In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
LOL I need to read the list completely too before I respond.
Hopefully you didn't waste to much time :))
Thank you anyway!
--
Tomislav Parcina
[EMAIL PROTECTED]
___
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I don't think Outlook supports doing a contact lookup from an inbound call.
I know Act! Supports that though.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tomislav Parcina
Sent: Tuesday, January 03, 2006 11:20 PM
To:
MusiconHold don't start at begin. What I can doing for setup the musiconhold start at begin?Thanks Fabio
Yahoo! Mail: gratis 1GB per i messaggi, antispam, antivirus, POP3___
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Asterisk-Users
Ray Van Dolson wrote:
On Sat, Dec 31, 2005 at 10:05:19AM +0100, Olle E Johansson wrote:
We're currently planning a new generation of chan_sip that will have a
different authentication scheme, not based on the from: header unless
it's a local policy to require the From: header to be the same
Senad Jordanovic wrote:
Since the device status system relies on it, I rewrote the
incominglimit and outgoinglimit into the combined call-limit.
The keywords incominglimit and outgoinglimit will be removed, but
call-limit will stay.
/O
Olle///
What happens when it not a simple phone/ATA but
Will that give a fuller bt's than the two below? These were done from
core dumps with asterisk compiled with dont-optimize. I can run asterisk
through gdb but at the moment running with safe_asterisk at least it
automatically restarts after a crash. Though if it will further help
sorting the
I have already looked at
Asterisk Events but no one seems to be helpful for my application.
I need to get caller and
calling number as soon as the communication is started (before call answer) but
cdr only log calls after their end.
Is there another way to
recover these numbers and to
I have * server that has public IP. Some users with their softphones
(and other with hardphones) need to connect to that * server and call
out thrue Zap lines. As far as I know when someone tries to authenticate
to * server using SIP protocol, he sends data in plain text format.
Right? How can
Tomislav Parcina wrote:
I have * server that has public IP. Some users with their softphones
(and other with hardphones) need to connect to that * server and call
out thrue Zap lines. As far as I know when someone tries to authenticate
to * server using SIP protocol, he sends data in plain
I did some research about Asterisk and High Availability and some sort
of load balancing. The High Availability issue isnn't much of a
problem. I did it with heartbeat en realtime. But the load balancing
issue is realy a problem. You want a load balancer to make decisions
based on call ID. The
On Wed, 2006-01-04 at 10:00 +0100, Olle E Johansson wrote:
Tomislav Parcina wrote:
I have * server that has public IP. Some users with their softphones
(and other with hardphones) need to connect to that * server and call
out thrue Zap lines. As far as I know when someone tries to
Hi,
I have Asterisk connected to BRI interface in parallel to my ordinary ISDN
phone. Can I make internal calls between those two without going through
telco provider and taking both voice channels ?
Thanks in advance,
regards,
Rob.
___
is it possible only monitoring call between phone A and B from phone C?
--
[EMAIL PROTECTED]
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To UNSUBSCRIBE or update options visit:
Tomislav Parcina schrieb:
I have foloved instructions at this web pages
http://www.omniis.com/ntsgr/cms/page.asp?688 and now I'm able to call
contacts from Outlook. Now I have few questions. When I place a call, my
phone rings before * tries to dial out. Is it posible that * first dials
out,
Mike McMullen wrote:
I found the problem. There was a misconfiguration in the person's
firewall that once
fixed cleaned everything up. Sorry for the wasted bandwidth.
Just for curiosity's sake, what was the misconfiguration?
--
Cheers,
Matt Riddell
Kevin P. Fleming wrote:
If the two servers service distinctly separate groups of endpoints,
they can share the same table since they won't care about the other
server's entries. If the two servers service the same endpoints but in
an active/passive arrangement, that would also work.
Can the
Alistair Cunningham wrote:
We've been asked to quote for a large cluster running Asterisk and our
ITSP in a box product. The system will be SIP throughout, with mixed
codecs.
We're considering using Dell blade servers, 1855 or similar, on the
grounds that we normally use Dell machines and
I attached the logs: any idea?
I use SjPhone + STUN and using a dinamyc DSL router with NAT but without
any firewall.
Sip read:
REGISTER sip:172.16.0.4 SIP/2.0
Via: SIP/2.0/UDP
62.0.0.1:49917;rport;branch=z9hG4bK0a03001043bac9d4586a0001
Content-Length: 0
Contact: sip:[EMAIL
Hi friends,
How i can enable and disable RBT in asterisk for SIP users.
We have linksys IP Phones but its give ring to the caller before
ringing the called phone.
--
Thank You,
Code Lover
___
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Armin Schindler wrote:
On Tue, 3 Jan 2006, Steve Beaumont wrote:
All,
I seem to have a problem with Asterisk 1.2.1.
Version 1.0.?? used to allow me to set the Type of Service bits to ef I.e.
tos=0xb8 Diffserv EF (Expedited Forwarding) bits. This seems not to be working
with 1.2.1:-
Hi,
Is there any way to get asterisk to be able to use USB devices like the
AUP-03 shown on this website?
http://www.chronos.com.tw/products/usb/Skype/skypephone.htm
Thanks,
Zoltan
--
==
Geograph (Pty) Ltd
P.O. Box 31255
Tokai 7966
South Africa
Tel:
Hi,
Thanks for the reply.
What happens is that all users are
registered with SER (a sip proxy). I have set SER up so when a user dials 0
followed by a pstn number it will be forwarded to asterisk which will forward
the call to a third party pstn gateway. I also use asterisk so that
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Example of Dial with multiple bindings for one SIP AOR, expanded by FOO:
Dial(${FOO(user1)}) = Dial(SIP/user1/10.1.1.1SIP/user1/10.2.2.2)
That's exactly the solution I've proposed many, many times in the
asterisk-dev mailing list
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
For this and another issues we reboot our phone servers every week, Saturday
02:00 am. You can do it with croon.weekly. That stopped all the issues
This will probably clear the symptoms, but it doesn't solve the problem.
Anyway, thank
On Wed, 4 Jan 2006, Steve Beaumont wrote:
Armin Schindler wrote:
On Tue, 3 Jan 2006, Steve Beaumont wrote:
All,
I seem to have a problem with Asterisk 1.2.1.
Version 1.0.?? used to allow me to set the Type of Service bits to ef
I.e.
tos=0xb8 Diffserv EF (Expedited
On Tue, 2006-01-03 at 12:42 -0600, Brent Torrenga wrote:
I use IP Kall to forward my missed cell phone calls to. This way, if my
phone is off, or out of a service area, calls will go to my * box.
Concurrently, all incoming calls to my * box cause it to dial my local
extensions at home, my
Is there anybody in this group that is using vISDN on an [EMAIL PROTECTED]
server?
I have a couple of questions, which are quite lengthy, and I do not want
to pollute this list of there's no use in asking to begin with!
TIA BRgds
--
F Peeters
PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1
On Mon, 2006-01-02 at 09:35 -0800, Don Fanning wrote:
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Moises
Silva
Action: Originate
Channel: SIP/13 -- this should be the first phone you want to ring
(your own phone usually)
I don't want it to ring a REGISTERED device
On Wed, January 4, 2006 10:58, Matt Riddell said:
Mike McMullen wrote:
I found the problem. There was a misconfiguration in the person's
firewall that once
fixed cleaned everything up. Sorry for the wasted bandwidth.
Just for curiosity's sake, what was the misconfiguration?
I'd love to
On 03/01/06, Brent Torrenga [EMAIL PROTECTED] wrote:
I use IP Kall to forward my missed cell phone calls to. This way, if my
phone is off, or out of a service area, calls will go to my * box.
Concurrently, all incoming calls to my * box cause it to dial my local
extensions at home, my
On Wed, 2006-01-04 at 12:16 +0100, Armin Schindler wrote:
On Wed, 4 Jan 2006, Steve Beaumont wrote:
Armin Schindler wrote:
On Tue, 3 Jan 2006, Steve Beaumont wrote:
All,
I seem to have a problem with Asterisk 1.2.1.
Version 1.0.?? used to allow me to set the
On 3 Jan 2006, at 19:10, Francisco Pérez Botella wrote:Hi.I will have to manage From asterisk to clients IP-phones, so biefly the idea is to multiplex voip flows in large packets and multicast them from asterisk/AP to client stations. flows from client stations to asterisk gateway go unicast. I
On Fri, 2005-12-30 at 20:04 +0530, [EMAIL PROTECTED] wrote:
Thanks a lot Mr. Alexander Lopez for your prompt attension.
I tried the same thing but it wouldnot happen. I use it as:-
exten = 12,1,Dial(Agent/12)
exten = 12,2,Hangup
where agent 12 is configured as :-
agent = 12,12, vivek
Tomislav Parcina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Example of Dial with multiple bindings for one SIP AOR, expanded by FOO:
Dial(${FOO(user1)}) = Dial(SIP/user1/10.1.1.1SIP/user1/10.2.2.2)
That's exactly the solution I've proposed many, many times in the
On Wed, 4 Jan 2006, Pete Barnwell wrote:
On Wed, 2006-01-04 at 12:16 +0100, Armin Schindler wrote:
On Wed, 4 Jan 2006, Steve Beaumont wrote:
Armin Schindler wrote:
On Tue, 3 Jan 2006, Steve Beaumont wrote:
All,
I seem to have a problem with Asterisk 1.2.1.
Tijmen,
We use SER for this to load balance across multiple Asterisks. We then
use a custom program to monitor the health of the Asterisks and update
SER's configuration should one go down. 2 SERs share a single IP address
for users to contact using heartbeat.
It works well, and we have
Linus,
Good point, we'll bear this in mind.
Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/
Linus Surguy wrote:
One thing to be aware of is that Dell blade (as well as many other
brand) servers are very heavy beasts.
In any deployment with these, check the
On 04/01/06, Alistair Cunningham [EMAIL PROTECTED] wrote:
Tijmen,
We use SER for this to load balance across multiple Asterisks. We then
use a custom program to monitor the health of the Asterisks and update
SER's configuration should one go down. 2 SERs share a single IP address
for users
Hello all,
Can this be done ?
Would setting the variable FAXRESOLUTION to a appropriate value affect
this change ?
http://www.asteriskguru.com/tutorials/rxfax.html
Variables connected with the application
LOCALSTATIONID - used by to application to identify itself to the remote end
- Original Message -
From: Matt Riddell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 04, 2006 10:11 AM
Subject: Re: [Asterisk-Users] Asterisk on Dell blade servers
Alistair Cunningham wrote:
El Miércoles, 4 de Enero de 2006 12:28, tim panton escribió:
On 3 Jan 2006, at 19:10, Francisco Pérez Botella wrote:
Hi.
I will have to manage From asterisk to clients IP-phones, so biefly
the idea
is to multiplex voip flows in large packets and multicast them from
asterisk/AP to
On Tue, Jan 03, 2006 at 06:43:16PM -0500, Michael Stearne wrote:
and when I try to update from binary:
[EMAIL PROTECTED] ~]# rpm -Uvh asterisk-1.2.1-15.rhfc3.at.i386.rpm
warning: asterisk-1.2.1-15.rhfc3.at.i386.rpm: V3 DSA signature: NOKEY,
key ID 66534c2b
error: Failed dependencies:
Hi everybody,
was anybody able to compile whole OpenH232 package under CYGWIN?
I was not able to link plugins...
Regards everybody and thank you
Mauro
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To
Dushyanth Harinath wrote:
Hello all,
Can this be done ?
Would setting the variable FAXRESOLUTION to a appropriate value affect
this change ?
http://www.asteriskguru.com/tutorials/rxfax.html
Variables connected with the application
LOCALSTATIONID - used by to application to
In article Pine.LNX.4.44.0601031441190.9209-10
@dulles1.contactgga.com, [EMAIL PROTECTED] says...
For anyone interested, our company released a PHP/MySQL based content
manager for the Cisco 79XX series IP Phones compatible with the SIP load
yesterday.
It's available via:
Hi everyone ,
I want to compile asterisk on Fedora 4 64 bit edition .
is there any one has experience on compiling asterisk on 64 bit
linux? ..
could you suggest me cpu , main board for x86 64 architecture?
Followings are my sample configuration :
Dual-Core AMD BOX OPTERON CPU
Asterisk cmd ZapBarge
ZapBarge(channel)
Lets you listens to the conversation on a specified Zap channel, or prompts if
one is not specified. You can hear them, but they can't hear you. No indication
is given to the other parties that their call is being listened to.
In article Pine.LNX.4.44.0601031441190.9209-10
@dulles1.contactgga.com, [EMAIL PROTECTED] says...
For anyone interested, our company released a PHP/MySQL based content
manager for the Cisco 79XX series IP Phones compatible with the SIP load
yesterday.
It's available via:
Subject: Re: [Asterisk-Users] Raw Hangup messages with IAX2?
Mike McMullen wrote:
I found the problem. There was a misconfiguration in the person's
firewall that once
fixed cleaned everything up. Sorry for the wasted bandwidth.
Just for curiosity's sake, what was the misconfiguration?
--
I would instead recommend the SuperMicro 1U servers - we have had a really
great run with these.
Which 1U models have you found work best? Do you know if ABE has been tested or
certified on any SuperMicro platforms?
___
--Bandwidth and Colocation
Hi,
Asterisk is new for me. I had a working configuration, but suddenly I can't
call anymore
with my voip provider. I am not aware that I changed anything in the
configuration, but
who knows. Can somebody explain me what is happening here? I changed username,
password and number.
--
On Wed, January 4, 2006 14:53, Mike McMullen said:
Subject: Re: [Asterisk-Users] Raw Hangup messages with IAX2?
Mike McMullen wrote:
I found the problem. There was a misconfiguration in the person's
firewall that once
fixed cleaned everything up. Sorry for the wasted bandwidth.
Just for
Hi,
Happy New Year to all of you!
I was wondering what would be the best recommended
entry level IP phone that
works well with * if buying say around 10
handsets.
Linksys spa-941 and the grandstreamgxp-2000look like good
phones but
I'm open to recommendations
Cheers,
Reggie
I think I have 4 options.
1, Modify chan_sip.c to update a new field in sipusers realtime table
with the status of the sip peer/user. Then use agi to dial sip calls.
Check the status field if OK then dial the fullcontact from the sip
table. If not goto voicemail or where ever else I want the
In article [EMAIL PROTECTED],
Gerald Dachs [EMAIL PROTECTED] wrote:
Hi,
Asterisk is new for me. I had a working configuration, but suddenly I can't
call anymore
with my voip provider. I am not aware that I changed anything in the
configuration, but
who knows. Can somebody explain me what
Gerald Dachs wrote:
Hi,
Asterisk is new for me. I had a working configuration, but suddenly I can't
call anymore
with my voip provider. I am not aware that I changed anything in the
configuration, but
who knows. Can somebody explain me what is happening here? I changed username,
password and
Is this normal to have entries like this on a PRI?
Jan 3 10 43 22 DEBUG[7341] Exception on 83, channel 69
Jan 3 10 43 22 DEBUG[7341] Got event Event 131126(131126) on channel 69
(index 0)
Jan 3 10 43 22 DEBUG[7341] DTMF Down '6'
Jan 3 10 43 22 DEBUG[7341]
Got my latest Linux magazine (www.linux-magazine.com) and fetured is asterisk in home network.
I've also been in contact with Novel/SUSE about their asterisk pakages. Reinhard Max the maintainer.
He has hinted at new packages for SUSE 10. The current ones work well (in production) however he is
Mike Fedyk wrote:
Can the various *RT servers be configured to use different tables so
there won't be any conflicts even if there is any client overlap between
the servers?
Yes, but I'm not sure how you'd manage failover in that situation then.
What I'm thinking of in this instance is
LJ wrote:
Which 1U models have you found work best? Do you know if ABE has been tested or
certified on any SuperMicro platforms?
It has not. If you wish to see that happen, contact SuperMicro and
arrange for them to supply some systems for certification testing; we'd
be happy to see that
Thanks, that helped
Gerald
On Wed, 04 Jan 2006 14:39:51 +
Faris Raouf [EMAIL PROTECTED] wrote:
Gerald Dachs wrote:
Hi,
Asterisk is new for me. I had a working configuration, but suddenly I can't
call anymore
with my voip provider. I am not aware that I changed anything in the
Peter Bowyer wrote:
I was thinking along the same lines, but for a dynamic setup it should
be possible to have SER/OpenSER load balance REGISTER requests
according to some strategy/metrics, and then forward INVITEs and other
call-related traffic to the 'right' back-end server.
Probably lots of
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
In Asterisk v1.2.1 check the featuremap section of the features.conf
file. You also need to add the w or W option to your Dial cmd where
appropriate. So with the feature mapping below pressing *1 would start
recording.
On 04/01/06, Alistair Cunningham [EMAIL PROTECTED] wrote:
Peter Bowyer wrote:
I was thinking along the same lines, but for a dynamic setup it should
be possible to have SER/OpenSER load balance REGISTER requests
according to some strategy/metrics, and then forward INVITEs and other
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
I don't think Outlook supports doing a contact lookup from an inbound call.
I know Act! Supports that though.
To bad that tipicly user doesn't change his faworite mail reader because
of increased functionality in VoIP ;))
Thank you
There is an idefisk for mac available for alpha testers, contact me off
list for a copy.
Zoa
tim panton wrote:
On 27 Nov 2005, at 13:43, Chris Mason (Lists) wrote:
Is there a good quality stable (not free) IAX2 client for MAC? I have
a client wants to travel and make calls and I want to
as the subject says, suppose i want to do the phone-equivalent
of cutpaste on a messagging program, i.e.
play back a prerecorded file in the middle of a conversation,
is there anything that lets me do the trick by working
on the dialplan, or i should go and write my own res_features
trick ?
On 4 Jan 2006, at 13:28, Francisco Pérez Botella wrote:El Miércoles, 4 de Enero de 2006 12:28, tim panton escribió: On 3 Jan 2006, at 19:10, Francisco Pérez Botella wrote: Hi.I will have to manage From asterisk to clients IP-phones, so bieflythe ideais to multiplex voip flows in large packets and
In article [EMAIL PROTECTED], [EMAIL PROTECTED]
says...
No. TAPI works this way. It only helps you to get rid of memorizing all
kinds of phone number, but you first have to pick up the phone for the
dialing to occur.
Well I have to get use to press speaker bottun :))
There is at least one
I've just started using AMP and found that I have a problem with escaped
characters in config files.
In particular, I have a custom config item that needs a semicolon in...
SetVar(_ALERT_INFO=info=auto-answer;delay=1)
To get the part of the line after the ; to be accepted by Asterisk as
On Wed, January 4, 2006 15:45, Tomislav Parcina said:
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
In Asterisk v1.2.1 check the featuremap section of the features.conf
file. You also need to add the w or W option to your Dial cmd
where
appropriate. So with the feature mapping
From what ive read on this list and the wiki, centos 4.x has issues with the
TE110P card ( a lot of people having issues after first reboot).Would 3.5 be
better (I know [EMAIL PROTECTED] uses this)
Am I right in saying that OS's with the 1.6 kernel still require a lot more
tinkering than those
On 27 Nov 2005, at 13:43, Chris Mason (Lists) wrote:
Is there a good quality stable (not free) IAX2 client for MAC? I
have a client wants to travel and make calls and I want to avoid
the SIP blocking that is a problem for travellers.
I have heard good things about
Tomislav Parcina wrote:
I need to dail *1 to quickly. Can that be changed?
Speed dial button or programmable button for your IP phone works...
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To UNSUBSCRIBE
I have been told the next version of SUSE will contain the 1.2.1 build.
I am unsure if the zaptel module will be ready -- but I have hight
hopes!
Per my last post... 10.0 is working very well in production -- including the auto updates.-- A.G. (Tony) NicholsI.S. Manager
Old Ringer 2 4 will be available as 9 10 (in addition to the
existing melodies) in Version 5.1 to be released in a few days. Its
better than wasting bandwidth downloading such a custom melody, as
Ringer2 seems so popular. Hope that will suffice...
Regards,
Usman.
Message: 13
Date: Tue, 3 Jan
Hmzzz... That's not my problem though, so I quess I'll need to investigate
further! :-(
Thanks for the info tho!
--
F Peeters
PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0
Cologne HFC-S pins #52, #54, #55 connected in
CounterPath's X-Pro Tapi softphone has this I think?
http://www.xten.com/index.php?menu=X-Series (select the EU region)
I think they have a trial...downloading it now.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomislav
Parcina
Sent: Wednesday,
I have a extension 981 setup for entering VoiceMailMain:
exten = 981,1,VoiceMailMain,([EMAIL PROTECTED])
exten = 981,2,HangUp()
I want to pass the calling extension to the context (extension and mailbox numbers are the same).
This dosen't seem to work. I get this in the console:
Asterisk
use ${CALLERIDNUM} instead of [mailbox]
I have a extension 981 setup for entering VoiceMailMain:
exten = 981,1,VoiceMailMain,([EMAIL PROTECTED])
exten = 981,2,HangUp()
I want to pass the calling extension to the context (extension and mailbox
numbers are the same).
This dosen't seem to
Asterisk performs echo cancellation for all incoming and outgoing calls through the T1/PRI card. However, there are some things that canstill cause echo's. We had a similar situation a with a setup much like yours. We switched to the MG2 echo canceller which helped quite a bit. But finally, after
[mailbox] does not exist
use
exten = 981,1,VoiceMailMain,(${CALLERID(num)}@usvm)
this
is provided that your callerid settings in your sip, iax, and zap configs are
correct and relect the extension calling.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hi im running several gentoo servers with Asterisk, only using IAX2
and SIP. Recently we decided to implement h323. All the necessary
dependences for oh323-0.7.3 were installed by portage (package manager
of Gentoo distro), including openh323, pwlib etc. The module is
successfully loaded (load
Hi all,
Anyone got any VoIP traffic shaping rules for m0n0wall that they could
let me look at please?
Thanks
--
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
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Asterisk-Users
My apologies. My fingers just pressed enter before i completed my
email to you. Just a few more indications. As i said, the box respond
pings, but anything else does not work, no ssh, no web, no dns, no
email. NMAP told me that all the services ports are open (22, 25, 52,
80) even the H323 1720
From: sip:[EMAIL PROTECTED]
;tag=165427961757Max-Forwards: 70To:
sip:[EMAIL PROTECTED]Keep Attention: y
our softphone is sending internal ipCheers,Giovanni Miano2006/1/4, Antonio Gallo [EMAIL PROTECTED]
:I attached the logs: any idea?I use SjPhone + STUN and using a dinamyc DSL router with NAT
On Tuesday 03 January 2006 20:14, Darrick Hartman wrote:
I'm attempting to use an asterisk box with a Digium TDM01B as voicemail
for an existing Meridian/Norstar PBX with an ATA-2 adapter. We're
having problems where hangup is not always (but sometimes) detected.
It's not detected probably
Hi Tim!
Wow, I didn't imagine that asterisk on different systems would use
different date codes for the monitor filenames -- but aah isn't asterisk ;)
My monitor filenames include the date and time, embedded as seconds
since epoch iirc:
[EMAIL PROTECTED] monitor]$ ll
On Tuesday 03 January 2006 18:34, Casey Boone wrote:
you could try setting the * box to pull timing from each pri connected
to it and set the nortel to be a master for that circuit and see if that
helps any
That's actually what he has right now, and that's not such a good idea.
Digium PRI
Greetings,
For a couple of months, we ran a pilot implementation of Asterisk with
a TE110P on a HP NetServer LP1000r with perfect audio quality and no
echo on a T1.
Encouraged by this, we purchased two identical HP DL360 servers with
TE411P cards for a production installation of Asterisk.
On Wednesday 04 January 2006 01:34, Kerry Garrison wrote:
Not at all, I am right with you. I am listening to what Digium is saying
and letting them spin their resources on it. They say they have it working,
Who at Digium is saying that POTS inband progress detection will definitely
work and
On Wed, 2006-01-04 at 11:59 -0500, Mark Phillips wrote:
Anyone got any VoIP traffic shaping rules for m0n0wall that they could
let me look at please?
Running m0n0wall-1.21 now, I used the wizard to set the base
queues/pipes/rules then added two more rules:
If Dir Proto Src Dst
All of my phones are Cisco 7960's. Each one of them occasionally show up
in the logs with the follow message:
messages:Oct 21 09:35:43 DEBUG[25423] channel.c: Driver for channel
'SIP/210-7450' does not support indication 3, emulating it
What does this mean and how do I fix it? I am using
Steve Langstaff wrote:
I've just started using AMP and found that I have a problem with escaped
characters in config files.
In particular, I have a custom config item that needs a semicolon in...
SetVar(_ALERT_INFO=info=auto-answer;delay=1)
To get the part of the line after the ;
Paul Dugas wrote:
On Wed, 2006-01-04 at 11:59 -0500, Mark Phillips wrote:
Anyone got any VoIP traffic shaping rules for m0n0wall that they could
let me look at please?
Running m0n0wall-1.21 now, I used the wizard to set the base
queues/pipes/rules then added two more rules:
I don't
Andrew Kohlsmith wrote:
On Tuesday 03 January 2006 20:14, Darrick Hartman wrote:
I'm attempting to use an asterisk box with a Digium TDM01B as voicemail
for an existing Meridian/Norstar PBX with an ATA-2 adapter. We're
having problems where hangup is not always (but sometimes) detected.
It's
Jeremy Koski wrote:
messages:Oct 21 09:35:43 DEBUG[25423] channel.c: Driver for channel
'SIP/210-7450' does not support indication 3, emulating it
What does this mean and how do I fix it? I am using asterisk-1.2.1 with
What makes you think there is something to fix? This is a DEBUG
Hello.
Im using Asterisk like IVR card application.
It works very well in h323 and SIP, but when
the IVR generate a call in SIP it show:
Jan 4 15:39:32 NOTICE[3589]: frame.c:128 __ast_smoother_feed: Dropping
extra frame of G.729 since we already have a VAD frame at the end
As I see, this is a
I have a case where I need the option g to continue
execute after the hangup (I'm using 1.2.1)
and I have the following in my extensions:
exten = 309,1,System(echo /tmp/file)
exten = 309,2,Dial(Console/dsp,,g)
exten = 309,3,System(rm -f /tmp/file)
exten = 309,4,Hangup
However, after the
Darrick Hartman wrote:
Andrew Kohlsmith wrote:
On Tuesday 03 January 2006 20:14, Darrick Hartman wrote:
I'm attempting to use an asterisk box with a Digium TDM01B as voicemail
for an existing Meridian/Norstar PBX with an ATA-2 adapter. We're
having problems where hangup is not
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