running through it for a
> call not even goin
The "delay" is a time out. The SPA does not know how many numbers it
is expecting before it has a complete number for your system. The
invite message is sent as a single message to asterisk containing the
whole number string, as apposed to each number individually.
In simple terms you have 2
Yes. This will work provided you don't need to use the light for
anything else. You are correct about sending a NOTIFY. There is
a specific field you need in the message body. I think it is called
mwi waiting but you should check the rfcs on this one. Sending a
value yes or no will turn the light o
I've tried setting the QoS settings on the card and using the Microsoft
QoS packet scheduler, in all combinations, but no changes.
I don't think these applications use QoS anyway.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Friday,
Thomas Dingermann wrote:
Does any one know if attended call transfer has been added into the
STABLE
release of asterisk yet?
Any news? I am also looking for #-Transfers for asterisk-stable.
Thomas
1.0.x is for bug fixes only. No new features are added to 1.0.x.
Blind XFER using # has been in As
I'm using Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b - when anyone picks up a
call with '*8' - the call will drop after about 20 or so seconds. Is
this a general problem with Asterisk 1.0.2?
As this is the latest release that it appears Klaus-Peter Junghanns has
for public consumption - is there anythin
On Fri, 28 Jan 2005, David John Walsh wrote:
The "delay" is a time out. The SPA does not know how many numbers it is
expecting before it has a complete number for your system. The invite
message is sent as a single message to asterisk containing the whole number
string, as apposed to each numb
Hi Mark,
please take a look at bristuff 0.2.0-RC5 which uses * 1.0.5:
http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-RC5.tar.gz
best regards
Klaus
Am Freitag, den 28.01.2005, 14:35 +0200 schrieb Mark Elkins:
> I'm using Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b - when anyone picks up a
Thanks for that - I have got the mechanism working using system calls to
touch and remove txt files in the appropriate voicemail directories.
Is there any dialplan command to do this more elegantly ?
exten => ,1,SipNotify(${CALLERIDNUM},mwi=yes)
exten => 1112,1,SipNotify(${CALLERIDNUM},mwi=no
Joseph <[EMAIL PROTECTED]> wrote:
> [snip]
> >
> > Do you get a call-waiting beep when you're on the phone with the
> > original party?
>
> I think this is it, I can hear the "beep" so that would explain why my
> phone rings when I'm using it.
>
> [snip]
>
> What am I doing wrong?
In your SPA
On Thu, 27 Jan 2005 22:10:43 -0500, David Mallwitz wrote:
>Andrew Thompson wrote:
>> David Mallwitz wrote:
>>
Their isn't any indication of whether or not clicking the Add button
will immediately add a number to my account or take me to another screen
to pick a NXX.
>>
>>
>>
>>>
Hi i was wondering the same, but one question
what do you use for instant massenging, Xten Eyebean, it is so do you
figure out
if the video works? because the eyebean besides audio and video
support instant messenging
Thank You
On Fri, 28 Jan 2005 10:15:43 +0100, Paolo Elefante <[EMAIL PROTECTE
You can also adjust the Interdigit Long Timer and Interdigit Short
Timer values found in the Regional settings config screen.
- Pedro
On Fri, 28 Jan 2005 13:36:14 +0100 (CET), Remco Barende
<[EMAIL PROTECTED]> wrote:
> On Fri, 28 Jan 2005, David John Walsh wrote:
>
> > The "delay" is a time out
Hi,
I would like to use Realtime extentions with a four bri card, the
classic "quodbri".
Normally with that card I would use * "bristuffed" from Klaus-Peter
Junghanns, but since that package is based on stable version there is
no Realtime at all in it (I suppose).
Any idea, other than wait for r
There is a way to execute an external script from the exten statement
but I
don't have the command handy. My asterisk server is down right now. Check
the documentation on this command.
Asterisk wrote:
Thanks for that - I have got the mechanism working using system calls
to touch and remove txt
Hello Stojan ,
This issue is related to a hardware problem (Zaptel ) . Probably , you
have loaded
"locked" zaptel modules or not loaded correctly.
Try to run ztcfg and see if you have any errors on your config. Check if
u have multiples
instances of mpg123 ( this kind of problem .. usually leave
Simply dial a # after the number.
Remco Barende ha scritto:
When I use an analog phone connected to a Sipura SPA-2000 it takes
about 3-4 seconds before the number is actually dialled.
Very annoying especially if you are connecting an intercom to it.
Can I change this behaviour and do I need to lo
Who has an
answer for this desperate problem? file has an answer for this desperate
problem! Who me? Yes you! So true!You wouldn't have happened to have
downgraded from CVS head to CVS stable by any chance? Stable has no idea
what realtime is... so if the old realtime modules are present, they chok
[EMAIL PROTECTED] wrote:
> The invite message is sent as a single message to asterisk
> containing the whole number string, as apposed to each number
> individually.
Does SIP support non en-bloc dialling mode?
--
Andreas SikkemaRits tele.com
Van Vollenhovenstraat 33016 BE R
Pedro,
You can also instruct your users to press the # key after dialing the number
to get the dial to start immediately.
-- Mike
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Hi
I have a simple question but I cannot find the answer.
I have a line with 2 different phone numbers
I want to redirect each phone number called to a different IP phone
Example
Someone calls 5551234 and the call is redirected to IP phone 192.168.0.2
Someone calls 5551235 and the call is redirecte
Remember I'm new here too.
You might need to edit /etc/zaptel.conf
Check fxsks=1-4 I have four X100P cards.
If you have one X100P change it to fxsks=1
I have no idea what AMP configurator is?
David
On Thu, 2005-01-27 at 12:17 -0500, Jeff R Glassman wrote:
> I also edited the Zapata.conf f
> I have a simple question but I cannot find the answer.
>
> I have a line with 2 different phone numbers
>
> I want to redirect each phone number called to a different IP phone
>
> Example
>
> Someone calls 5551234 and the call is redirected to IP phone 192.168.0.2
> Someone calls 5551235 and
I have a SER server and an * server, both have private
addresses and have static nat's on the router to the
internet. I have installed STUN (by vovida) on the SER
server by giving the SER server a second private
address on a sub interface (which is probably not
right). I understand I need a public
Long time coming, but we finally have a 3rd party interface on the
website to add block of enum numbers in regex form...
eg
+4412345[678]
which will match
+44123456
+44123457
+44123458
also
+4412345[16-18]
which will match
+441234516
+441234517
+441234518
or just short prefixes
+4412345
so anything
VAD,
Voice Activity Detection
On Thu, 2005-01-27 at 21:16 -0500, Brian Dingman wrote:
> PCMU is g711 ULAW and PCMA is G711 ALAW. ALAW is more common in
> Europe. Not sure about VAD.
>
>
> On Thu, 27 Jan 2005 18:44:09 -0700, Kim Lux <[EMAIL PROTECTED]> wrote:
> >
> > Thanks for the tips.
> >
>
Title: Message
Hi
all,
Thanks
for the information.
Yes, I
have been downgrading from HEAD to 1.0.5.
I have
removed the /usr/lib/asterisk/modules and I do not get previous error, but
apparently a new one appeared:
[cdr_tds.so]Jan 28 15:16:28 WARNING[25289]:
loader.c:258 ast_load_resourc
Video Dery / Internet du Royaume wrote:
Hi
I have a simple question but I cannot find the answer.
I have a line with 2 different phone numbers
What kind of line?
There has been some questions in the last day or so about DNIS, so I'm
not sure that it can be done on inbound analog lines.
I want to
On Thu, 2005-01-27 at 16:14 -0600, Eric Wieling wrote:
> You might consider upgrading to 1.0.5 release
Thanks, I checked it out. With same config as for 1.0 I get:
Asterisk Ready.
-- Accepting AUTHENTICATED call from 192.168.0.10, requested format = 1024,
actual format = 1024
-- Execut
Manjit, Do you have 3 lines with BroadVoice? If so how do you tell which
number is ring in on or which line to dial out on I have on line
with him now and would like to add two lines..
Thanks, David.
On Thu, 2005-01-27 at 14:14 -0800, Manjit Riat wrote:
> I had a lot of problem with them
Hi,
I have searched the list/Wiki, web and I am not able to find a decent
documentation of the AGI/FastAGI interface with examples.
Am I looking in wrong places?
Help will be greatly appreciated.
Robert
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Hi !
On Cisco 7960 (with or without 7914 add-on module) when I press speakerphone
button (or select line with line button - which automatically put second
line on speakerphone) after about 15-20 seconds of dialtone Asterisk stable
dies (seg fault). Tested versions of Asterisk are 1.0.2, 1.0.3 or 1.
> I have no idea what AMP configurator is?
http://amp.coalescentsystems.ca/
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I'm new at this too.
In my voicemail.conf under general I have attach=yes.(This works for all
users) I did try removing it and adding to the end of my users voicemail
entries. Run the test and no attachment. But I'm still new.
David
On Thu, 2005-01-27 at 18:42 -0500, Jeff R Glassman wrote:
> I a
VAD == voice activity detect
Steve
Alberto Fernandez wrote:
VAD,
Voice Activity Detection
On Thu, 2005-01-27 at 21:16 -0500, Brian Dingman wrote:
PCMU is g711 ULAW and PCMA is G711 ALAW. ALAW is more common in
Europe. Not sure about VAD.
On Thu, 27 Jan 2005 18:44:09 -0700, Kim Lux <[EMAIL PROTEC
Hi,
Just received my new SPA-841 phone and I am trying to find a
comprehensive “how-to” with Asterisk without luck. Anyone has that
working? Anyone can list high level steps or point me to a how-to somewhere ?
Thanks
Stephane
[EMAIL PROTECTED]
___
I lied it did email me an attachment. Check voice-mail entree line. it
has two comas ,, in it.
I rem out the attach=yes in my voicemail.conf file.
Then added attach=yes at the end of my entree.
101 => {passwd},David,[EMAIL PROTECTED],attach=yes
Works great..
David
On Thu, 2005-01-27 at 18:42 -
Is there any way to configure a zap channel to wait for some period of
time or number of rings before answering the line? I would like to have
a line shared between in-house emergency phones and the asterisk PBX.
Thanks.
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On Fri, Jan 28, 2005 at 09:25:55AM -0500, Andrew Thompson said:
> Video Dery / Internet du Royaume wrote:
> >Hi
> >
> >I have a simple question but I cannot find the answer.
> >
> >I have a line with 2 different phone numbers
>
> What kind of line?
>
> There has been some questions in the last da
I would like to allow my remote users to dial in from their homes,
cells, etc., and instruct Asterisk to forward calls made to their office
extension to a number of their choosing. The wiki entry on "Asterisk
call forwarding" shows how to do this. For security purposes, I would
like to front-end
Extension.conf
exten =>
11,1,Dial(SIP/11,20,D(987))
CONSOLE:
-- Executing Dial("SIP/03-69f6",
"SIP/11|20|D(987)") in new stack -- Called
11> -- SIP/11-022a is ringing --
SIP/11-022a answered SIP/03-69f6 -- Attempting native
bridge of SIP/03-69f6 and SIP/11-022aJan 28 16:05
I'm wondering why are you using SCCP and not SIP as most of us that use Cisco
7960 phones?
Martin
From: "Nenad Radosavljevic" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Problem with chan_sccp and cisco 7960
Hi !
On Cisco 7960 (with or without 7914 add-on module)
http://home.cogeco.ca/~camstuff/agi.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Augustyn
Sent: Friday, January 28, 2005 4:28 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Where can I find good doc on AGI?
Hi,
I have s
I don't think you can use NAT = yes unless there is a STUN server
involved. See my post yesterday for my Grandstream settings.
On Fri, 2005-01-28 at 10:28 +0100, Radovan.Mihalik wrote:
> Hello,
>
> I try to connect VoIP phones to Asterisk on private network,
> And use Asterisk as outbound pr
> I don't think you can use NAT = yes unless there is a STUN
> server involved. See my post yesterday for my Grandstream settings.
No, I had nat=yes working with my Cisco 7960 which did not provide it's
public IP. However, you need to tell the IP Phone to start using the IP
and port that * receiv
Hallo all
do any of you know if the toll free access to the
Netherlands is still working via FWD or Iaxtel?
thanks
liaan
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NAT=yes Rules
STUN=SUCKS
rtp streams =Rules
I have lots of devices connected behind NAT without trouble but in
fact with STUN was a real MESS
regards
Humberto
On Fri, 28 Jan 2005 10:40:25 -0500, Nabeel Jafferali
<[EMAIL PROTECTED]> wrote:
> > I don't think you can use NAT = yes unless there
On Thu, 2005-01-27 at 23:59 -0800, Mazhar Hussain wrote:
> Hi to all again,
>
> Thanks for your quick response
>
> as you siad "set callerid via extensions.conf by using apps available. Look at
> "show applications" via the asterisk CLI."
>
> can i write context for incoming ,
> like
Why don't
Stephane Ricard wrote:
Hi,
Just received my new SPA-841 phone and I am trying to find a comprehensive
"how-to" with Asterisk without luck. Anyone has that working? Anyone can
list high level steps or point me to a how-to somewhere ?
This assumes that Asterisk and the phone are on the SAME LAN a
I'm wondering why are you using SCCP and not SIP as most of us that use
Cisco 7960 phones?
Martin
Mostly because 7914 addon module is not supported in SIP images for
7960. Alternative, SIP solution, for a device like 7960+7914 could be Snom
220 + Keypad 220, but I still didn't managed to get
On Thu, 2005-01-27 at 20:35 +0100, Bruno Hertz wrote:
> Anybody found a way around this (bug?), i.e. avoiding retries with
> Queue(...|t) properly timing out at the same time ?
OK, I took a look at app_queue.c, and while the described behavior isn't
a bug, I still hacked the source to give me a d
From the asterisk demo:
exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,1,Answer ; Answer the line
You can also wait 10 sec, 30 sec, etc... to allow as many rings
as you like.
Cheers,
Jon.
On Friday 28 January 2005 09:08 am, Steven P. Don
Hello list,
Here is the error i’m getting when i try to « make
install » with astcc. Can somebody know this error and how to fix it?
[EMAIL PROTECTED] astcc]# make install
mkdir -p /var/www
mkdir -p /var/www/html/_astcc
mkdir -p /var/www/cgi-bin/astcc-admin
chmod 755 ./astcc.agi
c
I tried everything and only got that configuration with all bv servers
listed to work.
-Original Message-
From: Luki [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 27, 2005 8:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Stumped by Broa
Well, it seems to be acting normal since I wrote the message yesterday. On
your thoughts, I haven't messed with ethereal yet, so I am not sure about
that. Plus I am not sure if the network item that this phone plugs into is
a hub or a switch, I think it is a switch though. I know ethereal won't
So when should you receive a NOANSWER back? Doesn't that imply you are
using DIAL with a timeout value? Otherwise I can't see how you would
ever get there.
I agree with you about LiveVoip. They claim to be an Asterisk service
provider but anytime you have a problem they tell you that asterisk is
f
None just one single line.
-Original Message-
From: David Shaw [mailto:[EMAIL PROTECTED]
Sent: Friday, January 28, 2005 6:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Stumped by BroadVoice SIP {Scanned}
Manjit, Do you have 3 lines with B
Adam Robins wrote:
I would like to allow my remote users to dial in from their homes,
cells, etc., and instruct Asterisk to forward calls made to their office
extension to a number of their choosing. The wiki entry on "Asterisk
call forwarding" shows how to do this. For security purposes, I would
Roy Sigurd Karlsbakk wrote:
When was this?
Sorry, I don't remember when... it may have been on Asterisk Daily News.
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Hi!
I have 1.0.3-BRIstuffed and hfc-s card ztcfg says that card is configured (2
B and 1 D channel) but asterisk don't pickup any calls.
Any ideas?
Regards,
Corvin
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Title: Minimum Setup
Hi all, I have asterisk installed and working just fine with a couple of Cisco IP Phones. I am now ready to pilot connectivity to PSTN and am wondering what hardware would be recommended to make minimum connectivity to the public telephone network. I am think ISDN as I
Hi,
is it possible, that my * identifies himself as ip-phone? I.e. Im using
a grandstrem 100 phone and if I use * as proxy, the authentification
string should be changed.
Im not sure where looking for this.
Hfh,
Oliver
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A
Contacted Scott Willard at Polycom this morning, he has since been
reassigned to other duties within the organization. Mr. Willard's tone
seemed optimistic, and he referrred me to Roger Austin, Regional Channel
Manager for Voice.
Roger's reply to my inquiry is as follows:
"Cory,
We appreciate
Hello... I'm having problems with H323/G729 setup. Below is the output
of h.323 debug when making a call. I use a SIP phone connected to an *
box in the same LAN. The * connects to a h323/g729 PSTN terminator
through internet. Calls rings and are answered in the other side, but I
get no sound at al
Hi all,
I would like to connect in sip mode an Eyebeam client to a messenger via
Asterisk.
I want to use video.
Nat is not an issue as vpn connections will be used.
Is this a difficult tasks, can someone give me some pointers to get
started...
Have a good week-end,
Francois
Random Thought:
Hi,
I was messing around with FireFly last night and got asterisk to crash
hard. It looks like the bug is a division by zero in chan_iax2.c.
I reproduced it and here are some infos I got from gdb:
[Switching to Thread 245775 (LWP 23251)]
0x41154918 in calc_timestamp (p=0x816b710, ts=0, f=0x424eef
> dzien dobry!
>
raczej dobry wieczór it means good evening :-)
> what does it say on the console when you start asterisk with
> asterisk -c -vvv
>
> ?
>
>
> that should get you further :)
Heh, that is problem I've compilled bristuff, launch make load for TE mode.
I also added to modules.c
Hi Michael and Everyone
I'm trying to connect Asterisk to a CISCO AS5350 using oh323 and I'm getting
this error
"reason 24 (Call ended with Q.931 cause)"
I've checked the Asterisk wiki and several other resources. Please can
anyone give me a hint on what the problem is I reach my wits end. Th
did you find how to configure video with eyebeam using asterisk
because i wasn`t able to do it yet
as well i want to se messangin with it
ThanK You
On Fri, 28 Jan 2005 13:23:46 -0500 (EST), Francois Meehan
<[EMAIL PROTECTED]> wrote:
> Hi all,
>
> I would like to connect in sip mode an Eyebeam
Eric Wieling aka ManxPower wrote:
Stephane Ricard wrote:
Hi,
Just received my new SPA-841 phone and I am trying to find a
comprehensive
"how-to" with Asterisk without luck. Anyone has that working? Anyone can
list high level steps or point me to a how-to somewhere ?
This assumes that Asterisk
> that should get you further :)
>
, so I wan't to try.
> i4l causes a lot o echo and pure quality
ehh, want and poor quality :/
BR,
Corvin
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Hi!
> I'm testing ChanIsAvail context and it is not working for me.
>
> exten => 55,1,ChanIsAvail(SIP/11&SIP/21)
> exten => 55,2,Cut(theChannel=AVAILCHAN,,1)
> exten => 55,3,Dial(${theChannel},r)
> exten => 55,4,Hangup
> exten => 55,102,Goto(s,4)
>
> According to notes:
> The channels are checke
Turn on "debug isdn q931" "term mon" on your 5350. It is an ISDN
signalling error. Strange it is showing up in asterisk through a 323
trunk though...
What happens when you do a "csim start xxx" where xx = phone
number to dial from the 5300?
-Greg
Tola Ogunsan wrote:
Hi Michael a
Hi everybody,
I am trying to make up call flow diagrams for for a setup which
include ser as a sip proxy/registrar and asteriks as a voicemail
server.
Is my sequence correct?:
UA 1 send an invite to SER. SER forwards this invite to UA2. UA2
sends back a sends back a 100 trying and 180 ringing mes
Just installed V 0.4 of [EMAIL PROTECTED]
Programmed up 3 sip budgetone extensions, they call call each other
fine.
Tried to dial '9' for an outside line through an X100P to a packet8 ATA
but got 'all circuits are busy now'.
Here is the console output.
== Spawn extension (from-internal, h, 1) e
www.sans.org has two vulnerabilities in OpenH323, one as 'high', one as
'other'.
Jason Sjobeck
ICQ 5579183
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Just installed festival from source and the voice is very
jittery and I get this a lot in the asterisk CLI (at least once on every call)
NOTICE[3236]:
rtp.c:430 ast_rtp_read: RTP: Received packet with bad
UDP checksum
Maybe the packets are malformed so I get the jittery sound.
Thanks,
I have seen that but this is over 2 years old, does it mean that it is still
current?
robert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vassil Kolarov
Sent: Friday, January 28, 2005 10:32 AM
To: 'Asterisk Users Mailing List - Non-Commercial D
Hi
We have a fairly simple Asterisk setup for a callcenter: around 10-15
operators running SIP softphones (X-Pro) and an Asterisk box connected
to a E1 service using a Digium T100P, and to our legacy PBX (NEC) over
a Digium TDM400P FXO interfaces.
Everything is working except our Music on Hold af
Hi KPJ,
btw, there is a problem with "make loadNT" (zaphfc) and Kernel 2.4
systems that should be fixed. I hope you already know about this
"IRQ_NONE" issue! the problem is with line 578 in zaphfc.c
saturn:/usr/src/bristuff-0.2.0-RC5/zaphfc # make loadNT
cc -c zaphfc.c -D__KERNEL__ -DMODULE -DEX
Been using them for just over a month for all outbound calls. Their
customer service is prompt and courteous. I use Voicepulse for inbound
and turn around for the average ticket I open is about three days
whereas for Sixtel it's often been same day.
As for the 800 numbers...I don't have one. Bu
Nat=yes with the phone behind a nat box and asterisk on a registered
IP works just fine with Cisco, Snom, Xlite and others (I haven't tried
many of the others, however).
> I don't think you can use NAT = yes unless there is a STUN server
> involved. See my post yesterday
[snip]
> > > Do you get a call-waiting beep when you're on the phone with the
> > > original party?
> >
> > I think this is it, I can hear the "beep" so that would explain why my
> > phone rings when I'm using it.
> >
> > [snip]
> >
> > What am I doing wrong?
>
> In your SPA-3000 setup screens
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