I have done intergration of mediant 2000 and asterisk mysetup is
[soft_ph][asterisk]---[mediant2k]E1---[mediant2k][asterisk]---[soft_ph]
This is my setup i have done all configuration and it is working fine finally i
have done all configuration on all devices
1) i have create
Yes Dave,
We want to use to principle for the following reason.
If the outbound call is not picked up, the inbound caller won't be
charged for the call, because there was no answer.
Arjan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Boyd
Sent:
I got a warning during zaptel compilation:
qozap.ko needs unknown symbol
zt_alarm_notify_no_master_change
Is this critical/what am I missing?
Thanks
Need a vacation? Get great deals
to amazing places
somebody knows, what this mean, or how to avoid this messages?
I have clock synchronized on asterisk server using ntpd.
Internal RTCP NTP clock skew detected: lsr=4103127456, now=4103296271,
dlsr=168820 (2:575ms), diff=5
Internal RTCP NTP clock skew detected: lsr=4103522652, now=4103656826,
Hi,
after updating from asterisk 1.4.4 to 1.4.5 I get a warning for
chan_features.so:
Your Asterisk modules directory, located at
/usr/lib/asterisk/modules
contains modules that were not installed by this
version of Asterisk. Please ensure that these
modules are compatible with this version
Using trixbox (or a custom dialplan if needed) has anyone been able to
convert a number dialled like
+61242110 to something like 02422110 ie (remove the +61 and
replace with 0)
i just dont know how to set it up, there seems to be no dialplan
wildcard i can use to match +.
I was
Dear all
i have one confusion about how to dial outgoing call through
asterisk like when i press 0 i got dial ton of exchange for outgoing call my
setup is
[sip_phone]-[*][mediant2k]-[Avaya_PBX]--e1-[Exchange_PSTN]
now i want to setup whn i press 0 in my sip
On Fri, Jun 22, 2007 at 12:13:42AM -0700, Vieri wrote:
I got a warning during zaptel compilation:
qozap.ko needs unknown symbol
zt_alarm_notify_no_master_change
Is this critical/what am I missing?
patches/zaptel.patch
You need a version of zaptel patched with the Bristuff patch.
--
Tracked this down (or more to the point found the issue causing it), it
was high levels of bursty disk activity.
The iowait went through the roof (30-40%).
The disks are scsi serviced by an MPT-Fusion controller in a Dell
Poweredge 2850.
We're using LVM to bind the disks into a JBOD set.
Steve
Hi all,
Can anybody tell me that wether I should install DIGIUM-TE120P card on
redhat9.0 2.4.20-8 or 2.6.18 kernel. I am using kernel 2.6.18 but facing a very
serious problem of modutils and iptable.
Can anybody help me out.
Thanx and Regards
sanchal singh
Hi, I use a isdn card with chipset HFC and now I have needed of to config
the TEI parameter to 0 (alway 0 therefore must be TEI static).
But what is the parameter that I must modify in config file ?
Thanks.
Salvatore.
___
--Bandwidth and Colocation
On 6/22/07, satish patel [EMAIL PROTECTED] wrote:
Dear all
i have one confusion about how to dial outgoing call through
asterisk like when i press 0 i got dial ton of exchange for outgoing call my
setup is
I have a company with asterisk 1.2.19 and polycom 501 phones. I get one way
audio. A caller from the pstn world hits the tdm400 card, This rings two
phones in a ring group. My client answers the phone, the calling party is
told the customer here her but she can not here them. The customer
Hi all,
Can anybody tell me that wether I should install DIGIUM-TE120P card on
redhat9.0 2.4.20-8 or 2.6.18. I am using kernel 2.6.18 but facing problem of
modutils and iptable.
Can anybody help me out of this.
Thanx and Regards
sanchal singh
On Fri, Jun 22, 2007 at 12:35:09PM +0200, Salvatore wrote:
Hi, I use a isdn card with chipset HFC and now I have needed of to config
the TEI parameter to 0 (alway 0 therefore must be TEI static).
But what is the parameter that I must modify in config file ?
Thanks.
Use ptp rather than ptmp.
Hi Folks -
This may sound weird - but here goes:
I live in Japan and on my home POTS line I have a Fax/Phone machine.
If I receive a fax, the thing automatically switches to 'fax mode' and
prints the fax.
If the call is a 'voice call', it sits there rings until answered.
The above is very
Hi,
Quick reminder that the conference is happening today at 12:30 PM EDT.
I'd like to talk more about updating to 1.4. I now have a test box
running asterisk 1.4.5, CentOS 5 and Lumenvox speech rec software.
Seems to be fine except for some double NAT issues that could be
router specific.
So I'll ask the question. What's wrong with top posting. I use a blackberry to
read most of my email, and bottom posting means excessive scrolling, often
waiting to download additional content resulting in higher usage fees and rsi
on my thumb for scrolling
90% of messages including all
On Fri, Jun 22, 2007 at 03:20:07PM +0530, [EMAIL PROTECTED] wrote:
Hi all,
Can anybody tell me that wether I should install DIGIUM-TE120P card
on redhat9.0 2.4.20-8 or 2.6.18 kernel. I am using kernel 2.6.18 but
facing a very serious problem of modutils and iptable.
What problems,
On Fri, Jun 22, 2007 at 08:06:39AM -0400, Dave Bour wrote:
So I'll ask the question. What's wrong with top posting. I use a
blackberry to read most of my email, and bottom posting means excessive
scrolling, often waiting to download additional content resulting in
higher usage
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Fri, Jun 22, 2007 at 12:13:42AM -0700, Vieri
wrote:
I got a warning during zaptel compilation:
qozap.ko needs unknown symbol
zt_alarm_notify_no_master_change
Is this critical/what am I missing?
patches/zaptel.patch
You need a
Is it just me, or is the AGI interface at cnam.got-name.com failing for
others? Anyone know how to contact them without sending postal mail or
telegram?
smime.p7s
Description: S/MIME Cryptographic Signature
___
--Bandwidth and Colocation provided
can u give me example how do i create plan for this task or job
ram [EMAIL PROTECTED] wrote:
On 6/22/07, satish patel [EMAIL PROTECTED] wrote:
Dear all
i have one confusion about how to dial outgoing call through
asterisk like when i press 0 i got dial ton of exchange for
Hi Arjan,
As I see it, the issue at hand is as follows:
You are attempting to provide a tandem service, meaning as you say no
charge to the originator unless the called party answers. However under
this circumstance you want to also provide a non-standard call treatment
to the line without an
Let's look at your message:
On Fri, Jun 22, 2007 at 08:06:39AM -0400, Dave Bour wrote:
The problem with top-posting is that answer comes before the question.
And hence you don't really know what the question was.
So I'll ask the question. What's wrong with top posting.
(The above answer
That's exactly what is happening. The *caller* is hitting #0 and
transferring the *agent* (my operator) to the new number. I don't have
the 'T' flag set [exten = s,n,Queue(queue-answer|t|||20)], so I was led
to assume that the caller could not transfer. Am I wrong?
Jay
Wes Baehr wrote:
It
On 6/22/07, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi all,
Can anybody tell me that wether I should install DIGIUM-TE120P card
on redhat9.0 2.4.20-8 or 2.6.18. I am using kernel 2.6.18 but facing
problem of modutils and iptable.
Can anybody help me out of this.
Thanx and
Klaverstyn, David C wrote:
Hi All,
Once touch recording only seems to work between extensions. When
calling an external party when pressing *1 does nothing. The person
you have called can hear 2 DTMF tones.
Is there a trick to getting once touch recording working over a zap
channel?
In my asterisk 1.4.5 chan_features.so has been installed properly...
check in your asterisk-source if /channels/chan_features.so is present
regards,
Jack escribió:
Hi,
after updating from asterisk 1.4.4 to 1.4.5 I get a warning for
chan_features.so:
Your Asterisk modules directory,
On Fri, 2007-06-22 at 05:59 -0700, satish patel wrote:
can u give me example how do i create plan for this task or job
ram [EMAIL PROTECTED] wrote:
On 6/22/07, satish patel [EMAIL PROTECTED]
wrote:
Dear all
ram wrote:
On 6/22/07, [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]*
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi all,
Can anybody tell me that wether I should install
DIGIUM-TE120P card on redhat9.0 2.4.20-8 or 2.6.18. I am using
kernel 2.6.18 but
Hi Demuel,
On 6/22/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Yeah, just the same as the sample configuration by mog. However, if I am
using a gtalk
application in asterisk to dial googletalk buddy, my voip phone is suddenly
connected to
the googletalk buddy though at the googletalk
Hi, I have Asterisk 1.4.0 using Queue App.
I use PRI connecting my Asterisk with Siemens EWSD
This was working OK but since two days I have this error:
[Jun 22 10:53:09] WARNING[8050] chan_zap.c: Ring requested on channel 0/3
already in use on span 1. Hanging up owner.
[Jun 22 10:53:09]
Bill Michaelson wrote:
Is it just me, or is the AGI interface at cnam.got-name.com failing
for others? Anyone know how to contact them without sending postal
mail or telegram?
I don't know how to contact them, but I am having the same problem.
Hi,
what about this:
when user X (Sip) call 111 extension in default context. The Asterisk
response should be in GSM codec
exten = 111,1,Set(SIP_CODEC=gsm)
exten = 111,2,Dial(SIP/.)
When user X (Sip) call 222 extension in default context. the Asterisk
response should be in G711
Thanks Lee,
That really helped me to get my project started... I am in process of
developing IVR based Notification System which is going to integrate
with my IVR based Time clock system.
Notifications will be based on, if an employee is late to clock in,
event should trigger and generate a
On Fri, 22 Jun 2007, Gary wrote:
Basically, how does the machine know if the incoming call is a fax or
voice call?
It quickly listens for fax tones - certain sequences of detection tones
from the other end that are in a particular acoustic band.
Zaptel supports this on its interfaces
Daryl Jones wrote:
Bill Michaelson wrote:
Is it just me, or is the AGI interface at cnam.got-name.com failing
for others? Anyone know how to contact them without sending postal
mail or telegram?
I don't know how to contact them, but I am having the same problem.
Is this who you mean?
Try 00 as a sub for the + in the search. That's how the chan_skype dials it so
possibly your dial range becomes:
0061|0+. on the outgoing route. Just guessing
Let me know if it works
D
Dave Bour
Desktop Solution Center
905.381.0077
[EMAIL PROTECTED]
For those who just want it to work...
Giving
Im trying to find the best hardware to run asterisk on. I see that the
compatibility list is a little dated. Any recommendations out there? This
is for a 19 phone system with 2 tdm cards…
Thanks
Hart Green
--
Internal Virus Database is out-of-date.
Checked by AVG Free Edition.
You can find detailed info about command Transfer at
http://www.voip-info.org/wiki/view/Asterisk+cmd+Transfer .
_
From: Lucian Romi [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 19, 2007 2:55 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to config SIP blind
Don Briggs wrote:
I have a company with asterisk 1.2.19 and polycom 501 phones. I get one way
audio. A caller from the pstn world hits the tdm400 card, This rings two
phones in a ring group. My client answers the phone, the calling party is
told the customer here her but she can not here
I used to run Asterisk 1.4.4 but had to revert back to 1.2.13 to
minimize a bug we were coming across. 1.4.5 looked promising, but the
hints are broken and making it so I'll likely have to go back to 1.2.13
until I get the hints fixed. I'm using Grandstream phones hints on
the parked
Dave Bour wrote:
So I'll ask the question. What's wrong with top posting.
WOW! Is this a mine field, or what?
You have stumbled into one of the hot religious arguments on just about
all lists.
There will NEVER be an agreement on which is acceptable.
Many anchored in the past hotly content
I'd like asterisk to bind to multiple ports in sip.conf. Is this
possible? Something like
bindport=5060,
Thanks
Suresh
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options
Hi Philippe,
In my /etc/asterisk/extensions.conf, I tested both
Gtalk/asterisk/googletalkbuddy and
Jingle/asterisk/googletalkbuddy.
When using Gtalk/asterisk/googletalkbuddy, it is consistent with making a
call to
googletalk buddy but it just ring once. After the ringing, it just displayed on
hi
i have configured an asterisk server which i have tested locally with x-lite
and that's ok but when i wanted to access to it since internet that hasent
taken place
knowing that my server has access to internet by a wifi router that has a
public ip address (e-g a.b.d.c) and asterisk server has
I don't know how to contact them, but I am having the same problem.
The domain is registered to Jed Stafford. If you want the domain contact
details you can do a whois.
The same Jed Stafford as SellVoip.net? Oh yes, that explains it... see archives.
--Luki
I have this in sip.conf:
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
progressinband=yes
[19256002182]
type=friend
username=19256002182
callerid=Test hone 1 +19256002182
host=dynamic
canreinvite=no
secret=password
context=test
Just out of curiosity, could you 'show queues'?
Thanks.
Wes Baehr
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore
Sent: Friday, June 22, 2007 9:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hi,
This question was posted earlier, but there was no satisfactory answer to
it. Afterwards I tried everything but to no avail.
The problem of audio going one way during the call for a few seconds is
still there.
Its Asterisk 1.2.18 hosted Dell server with no NAT.
Phones connect remotely
Luki wrote:
I don't know how to contact them, but I am having the same problem.
The domain is registered to Jed Stafford. If you want the domain contact
details you can do a whois.
The same Jed Stafford as SellVoip.net? Oh yes, that explains it... see
archives.
It looks like they're in
HI I have two servers both of which get this message on one of the lines.
Ring/Off-hook in strange state 6. The one server seems to be ok with it, but
the other one when an extension picks up there is no one there and the
incoming call keeps ringing. I tried to adjust the levels in wcfxo.c like
Everyone is going to have their sacred cow on this one so suspect you might
have opened a can of worms ...
I can tell you that I have very good results using a number of different
Intel based SuperMicro servers ... these seem to be very mundane and
extremely well behaved ... I have used both
On Fri, 22 Jun 2007, Luki wrote:
I don't know how to contact them, but I am having the same problem.
The domain is registered to Jed Stafford. If you want the domain contact
details you can do a whois.
The same Jed Stafford as SellVoip.net? Oh yes, that explains it... see
archives.
Does anybody know why Asterisk does not support inband DTMF for G.729?
Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it
for our Asterisk IVR system.
Any suggestion to solve this problem?
Gary___
--Bandwidth and
Nuance Communications has agreed to buy Tegic Communications, the
developer of the T9 predictive text input software for mobile phones,
from AOL for $265 million in cash.
http://www.wirelessweek.com/article.aspx?id=149702
Article goes on to say T9 is in use on over 2.5billion phones -
KW == Kevin Withnall [EMAIL PROTECTED] writes:
KW Using trixbox (or a custom dialplan if needed) has anyone been
KW able to convert a number dialled like +61242110 to something
KW like 02422110 ie (remove the +61 and replace with 0)
KW i just dont know how to set it up, there seems to
Hi List;
I saw sip.conf and iax.conf but I do not see a files
for H.323 IP Phones, does that mean Asterisk does not
support H.323 IP Phones?
Also, what if Asterisk need to talk with another IP
PBX that support H.323, so the IP Trunk in that case
should be H.323 IP Trunk, does Asterisk support
I have a simliar problem as the port binding question.
I have a four port parelell processing NIC, I would like to team them
together. Can I do this in asterisk if they are not actually teamed in
hardware. I would be binding to several addresses simultaniously.
Sounds like you need a new SIP carrier. G.729 has a way of
destroying inband DTMF tones.
---
Matthew Fredrickson
Software Engineer
Digium, Inc.
On Jun 22, 2007, at 1:20 PM, Gary Chen wrote:
Does anybody know why Asterisk does not support inband DTMF for G.729?
Our SIP carrier use inband
I am seeing a peculiar message on my console screen on my new installation of
Asterisk 1.4.5I would appreciate any comments.
Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS
Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS
Really destroying SIP dialog '[EMAIL
On 6/22/07, randulo [EMAIL PROTECTED] wrote:
Quick reminder that the conference is happening today at 12:30 PM EDT.
Listen to the conference here:
http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622
The little round orange Listen button will open a player. You can
also just download an mp3
I started doing HTTP queries with curl from my own AGI script and that
still works.
Their example doesn't work.
You can add this to the callerid_shell.agi script floating around.
lookup_gotname() {
out=
out=`/usr/bin/curl -s -m 2 -A Mozilla/4.0
I have a similar issue with Qwest SIP. They only support rfc2833 in g729AB,
and Asterisk is only G729A. Sprint works fine for me.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Fredrickson
Sent: Friday, June 22, 2007 3:21 PM
To: Asterisk
We are using Level 3. At this point, changing carrier is not an option.
- Original Message -
From: Matthew Fredrickson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, June 22, 2007 3:20 PM
Subject: Re:
Mojo with Horan Company, LLC wrote:
For real? I thought _ was to tell asterisk it was time for some pattern
matching:
; exact extension, exact cid
exten = 5000/19256002182,1,Answer
; any extension beginning with 5, from specific cid only
exten = _5./19256002182,1,Answer
; match exactly
Yes, of course. What happens when you dial the number, Daryl?
Daryl Jones wrote:
Bill Michaelson wrote:
Is it just me, or is the AGI interface at cnam.got-name.com failing
for others? Anyone know how to contact them without sending postal
mail or telegram?
I don't know how
On 6/22/07, Gary Chen [EMAIL PROTECTED] wrote:
We are using Level 3. At this point, changing carrier is not an option.
Gary,
I use Level(3) with G729a and RFC2833. No problems, no requirement
for inband G729.
--
Kristian Kielhofner
___
Alex,
I had this problem with a new TDM2400 card that we purchased.
Specifically I would get that message and then it would pick up the
ringing line AND the line next to it. Basically, lines 1 2 had
been cross-linked somehow. After a few weeks of trouble-shooting
with Digium tech
Hi all,
I'm having an odd problem with my polycom 301. I am initiating a call
to it with AMI Originate() function:
Action: Originate
Channel: local/[EMAIL PROTECTED]
Context: to_meetme
Exten: s
Priority: 1
Variable: dropped_conf=111
The to_meetme context is very simple:
[to_meetme]
What is the main distinction between Jingle and Gtalk here? How should I
generate the
file streamed to the SIP phone by Asterisk?
I really have no clue :). Maybe you can open a bug report so that we
can dig into this problem.
Thanks!
Philippe
___
John Novack wrote:
Dave Bour wrote:
So I'll ask the question. What's wrong with top posting.
WOW! Is this a mine field, or what?
You have stumbled into one of the hot religious arguments on just about
all lists.
There will NEVER be an agreement on which is acceptable.
Many
Ronaldo Z. Afonso wrote:
Hi all,
I just started programming using AGI and I have a simple question about
STDERR.
If I understood it right, all the messages sent to STDERR should be
shown in the Asterisk console, but using the following python code I
just can't see anything.
Nitesh Divecha wrote:
Thanks Lee,
That really helped me to get my project started... I am in process of
developing IVR based Notification System which is going to integrate
with my IVR based Time clock system.
Notifications will be based on, if an employee is late to clock in,
event
The best person to check with is Digium support. They have support matrix for
Kernel hardware on which ur card will perform.
Please check the compatibility matrix. Should work fine with
http://www.digium.com/en/supportcenter/documentation/viewdocs/TE120P
Digium support. 256-428-6000
inband is for G711 (uLaw) only.
Try rfc2833
Jon Weisman | Sales Engineer
International Bell Communications
www.ibell.net
- Original Message -
From: Matthew Fredrickson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
I have an existing Hylafax system using a mainpine 4 port board, 4 POTS lines.
Have a recently installed Asterisk system, with a dedicated T1 line. (Well,
it's really a fonality system).
What would I need to do, or where is the reading material, for what I need to
do, to convert the Hylafax
On Fri, 2007-06-22 at 17:43 -0400, Joe acquisto wrote:
I have an existing Hylafax system using a mainpine 4 port board, 4 POTS lines.
Have a recently installed Asterisk system, with a dedicated T1 line. (Well,
it's really a fonality system).
What would I need to do, or where is the
Thats exactly what i would do. install a channel bank on asterisk with
an fxs card in it and using option D of the dial app you could do DID
routing
On 6/22/07, Joe acquisto [EMAIL PROTECTED] wrote:
I have an existing Hylafax system using a mainpine 4 port board, 4 POTS
lines.
Have a recently
On 6/22/07, Douglas Garstang [EMAIL PROTECTED] wrote:
I have this in sip.conf:
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
progressinband=yes
[19256002182]
type=friend
username=19256002182
callerid=Test hone 1 +19256002182
I have Asterisk 1.4.5 and addons 1.4.1. Can anyone tell me if I can just
install addons 1.4.2 on this system without re installing Asterisk?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
clive.chan(Alpha Trilogies Networks)
Sent: Wednesday, June 20, 2007 9:06 PM
To:
Hi all,
Recently I am trying to install the Asterisk 1.4, I has some error while
loading the following modules, can some one help on those issues?
Error during loading the modules;
Basically, chan_ooh323.so, and res_config_mysql.so
[Jun 23 12:10:01] WARNING[30257] loader.c: Error loading
Hi list,
Does anyone have any advice on the following:
Incoming calls to our office come in on a SIP trunk. Since all our
offices/desks are in close proximity, we would like just a single
phone to ring when a call comes in instead of ringing every person's
phone.
Currently we've got this
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