I have ser sitting on my iptables nat box and my asterisk box on the lan .
Ser does forwarding so that any requests (register,invite,ack,...) to the
nat box at 5060 r sent to my asterisk box on the lan .I can register from
outside
to my asterisk box but there is only one way audio , reason being
William, thanks for the info on macros. I'll try to implement some
macros using several different callgroups. I have in mind: ALL, all
upstairs, all downstairs, her normal domain and my normal domain.
Normal domain for me is my upstairs office, ham radio 'shack' and lab
and for her is her
Hi,I have trixbox and Audiocodes MP-124 FXS. In Asterisk console I often get this message:Got SIP response 481 Call/Transaction Does Not Exist back from
86.38.10.233
So I have traced the sip packets, and I saw that Audiocodes MP-124 FXS sends this message ė81 Call/Transaction Does Not Exist,
Hi Everyone,
Currently in my country, there is no toll free service provider. My company has been thinking of starting such a service (using Asterisk as a soft switch)but really we dont know how to go about this. Can anyone assist us with information/documentations, etc
Thanks
Chuck Bunn wrote:
Can anyone recommend a large button/type sip phone (VOIP) that an older
person could use. I have a client that needs to have large button phones
for elderly residents in her facility.
You might want to look into the original Grandstream Phone, the BT-101.
I havn't found
[Aug 31 04:32:22] NOTICE[20241]: chan_sip.c:5291 sip_reg_timeout:--
Registration for '[EMAIL PROTECTED]' timed out, trying
again (Attempt #984)
[Aug 31 04:32:23] NOTICE[20241]: chan_sip.c:9600
handle_response_register: Failed to authenticate on REGISTER to
'[EMAIL PROTECTED]' (Tries 3)
I cannot explain why I get all the time:
Got SIP response 486 Busy Here back from 192.168.250.244
I have a Wellgate 3804a there.
How can I solve it?
bye
Ronald
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Siqhamo Sifo wrote:
I have ser sitting on my iptables nat box and my asterisk box on the lan .
Ser does forwarding so that any requests (register,invite,ack,...) to the
nat box at 5060 r sent to my asterisk box on the lan .I can register from
outside
to my asterisk box but there is only one
the configuration is this :
NT
PRI
TD405P TE
A
-- B
(Asterisk)
A make a call to
B.
A can display the ID
(caller ID , example John) of B ?
these information
are exchanged in the call progress ?
B can change the
called number and communicate this change to A whene the
Hi Steven,The provider's implementation will have a bigger affect than any differences within Asterisk, e.g. how they are load-balancing and whether in fact SIP is serviced by Asterisk at all. Compared like-for-like within Asterisk we find there is not a lot in it, with each having their own pros
Hi,
Does anyone can tell me how to set the caller id shown in the callee
phone? When I use hard IP phone to make a PSTN call, the number
displayed in PSTN phone correctly using set(callerid(num)). However,
the caller id won't be displayed when I use software IP phone to PSTN.
Does any
Hello All,
Am relatively new to Asterisk, but kinda slogging my ass off on it.
My first couple of qs to begin with :
1) I tried the voicemail on no-answer thing. and my line in the
voicemail.conf, duz have an email address and also attach=yes,
5600 = 5600, Benjamin Jacob, [EMAIL
Hi,I am trying to record a speech with this command:exten=205,3,Record(speech:wav).But it records aproximately about 10 seconds and asterisk hangs up. Does somebody know how to solve this problem, I also tried with max duration, but it didn't help..
___
asterisk uses the sendmail daemon.
Make sure it is installed and working.
--
--
Steven
http://www.glimasoutheast.org
Benjamin Jacob [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Hello All,
Am relatively new to Asterisk, but kinda slogging my ass off on it.
My first couple
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Err, wasn't the patch for H.264 just changing one digit for another?
Hi Thomas,
I don't know. I should check BUG page for that.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Other cool things:
make menuconfig
Jingle/jabber support
IAX2 media transfers
new sound files
New answer machine detection (AMD)
and much much more!
Hi Matt, thank you for info!
Bye.
--
Tomislav Parčina
Lama Computers Split
On Thu, Aug 31, 2006 at 07:25:22AM -0400, Steven wrote:
asterisk uses the sendmail daemon.
A sendmail daemon. could be sendmail, postfix, exim, qmail, xmail,
smail, or whatever. Or even a non-queueing non-daemon /usr/sbin/sendmail
such as ssmtp and nullmailer .
--
Tzafrir Cohen
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Seems to be working ok on my handset for the past couple of weeks.
No major bugs, registration, xml services and MWI works etc..etc..
Have not given it a thorough testing though.
Hi Nathan,
Does it have any new options? I would like to
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Cisco released last Aug 23 the latest SIP firmware for Cisco 7960G. Any
info?
What version should I download? Is this one all right?
cmterm-7940-7960-8.4.00-sip.cop.sgn
Signed SIP Firmware for CCM versions 5.0(4) and later
--
Tomislav
Hello
I want to know which hardware I have to use in order to use a half E1 with
Asterisk (the second half will be used by a PABX PANASONIC).
I have already a succesfull experience in Asterisk with an entire E1 (TE110P
card) or 4 analogic channels (TDM400P) but I have no idea how physically
Hi,
I use here mgetty+sendfax with a modem to receive and send fax
messages. Is it possible to receive and send a fax with asterisk
directly?
I have two passive ISDN card (HFC-S chipset, one in NT mode the other
in TE-mode) and a old ELSA Microlink modem via serial on my computer.
The OS is
Matthias Fechner schrieb:
...
I use here mgetty+sendfax with a modem to receive and send fax
messages. Is it possible to receive and send a fax with asterisk
directly?
Hi,
did google for asterisk and fax show no results?
Strange!
Ok, what you need is Steve Underwood's package
spandsp and
Na, this will be fine for that... when you said 15 phones, I thought of a call center. Having queues gives you reporting tools. For what you are talking about though... the macro will be fine.
bp
On 8/31/06, Larry Alkoff [EMAIL PROTECTED] wrote:
William, thanks for the info on macros.I'll try to
Hi,
Can I do RTP Proxy in asterisk? As our requirement says that
voice packet should also go though sip server, so that billing should be
perfect.
Thanks,
Ranjeet
Thanks,
Ranjeet
The information contained in, or attached to, this e-mail, contains confidential
canreinvite=no will force all rtp packets through *.
Ranjeet Kumar wrote:
Hi,
Can I do RTP Proxy in asterisk? As our requirement says that voice
packet should also go though sip server, so that billing should be perfect.
Thanks,
Ranjeet
Thanks,
Ranjeet
The
Figured it out, so here it is for archives sake:
I set the dtmf mode to info instead of rfc2833 works with asterisk
clients and sipura (Cisco gateway sends everything rtp-nte).
Thanks to all who helped.
Ben
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
I am using format_mp3 to play mp3 files for musiconhold.
I am getting warning's like:
2006-08-31_08:53:28 WARNING[4961]: interface.c:215 decodeMP3: Junk at
the beginning of frame 49443302
Is this something to worry about?
FB
___
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Barzilai escribi:
Franciso,
can you make a call to the outside world, from the FXS port and going
out the FXO port?
I mean, without Asterisk in between. (The SPA300 can be configured that
way)
I'm asking because I remember having trouble with the SPA recognizing
that the FXO line was
it is fixed !!! i tried again this morning and it worked the first time ! it will remain a mystery
2006/8/31, shadowym [EMAIL PROTECTED]:
I don't remember all the details. I think you have to set the IP of the PC with the TFTP client as the tftp server on the phone. I seem to recall
Thanks for the sendmail tip guys.
Now the 2nd q was the more urgent one and still is.
How on earth do you edit cofigurations in Asterisk. (na.. am not talking
thru your fav editor).
Like say a web application wants to add an exten, or change the
forwarding of some extension, etc. all this
Ted Wallingford wrote:
Hi List,
I am working with an Asterisk server running on Fedora Core 4. It has
two TDM400P cards installed. There are 6 trunk ports and 2 (unused)
analog line ports. There are 5 Polycom SoundPoint 501 SIP phones
connected to the server, and a Linksys 24-port powered
Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP phone? I have upgrade my
phone and now it doesn't register with Asterisk. In full.log file I don't see
any reason why phone doesn't register.
Has anybody head problems like this one?
--
Tomislav Parčina
Lama Computers Split
Stinice 12,
I got follwing error when tried to compile asterisk
1.2.11 on redhat linux 9:
make[1]: Entering directory
`/home/voipuser/asterisk-1.2.11/db1-ast'make[1]: `libdb1.a' is up to
date.make[1]: Leaving directory
`/home/voipuser/asterisk-1.2.11/db1-ast'make[1]: Entering directory
Kevin P. Fleming wrote:
Robert Roach wrote:
I have a customer request to deploy an HP rack server (ProLiant DL
series) as the base system for an Asterisk install. They also want to
use the Digium 24xxp card. I have heard that the Digium card is
oversized and does not fit in a normal size
anyone got any views on what card I should get for a single isdn BRI
line, and the pros / cons of the card ?
Thanks.
Julian
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To UNSUBSCRIBE or update options
Hello Guys,We have a problem in configuring Sangoma A104. We want the 2 ports to beconfigured as E1 and the 2 ports as T1.We already run wancfg and configure the 2 ports as T1 and the last 2 ports as t1.
Below is the logs when we issue wanrouter restart.[EMAIL PROTECTED]:/tmp# wanrouter
I tried that image for about 5 minutes. Kept getting errors in asterisk
from the phone and it wouldn't stay registered. Rolled back to 8.0.2
and that works fine for us for now.
On Thu, 2006-08-31 at 15:30 +0200, Tomislav Parčina wrote:
Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP
Quoting Julian Lyndon-Smith [EMAIL PROTECTED]:
anyone got any views on what card I should get for a single isdn BRI
line, and the pros / cons of the card ?
I'll add to the question - anyone found any that work with ISDN in Canada, and
what provider did you get the lines from ?
If you had
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP phone? I have upgrade
my phone and now it doesn't register with Asterisk. In full.log file I don't
see any reason why phone doesn't register.
Has anybody head problems like this
Hello Roger,
* Roger Schreiter [EMAIL PROTECTED] [31-08-06 14:19]:
did google for asterisk and fax show no results?
yes I found spandsp but it will do everything in software.
Is it not a good idea to use my modem for the fax stuff?
Best regards,
Matthias
--
Programming today is a race
We have a problem in configuring Sangoma A104. We want the 2 ports to be
configured as E1 and the 2 ports as T1.
If I'm not mistaken, you can't do that with the A104D, that's why they
sold me 2 x A102 for the same price as a A104. Better check with
Sangoma.
hth
Sangoma has excellent support, why dont you ask them?
On 8/31/06, Angelito Manansala [EMAIL PROTECTED] wrote:
Hello Guys,
We have a problem in configuring Sangoma A104. We want the 2 ports to be
configured as E1 and the 2 ports as T1.
We already run wancfg and configure the 2 ports as T1 and
Does the 8.0.3 image has the same flaws as 8.0.4?
Wasn't even able to register with * at all since
most configuration examples from voip-info.org wouldn't
work...
Do you have any example config for me to try with SIP
image on 7970G?
Only tried 8.0.3 on my 7970G and had to switch to SCCP
We've been using iax with teliax.com for a couple of years, and it seems
the quality of calls varies with time. Sometimes it is good and next
time its not so good. There has been changes occurring to iax and the
jitterbuffer stuff over the last two years, and I'm reasonably certain
that some
I emailed then last 2 hours ago. Just waiting for their reply.ThanksOn 8/31/06, Moises Silva [EMAIL PROTECTED]
wrote:Sangoma has excellent support, why dont you ask them?On 8/31/06, Angelito Manansala
[EMAIL PROTECTED] wrote: Hello Guys, We have a problem in configuring Sangoma A104. We want the
Matthias Fechner schrieb:
...
yes I found spandsp but it will do everything in software.
Is it not a good idea to use my modem for the fax stuff?
Hi,
ok, you want to use an external faxmodem?
Something like that:
outside (PSTN or anythin else)
|
V
asterisk box
|
| (via
You need to update your version of libpri to the latest as well.
gc wrote:
I got follwing error when tried to compile asterisk 1.2.11 on redhat
linux 9:
gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE
Has anybody noticed that, if a call is parked; times out and returns to
the employee parking the call, but that employee fails to answer the
call for whatever reason, the caller gets hung up on?
I got the following log entry:
== Everyone is busy/congested at this time (1:1/0/0)
Aug 31
I had similar error messages when I configured an A101, using the latest
stable drivers, and found that restarting LINUX seemed to solve the problem
Seems wanrouter stop doesn't clean up after itself.
do a shutdown -r now and see if it comes up properly
John Novack
Angelito Manansala wrote:
Quoting Roger Schreiter [EMAIL PROTECTED]:
Matthias Fechner schrieb:
...
yes I found spandsp but it will do everything in software.
Is it not a good idea to use my modem for the fax stuff?
I have the configuration below and its fine (usr usb modem plugged back
into the
asterisk machine
I was expecting a more elegant answer to the 9 to dial out problem with
the Polycom 501. Sure I can change my dialplan, but that means I have to
adapt my dialplan to the phone, while the opposite seems like the way to go.
Thanks for the answer,
Mike
-Original Message-
From: [EMAIL
Hi, i'm using Asterisk 1.2.9.1 and i'm needing the AGENT function
since i need something to offer the agents a way to check if they are
logged in or not. i was specting to use AGENT function for this. and i
found out this:
asterisk*CLI show function AGENT
No function by that name registered.
Hi,Please Help me!!!I've installed TrixBox and VISDN
(snapshot 20060802) on a PC with anHFC-4s card. Outbound Calls work fine,
and inbound calls from Cellphoneswork fine too.I have a problem with
incoming calls beginning with 0 (national andinternational calls-I stay in
Italy)
Thanks in
The Flash Operator Panel (http://www.asternic.org/) can be configured to
change the color of a phone's icon to indicate whether that agent is
logged in or not. I've found it to be very useful and the agents don't
mind using that to check their status as well as the queue status (how
many
Hi all,I'm developing a Click to call Website, but now i'm getting worried with Click to Call fraud Imagine I just create one of this PhoneNumbers (extra charged numbers: like games, erotic lines...) in a remote country
Then i just go to a click to call website and start an attack
Hi all --
Perhaps I haven't been looking in the right place, but is there a T.38
capable version of app_rxfax?
I got T.38 working in passthru mode in Asterisk (thanks Steve!) with a
Sipura ATA and the PSTN switch, and so far so good. I got app_rxfax
working with the ulaw codec (which works most
Matthias Fechner wrote:
Hello Roger,
* Roger Schreiter [EMAIL PROTECTED] [31-08-06 14:19]:
did google for asterisk and fax show no results?
yes I found spandsp but it will do everything in software.
Is it not a good idea to use my modem for the fax stuff?
Why would it not be a
Title: RE: [asterisk-users] Polycom 501 config questions
Dumb question here: Why the
need to dial 9 for an outside line? If your extensions are less than 7 digits
long then you know anything "XXX." is an outside call
Maybe this isn't true everywhere, just
curious.
-Jonathan
On Thu, 2006-08-31 at 17:27 +0200, Andrea infoteam wrote:
Hi,
Please Help me!!!
I've installed TrixBox and VISDN (snapshot 20060802) on a PC with an
HFC-4s card. Outbound Calls work fine, and inbound calls from
Cellphones
work fine too.
I have a problem with incoming calls beginning
How do I get Asterisk to send streaming data, such as incoming calls, call times, etc. to a web page? I have a web app that I'm trying to use as a call manager.Thanks,David
___
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asterisk-users
Hi
when i want compile asterisk 1.2.11, i have this error :
make[1]: Leaving directory `/usr/src/asterisk-1.2.11/stdtime'
cd editline unset CFLAGS LIBS test -f config.h || CFLAGS=-O6
./configure
loading cache ./config.cache
checking for gcc... gcc
checking whether the C compiler (gcc -O6 )
Hi Please Post you Asterisk CLi when incoming is arriving.On 8/31/06, Patrick [EMAIL PROTECTED]
wrote:On Thu, 2006-08-31 at 17:27 +0200, Andrea infoteam wrote: Hi,
Please Help me!!! I've installed TrixBoxand VISDN (snapshot 20060802) on a PC with an HFC-4s card. Outbound Calls work fine, and
Jonathan k. Creasy wrote:
Dumb question here: Why the need to dial 9 for an outside line? If
your extensions are less than 7 digits long then you know anything
XXX. is an outside call
We did it, because most of the users expected it. No other reason.
Doug
--
Ben Franklin quote:
I noticed there is newer firmware for the GXP2000 so I updated (v1.1.0.16).
Release notes are dated June28. I was wondering how that phone is working
now with this latest firmware. I had sort of written it off awhile ago as
not good enough for production. Has anything changed? I doubt the
forgot to mention, it may help if you post your extensions.confAs you are using from-internal context for this calls,and you are using trixbox, have look in extensions_additional.conf and all extension_*.conf to find out your [from-internal] context.
By the way I wouldn't use the from-internal
Hi Julian,
I'm using beronet BRI cards which are good and have autoconfiguring sw
for installation. (I tried junghanns bristuff and I had more problems to
install but maybe it is been improved lately). The only little
disadvantage with beronet driver is that you have to use different
if you really DO need to dial 9 to get out because of the lengths of
your extension numbers (re: Jonathan's post) then Jerry was right -- you
have to modify the directory of the phone to 955.
Moj
Mike wrote:
I was expecting a more elegant answer to the 9 to dial out problem with
the
Hi,As far as I know you must have a look on Asterisk Manager Interface, the HTTP way to communicate with asterisk and send and receive commands/call states etcHave a look on wiki for AMI, or Asterisk Manager Interface.
On 8/31/06, David R. [EMAIL PROTECTED] wrote:
How do I get Asterisk to send
With regard to your question about adding a 9 to get the dial from the call list to work. We sis this in the dialplan by catching 10 digit numbers and adding the nine. However we have since moved away from needing the 9. I originally put it there to be consitent with our previous pbx.
On 8/31/06,
Marco Mouta [EMAIL PROTECTED] wrote:
Hi all,
I'm developing a Click to call Website, but now i'm getting worried with
Click to Call fraud Imagine I just create one of this PhoneNumbers
(extra charged numbers: like games, erotic lines...) in a remote
country
Then i just go to a click
Francisco Seratti wrote:
Hi pals, im trying to save some money
in
cellphones calls, so i bought a GSM gateway and a Sipura SPA3000
gateway.
The GSM gw is currently working, and now im trying to configure the
SPA, but every call i send, i get a 503 service unavailable.
It does that
Hi, is anybody using autofill option in queues??This option is not in the asterisk distribution.Is described in http://bugs.digium.com/view.php?id=5577I have problems with it.
Can sombody help me?Thanks, Esteban
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Polycom is announcing a technology called HD Voice in a new IP650 phone, which is basically support for G.722.What is the current status of G.722 support within Asterisk?
___
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asterisk-users mailing
Hi Everyone,
From my research I believe I am asking the impossible but perhaps I am
missing something. Any help would be greatly appreciated.
I receive many DIDs from the same SIP provider coming from the same IP.
I have a peer setup in sip.conf for this provider and this is where the
codec
On Tue, 29 Aug 2006, Benjamin Lawetz wrote:
Hello all,
we're having an issue with DTMFs being sent to Sipura's. Calls are
originating from a Cisco AS5300 being sent to asterisk which in turn sends
it to the Sipura. Connected to the Sipura is a legacy PBX (or actually shows
the same
Hello, I need some advice on the following problem I'm trying to solve:
At the office we are using 7940s as our phones, connected to an asterisk
box via SIP. Pretty standard setup, nothing fancy. Everyone has an
extension that comes out as a single line button on the phones, with the
second line
Yeah,Could be a solution! Thanks for your reply.On 8/31/06, Henry J. Cobb [EMAIL PROTECTED] wrote:
Marco Mouta [EMAIL PROTECTED] wrote: Hi all, I'm developing a Click to call Website, but now i'm getting worried with Click to Call fraud Imagine I just create one of this PhoneNumbers
(extra
Hello,
I found commands AddQueueMember and RemoveQueueMember so no need for
agent id and password. You just dial the extension and your extension
are in the game. Nice.
;Agent Login
exten = 450,1,Noop
exten = 450,n,AddQueueMember(q1)
exten = 450,n,AddQueueMember(q2)
exten = 450,n,Wait(1)
exten
service unavailable.
It does that if no line is plugged in.
Mark
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We're actually using a mix of 1.2.11 and 1.0.7 (in the process of
upgrading). The problem occurs on both versions. But I seem to have found a
solution by setting the dtmf mode to info (it's always the simple things
;-))
Thanks for the help
-Original Message-
From: [EMAIL PROTECTED]
Damien Gabrielson wrote:
Hi Everyone,
From my research I believe I am asking the impossible but perhaps I am
missing something. Any help would be greatly appreciated.
I receive many DIDs from the same SIP provider coming from the same IP.
I have a peer setup in sip.conf for this provider
Yeh, Ive been surprised that there
hasnt been more development in this space.
Is there a bounty needed to get this
happening?
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eldon Neustaeter
Sent: Thursday, 31 August 2006
12:38 PM
To:
service unavailable.
It does that if no line is plugged in.
Mark
__ NOD32 1.1733 (20060831) Information __
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Anyone have a idea ?
Noc Phibee a écrit :
Hi
when i want compile asterisk 1.2.11, i have this error :
make[1]: Leaving directory `/usr/src/asterisk-1.2.11/stdtime'
cd editline unset CFLAGS LIBS test -f config.h || CFLAGS=-O6
./configure
loading cache ./config.cache
checking for gcc...
Title: RE: [asterisk-users] Polycom 501 config questions
Pretty much like Doug said: because people expect
it.
Mike
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k.
CreasySent: August 31, 2006 11:45 AMTo: Asterisk Users
Mailing List - Non-Commercial Discussion;
Thanks!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce
ReevesSent: August 31, 2006 12:15 PMTo: Asterisk Users
Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users]
Polycom 501 config questions
With regard to your question about adding a 9 to get the
You cannot set callerid on POTs lines. You my have more luck if you
place your call via a T1 - but it's still up to your carrier. Some
VoIP providrs also allow you to set callerid on SIP calls, but you
need to check. I fear you'll have a hard time finding a carrier that
will allow you to set
On 2006-08-26 18:35:27 -0700, Tzafrir Cohen [EMAIL PROTECTED] said:
On Sat, Aug 26, 2006 at 02:02:51PM -0700, Martin Joseph wrote:
On 2006-08-22 01:59:09 -0700, Tomislav ParÄina [EMAIL PROTECTED]
said:
Hi list!
I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.
5
Hello,I am buying a server for my Asterisk PBX (with 2 Digium TE110P):- IBM x346: (2005-2006) ChipSet: INTEL E7520Bus: 800 MHz.Proccessor Support: 2.8, 3.0, 3.2, 3.4
, 3.6 y 3.8 GHz, del tipoExteneded Memory 64 Technology (EM64T)I read the Digium Compatibility List and
On 2006-08-28 00:30:22 -0700, Martin Joseph [EMAIL PROTECTED] said:
On 2006-08-24 08:43:01 -0700, Ronald Wiplinger [EMAIL PROTECTED] said:
I want that each call from PSTN goes to Asterisk to the context for
this line. Within this context can be a menu or a dial command, ...
As more I read,
On 2006-08-29 01:06:39 -0700, Tomislav Parčina [EMAIL PROTECTED] said:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
2) If the phone is answered on the first ring the call goes off to la
la land. Explaining to users (or myself) that you need to wait for
the
second audible ring on
Hi,
Seems that FOP is a great tool and the person who made it is from my
country :). But I'm having some problems configuring it. I made it
possible to connect to the Asterisk as manager. Also I see a lot of
output/input when I set debug=1.
But, at the flash interface, the button that is under
was performed on 1Gbps
ethernet with no packet loss with ulaw codec (no transcoding on the
way):
http://flz.sk.cx/audio/20060831-181241_59206988_to_.wav.mp3
Any help would be greatly appreciated. I looked at voip-info.org,
but saw no troubleshooting info for IAX.
Thanks
Hello,
Did you try a combination of qualify=yes in sip.conf and then try the
ExtensionState in the manager?
yes, I have qualify=yes in the IAX config for peers I'm interested in.
Seems like if qualify=yes or 2000... whatever, is not set then asterisk will
not always know the state of the
I write an enhanced CDR (adding AGENT, ANI, DNIS, GLOBALID (unique across
hosts), PRODUCT, PER-MINUTE, SURCHARGE, THEME, etc) at each major step in my
dialplan to MySQL and then the web pages are created dynamically using PHP (to
read from the database) and Smarty (to format for presentation).
Dear list,
it is possible to have one quadbri (with only two ports connected) and
one TDM400P card with only one FXO module connected, coexist on the same
machine ?
googling, and voip-infoing (tm) the answer seems no, anyway, maybe
lately something has changed ?
thanks so much for your support
Aaron, was the MWI working for you on 8.0.2? I've got a 7970 and 7961 sitting on a shelf because the MWI doesn't work. On the 8.0.4, it never registered, but I was able to make calls with it. I didn't try calling it, since I never saw it register. It appeared it was authenticating for outgoing
MWI has been working on our (2) 7970's, as far as I can tell. My boss
usually complains when his doesn't work, so it seems to be working fine
as far as that's concerned. The 8.0.4 firmware attempted to register,
but asterisk threw an error on a response it got back from the phone (I
don't
When the indicators are blinking it means that the op_panel.pl job isn't
running on the server. There are some init scrips located in the init/
directory of the tar-ball for Flash Operator Panel. You can copy the
appropriate script to your server's init directory (/etc/init.d on a
RedHat
Avi Miller wrote:
Hey guys,
Last week I changed my queues from using proper agents and
AgentCallbackLogin() to using the the FreePBX default with fixed
agents (which uses the Local/[EMAIL PROTECTED] style for the member=
field). I've also upgraded to Asterisk 1.2.10 and FreePBX 2.2.0 Beta 1.
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