Hello,
Is it possible to run Asterisk in a cluster?
OpenMosix or other cluster software.
Thanks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE
Dear Matt,
Yes indeed I did I have used cvs to download asterisk and it's addon
from CVS.
Thx
MAG
Matthew Boehm wrote:
Did you install res_config_mysql.so from asterisk-addons?
-Matthew
> From: "Mohamed A. Gombolaty" [EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial
Hello!
I'm quite unsure, if i'm right here with this question...
I've a customer with a IVR-based PrePaidSystem (DTMF-control, MySQL)
which he wants to port from dialogic/envox on ISDN to a SIP solution. I
think this should be solvable by an asterisk-solution - but i have far
too low
i ve tried to find a gnophone dependency libgtksuperwin.so i searched
every where in google in wiki pages but i didnt found it at all ,if any one
can help in finding it i ll be thankful,and thanks in advance.
_
Dont just
I have three video phones here for testing:
Extension 6003 is Eyebeam
Extension 6004 is a hard phone (model 8770)
Extension 6005 is a hard phone (model 8882)
Can anybody have a look at my settings and the output I get from all
kinds of dialings, please.
The sip settings for all phones is
Hi,
try videosupport=yes in the general section of sip.conf. Maybe it can work.
Giorgio.
Ronald_Wiplinger wrote:
I have three video phones here for testing:
Extension 6003 is Eyebeam
Extension 6004 is a hard phone (model 8770)
Extension 6005 is a hard phone (model 8882)
Can anybody have a
Giorgio Incantalupo wrote:
Hi,
try videosupport=yes in the general section of sip.conf. Maybe it can
work.
I have already set that. Without that NO video at all at any try.
bye
Ronald
Giorgio.
Ronald_Wiplinger wrote:
I have three video phones here for testing:
Extension 6003 is
Thanks mate - I had my voicemail context set up wrong
cheers - works a treat for me too! ;-)
Mark
On 7/11/05, Peter Bowyer [EMAIL PROTECTED] wrote:
On 10/07/05, Mark Edwards [EMAIL PROTECTED] wrote:
anyone managed to get MWI going on the GXP-2000 with * CVS-HEAD? I have set up the mailbox
On Monday 11 Jul 2005 05:02, Michael Stearne wrote:
On 7/10/05, Jim Archer [EMAIL PROTECTED] wrote:
Thanks William and John, I'll look again for that download. Comments
below...
--On Sunday, July 10, 2005 1:50 PM +0200 Wilson Pickett
[EMAIL PROTECTED] wrote:
FWIW? I bought that
Hello all,
I'm having trouble getting variables to work the way I want them to, let me
begin with a simple explanation of the problem, I'm using something like this
in my extensions.conf:
[default]
exten = 5000,1,SetVar([EMAIL PROTECTED])
exten = 5000,2,Goto(mailexten,s,1)
exten =
Hi,
-Original Message-
disallow=all
allow=ulaw
allow=alaw
allow=h261
allow=h263
allow=h263p
Have you tried permutations of this ? I have had working setups with
everything except h263p. My experience with leadtek phones is they tend to
crash when they are talking to any
Dear asterisk-experts,
i've got a problem with my Dialplan.
The task is to get the SIP-address of the called internal phone, that
answered as first.
In this example, two phones are ringing:
DIAL(SIP/1SIP/2|120|m)
But I want to trigger an event, if someone picks up
Hi,
It seems that you are using different audio codec (Unknown RTP codec
96 received)
Try to use standard audio code. Sometimes telephone use codec with bad
rtp code inside. I use alw or ulaw for my test.
Marino
On 7/11/05, Ronald_Wiplinger [EMAIL PROTECTED] wrote:
Giorgio Incantalupo wrote:
I have 2 Sipuras with the following configuration:
The first:
SAS Enable: no, NAT Mapping Enable: No, Sip Port 5060, USER ID 1002, Auth
ID: 1002, Preferred Codec G711a, Prefered Codec Only: no, DTMF Tx Method,
INFO, Enable IP Dialing: no.
The second the same with USER ID : 1003
* Name
Hi,I'm facing an issue with my * boxes. Some
calls are dropped while bridging after a transfer.I have 2 *
boxes, let's call them 'dell' and 'amd', they are connected via IAX2.'dell'
has extensions that matches 63XX while 'amd' matches 62XX.Here's
an example where call will be dropped.
Step
On Mon, 11 Jul 2005, Frank Schoep wrote:
Hello all,
I'm having trouble getting variables to work the way I want them to, let me
begin with a simple explanation of the problem, I'm using something like this
in my extensions.conf:
[default]
exten = 5000,1,SetVar([EMAIL PROTECTED])
exten
I found the problem was with eyeBeam when I had more than one video codec
enabled. Try on eyebeam to only have h263p enabled.
Does the video appear in the Echo test?
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ronald_Wiplinger
Sent: Monday,
Hi.
I would like to force RTP traffic for SIP to go through PBX. Is it
possible to somehow force it in configuration? Is there also possible
for IAX?
Regards
Marcin Okraszewski
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I believe you can find it here:
http://bugs.digium.com/bug_view_page.php?bug_id=759
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kamran Ahmad
Sent: Sunday, July 10, 2005 10:03 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] how
For SIP in sip.conf use the option
canreinvite=no
for a specifed device.
For IAX2 I think you have to use
notransfer=yes
Ivan
I would like to force RTP traffic for SIP to go through PBX. Is it
possible to somehow force it in configuration? Is there also possible
for IAX?
hi,
i need instructions on how to configure asterisk as
a media server.
i need your help.
thanks in advance.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
Hi,
I got having problem in my asterisk when i call i always see this and
degrade the voice quality of the call.how can i resolve thisplease
help
Jul 11 19:37:39 WARNING[74771]: samples/codec_g729.c:217
g729tolin_framein: Received a G.729 frame that was 2 bytes from
RTP
First, thanks to Kevin for the quick response to the 'minor' problem
that zaptel had with 2.6.13 kernels.
Interestingly in the kernel config you can now change the timer
frequency. According to the help messages
100 HZ is a typical choice for servers, SMP and NUMA systems
with lots of processors
On Mon, 2005-07-11 at 09:19 +0100, Bob Goddard wrote:
Compared to Rhetorical (http://www.rhetorical.com/cgi-bin/demo.cgi),
the ATT system sounds awful.
Yes it does sound considerably better, but what do I know I have a
hearing loss. Anyway, have you managed to integrate this with asterisk
Compared to Rhetorical (http://www.rhetorical.com/cgi-bin/demo.cgi),
the ATT system sounds awful.
You're 100% correct!
My mistake, I was thinking of rhetorical when I said ATT. I'm not
familiar with ATT at all - my bad!
Thanks for correcting this and reminding me of rhetorical.
I am having the same thing on my extensions.conf and it works fine. I am
using Asterisk 1.0.7
On Mon, 11 Jul 2005 12:04:59 +0200 (CEST), Armin Schindler wrote
On Mon, 11 Jul 2005, Frank Schoep wrote:
Hello all,
I'm having trouble getting variables to work the way I want them to, let
me
Yes I have faced with the same problem, try to upgrade your eyebeam,
some old versions have problem.
Regards.
On 7/11/05, Storm D. J. Petersen [EMAIL PROTECTED] wrote:
I found the problem was with eyeBeam when I had more than one video codec
enabled. Try on eyebeam to only have h263p
On Thu, 2005-06-30 at 23:34 -0700, snacktime wrote:
The manager action MailboxCount gives the number of old and new
messages in a mailbox. You would have to call the manager via an agi
but it would give you the info you want.
The count is given as an argument to the voicemailnotify program. I
Try over on the Asterisk-biz forum
Robert Schulz wrote:
Hello!
I'm quite unsure, if i'm right here with this question...
I've a customer with a IVR-based PrePaidSystem (DTMF-control, MySQL)
which he wants to port from dialogic/envox on ISDN to a SIP solution. I
think this should be solvable
Easy!!
Add another line to your musiconhold.conf file like this
stream = /var/lib/asterisk/stream,http://sourcepfstream.com:8001/
Then add an externsion number to extensions.conf that uses the stream
variable to play the hold music.
There's quite a bit about this in the wiki.
Mark
[EMAIL
Hello Guys,
I'm somewhat of a newbie and am desperately seeking for some help...
I've managed to get asterisk up and running on my server, and signed up with a
broadvoice account...
I'm having no problem dialing and communicating between extensions, but whenever
anyone tries to call my
Dave Cotton wrote:
First, thanks to Kevin for the quick response to the 'minor' problem
that zaptel had with 2.6.13 kernels.
You are welcome :-)
100 HZ is a typical choice for servers, SMP and NUMA systems
with lots of processors that may show reduced performance if
too many timer interrupts
Try prepending two _'s like this.
exten = 5000,1,SetVar([EMAIL PROTECTED])
exten = 5000,2,Goto(mailexten,s,1
It allows the variable to be exported.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Frank Schoep
Sent: Monday, July 11, 2005 4:40 AM
[EMAIL PROTECTED] wrote:
On 7/10/2005, trixter wrote:
I am considering using iax.cc (sixtel) and wondering if anyone had
opinions, good or bad. Are there outages with any regularity? How
responsive are tech support? How is packet loss? I am particularly
interested in termination to the
Mohamed A. Gombolaty wrote:
Dear Matt,
Yes indeed I did I have used cvs to download asterisk and it's addon
from CVS.
If you followed these instructions then it should be working:
cd /usr/src/asterisk
make; make install
cd /usr/src/asterisk-addons
make; make install
cp
apenon apenon wrote:
Yes I have faced with the same problem, try to upgrade your eyebeam,
some old versions have problem.
How to make the echo test?
bye
Ronald Wiplinger
Regards.
On 7/11/05, Storm D. J. Petersen [EMAIL PROTECTED] wrote:
I found the problem was with eyeBeam when
Benjamin Lawetz wrote:
I have a couple of bugs I'm trying to debug compiling asterisk with
valgrind. But of course when compiled like that the bugs don't occur.
What are the exact effects of Valgrind? Would there be a hit on performance
running asterisk compiled with valgrind ?
'make
On Mon, 2005-07-11 at 08:31 -0500, Kevin P. Fleming wrote:
100 HZ is a typical choice for servers, SMP and NUMA systems
with lots of processors that may show reduced performance if
too many timer interrupts are occurring.
It's always bugged me that my servers have to run with 1000Hz
On Mon, 2005-07-11 at 09:40 -0400, David Mallwitz wrote:
This pretty much sums it up for me as well. Except that it took two
months for my DID to become active. On the other hand, I've had zero
downtime and my 800 number was active within a day. I'm not noticing any
problems with call quality
Hello to all
I succefully installed Asterisk and an Eicon Diva Server 4 BRI (with
CAPI) to connect to a Siemens PBX, but I still cant forward calls to the
Siemens PBX (neither receive them from the PBX).
Here s the result in the asterisk console when I try to dial the 116 PBX
phone:
--
Hello!
I'm integrating an Asterisk-based voicemail system with an old switch, and
I want the call history from SMDI. My understanding is that the terminal
number in the SMDI message matches the channel's trunk number.
From within an Asterisk app, how do I get the trunk number?
Thanks!
Nate
Is your server behind a NAT? If so, make sure that you have configured
/etc/asterisk/sip_nat.conf with your proper settings (change the
localnet and externip settings to match your setup):
nat=yes
externip=xxx.xxx.xxx.xxx
localnet=10.0.0.0/255.255.255.0
sip_nat.conf may only affect your
On Thursday 23 June 2005 2:57am, Patrick Lidstone wrote:
I have a second-hand 7960 which I am attempting to upgrade to use a SIP
image.
The phone currently has a firmware release which doesn't seem to be listed
in Cisco docs - P003AM30. On reboot, it finds the tftp server and requests
the
On Monday 11 July 2005 14:33, jurczak wrote:
I am having the same thing on my extensions.conf and it works fine. I am
using Asterisk 1.0.7
Is it possible that using queues causes problems with regard to handling
variables? It seems that variable handling between contexts is broken after
an
Vikrant Mathur is the lead developer for the open source OSP Toolkit
available on SIPfoundry. Mr. Mathur began his career in
telecommunications as a software engineer at Hughes Software Systems
where he focused on softswitch development. After completing his
Masters degree in Electrical
On Mon, 11 Jul 2005, Joao Pereira wrote:
Hello to all
I succefully installed Asterisk and an Eicon Diva Server 4 BRI (with CAPI) to
connect to a Siemens PBX, but I still cant forward calls to the Siemens PBX
(neither receive them from the PBX).
Here s the result in the asterisk console when I
OK, here's the setup, AAH 0.8, Grandview 2000 phone, Digium TDM04B interface to
POTS lines. Everything seems to be working just fine, but I have some
questions on how to access voicemail options. I can leave a message for an
extension, but when I try to retrieve it by using *97 it asks for
Vikrant Mathur is the lead developer for the open source OSP Toolkit
available on SIPfoundry. Mr. Mathur began his career in
telecommunications as a software engineer at Hughes Software Systems
where he focused on softswitch development. After completing his
Masters degree in Electrical
On Sun, 2005-07-10 at 18:43 +0200, Michiel van Baak wrote:
[snip]
This won't answer your question, sorry.
How are you sending SMS ?
I'm in NL too, and can't seem to find a way to send SMS with
asterisk. The only way I found was some service on the
internet that sells SMS credits for asterisk
Try www.SIPphone.com or www.terracall.com
Seshu
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ellafi
FituriSent: Tuesday, July 05, 2005 2:07 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
[Asterisk-Users] SIP PHONE
Hi All,
I just got
Hello friends,
i make a call through queue to the agent
when agent lifts the call it gives one side voice and
i get this message in the debug
chan_sip.c:1880 sip_write: Asked to transmit frame
type 64, while native formats is 4 (read/write = 4/4)
in my sip.conf iam allowing only ulaw
can
Hi,
i have trouble to dial out over my sip-provider gmx.
I can register with my provider over port 5060 and also dial out.
It rings at the remote phone but when the call is answered there is no
sound / voice to hear.
This is the part from my sip.conf and extensions.conf:
register =
Thanks for all answers.
I begin telling you that I'm new to the asterisk world, so I started with
[EMAIL PROTECTED] because of its easy installation and setup.
I'm doing my tests at home. I have a local network 192.168.1.0 class C
(255.255.255.0), my asterisk box has IP 192.168.1.42 and my
On Wed, 2005-07-06 at 16:27 -0300, Angel Diaz wrote:
Hi,
I have to connect 30 phone lines to my asterisk server, can somebody
help on how I have to do it ?
I have a TDM405P and one TDM400P with 4 FXO ports.
Do I have to use 8 TDM400P ? Or, is there another way to do it ?
Thanks,
Hello
I am planning to build a small PBX using
TDM22B.
We have a Siemens Hipath 3750 in operation
already.
When I manage to complete my PBX using TDM22B
I would ofcourse like to be able to connect my Asterisks PBX
with the Siemens Hipath 3750 PBX.
Will there be any issues regarding my
Marc Fishman ha scritto:
the firmware image listed in OX79XX.txt correctly, displaying Upgrading
Software on the screen. It then continues to re-request the same image from
the tftp server at 10s intervals indefinitely. What am I doing wrong?
You need to upgrade to a older version first.
I'm having some trouble dialing phone numbers in Argentina with Digium Zaptel
cards. Does anyone have some sample configuration that works with Digium
TDM04B cards in Argentina? I'm mainly referring to the /etc/zaptel.conf
and /etc/asterisk/zapata.conf.
I have two Zaptel cards: the first one
We experienced the same problem on a Dell 2850 server. Our other asterisk
admin went a different route and inquired with Dell. They told him this was
completely normal and not to worry about it. I'm still skeptical.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Title: FW: [Asterisk-Users] Retrieving dtmf, passing to shell and getting the result
John thanks for the help. When I change my plan to this and then dial 2 it gives me a busy signal. When troubleshooting I added an exten = 2,1 Ringing (just as a check) it rang and went straight to busy. On
Hi,
what is the status of chan_cornet? Does someone here use it in
production? I can't find enough info about it. Some URLs will be great.
Thank you,
-David
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hello!
I tried to compile chan_sccp-20050705 but I receive the following
errors:
linux:/home/share/chan_sccp-20050705 # make install
sh ./create_config.sh /usr/include
Checking Asterisk version...
* no 'struct ast_channel_tech', using old pvt
* no 'struct ast_callerid'
* no 'AST_CONTROL_HOLD'
Hi,
I'm hoping someone can point me in the right direction to fix this issue..
I just recently have a need to have a group of people (5 to be exact) talk via
a conference call on a semi-regular basis.
The phone lines that are connected to a conference (meetme) are as such:
1. Local SIP
Holger Hornung ha scritto:
Hello!
I tried to compile chan_sccp-20050705 but I receive the following
errors:
What is the problem?
rm /usr/include/asterisk/*
rm /usr/lib/asterisk/modules/*
cd asterisk
make clean
make upgrade
cd chan_sccp-20050705
make clean
make install
asterisk -vvvcg
Hello,
I think there maybe an issue with my refer transfers. See below or attached:
No. TimeSourceDestination Protocol Info
1 0.00192.168.1.2 192.168.1.5 SIP/SDP
Request: INVITE sip:[EMAIL PROTECTED], with session
Cual es el problema en Argentina? La diferencia deberia ser la señalización
unicamente. El resto no cambia. Nosotros usamos TP410 sin problemas pero con
DS1 no E1.
Saludos,
Carlos Alperin
Senior System Engineer
Seneca Communications, LLC
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL
I'm rolling out an installation with snom 360s in the near future.
Simple SOHO configuration, 3 FXOs hanging off a TDM400B, 4 snom 360s, a
snom 200, some variant of IAX softphone, and an IAXy or Sipura 2002. I
have the 360's set up to subscribe and notify for the line use lights,
which works
Hello.
How can I check if the RTP traffic between two channels is bypassed?
Some * console command?
Thanks.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
What happens if you press *98 and enter the extension and password? are you
using speakerphone? tried it with the handset only?
- Original Message -
From: [EMAIL PROTECTED]
To: Asterisk-Users@lists.digium.com
Sent: Monday, July 11, 2005 7:51 AM
Subject: [Asterisk-Users] Asterisk @
Ive got a project where I need to sell a voip QOS
product from Australia
to US resellers.
I dont suppose anyone here knows where I can find a
list of a whole heap of US
resellers do you in either VOIP or IP space?
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Adam,
I've tried both the [heath] heading and the [31521] heading.
I figure the 31521 was right because the registration error message
says [EMAIL PROTECTED]
I've tried host = dynamic and defaultip = 172.x
No combination of those above settings scores me a successful
Peter Nixon will be making the trip to Chicago to speak at Cluecon,
he'll be speaking on the topic of Real world deployment of Open
Source. Peter has done tremendous amounts of work on the
FreeRadius project. In addition if you're wanting to get sponsorship
in this is the week to do so,
Peter Nixon will be making the trip to Chicago to speak at Cluecon,
he'll be speaking on the topic of Real world deployment of Open
Source. Peter has done tremendous amounts of work on the
FreeRadius project. In addition if you're wanting to get sponsorship
in this is the week to do so,
Pepe Aracil wrote:
Hello.
How can I check if the RTP traffic between two channels is bypassed?
Some * console command?
You can't. show channels and sip show channels will only show you
the SIGNALING (which always passes thru Asterisk). You will need to use
tcpdump or etherreal or
Ronald Wiplinger wrote:
apenon apenon wrote:
Yes I have faced with the same problem, try to upgrade your eyebeam,
some old versions have problem.
How to make the echo test?
Just add a line to your extensions.conf:
exten = 600,1,Echo()
And that should do it.
Also try the hardphones
It's a little odd. Something like asterisk -v4 seems more
appropriate. You can also use set verbose level so that you don't
have to restart your console session to change the verbosity. I really
don't know what the maximum effective verbose level is.
MARK.
George Garvey wrote:
On Sun,
Is asterisk able to forward it's ALERT_INFO data to another asterisk server
?
My situation should look like the following:
Call comes into asterisk1 in SIP. Asterisk1 sets the ALERT_INFO=Bellcore-r2,
Asterisk1 dials Asterisk2 (SIP), Asterisk2 dials our SIP device which should
ring with the
I'm not sure if this is a -users or a -dev question, since the answer
probably comes down to something in the code.
I'm running the latest CVS-STABLE, and am subscribed to PSTN service
using IAX2 via Voiptalk in the UK.
I've just been alerted by a customer that the sending of DTMF from my
Before to anything else, are you sending DTMF in-ound or out-bound?
Most of the time when DTMF is not sent is because is in-bound. Just choose
out-bound or RFC2833 (I don't remember if this is the right standard).
Carlos
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I've been experimenting with the zaptel TDMoE stuff and I've got it all
working. Calls go from one asterisk box to the other, with no issues,
except they don't bring the callerID along with them. I tried the em
signalling from the wiki and I thought maybe that had something to do with
it, so I
I don't suppose anyone here knows where I can find a list of a whole heap
of US resellers do you in either VOIP or IP space?
This might help:
http://www.voip-info.org/tiki-index.php?page=Asterisk+system+vendors
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F:
You need to upgrade to a older version first.
version 5 or 6 before upgrading it to version 7.
I appreciate the response but that's what isn't working. I have tried v5.3
and v3.0 with the same result. I suspect the firmware version (P003AM30) is
the problem as I haven't run across any Cisco
Carlos Alperin [EMAIL PROTECTED] wrote:
Before to anything else, are you sending DTMF in-ound or out-bound?
IAX always sends DTMF out-of-band, not inband.
Most of the time when DTMF is not sent is because is in-bound. Just choose
out-bound or RFC2833 (I don't remember if this is the right
Well, it looks like there is no way for Asterisk to
read the MWI fsk tones from the PSTN at this time.
Sigh... Anyways, I am going to get a couple outboard
boxes with MWI indicators on them, and see if my wife
can deal with that to tell if a message is waiting
instead of the MWI on the handset.
Perhaps we need to go over this again.
IAX2 CANNOT DO INBAND DTMF. IAX2 DOES NOT USE RTP. IAX2 DOES NOT DO
RFC2833.
Carlos Alperin wrote:
Before to anything else, are you sending DTMF in-ound or out-bound?
Most of the time when DTMF is not sent is because is in-bound. Just choose
I don't notice it on my TDMoE that is configured as PRI either. Looks like
you need to post a bug to the tracker.
MATT---
-Original Message-
From: Weezey [mailto:[EMAIL PROTECTED]
Sent: Monday, July 11, 2005 4:33 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] TDMoE and
Weezey wrote:
So, is there a trick to it or does callerID information just not go across
TDMoE?
Use PRI signaling on the TDMoE span, not quasi-analog signaling.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
MF Hulber wrote:
It's a little odd. Something like asterisk -v4 seems more
appropriate. You can also use set verbose level so that you don't
have to restart your console session to change the verbosity. I
really don't know what the maximum effective verbose level is.
MARK.
255
JN
Answering my own question here for anyone else fighting with this. From a
Cisco Field Notice (see http://www.cisco.com/warp/public/770/fn18246.shtml).
The problem appears to be that a 7960/7940 running P003AM30, the load shipped
from the factory, cannot load a new load file that is more than
Hi Folks,
I've Asterisk Bristuffed up and running behind an Auerswald Commander
Basic ISDN PBX on the internal ISDN Bus (BRI/PTMP). The HFC Card works
marvelleous for outgoing calls (as the parallely installed avm fritzcard
with chan_capi does), but when I'm trying to call in, I get a short
Brian West wrote:
Peter Nixon will be making the trip to Chicago to speak at Cluecon,
he'll be speaking on the topic of Real world deployment of Open
Is there an echo in here?
Steve
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Anyone know how I can push a firmware update to a Snom 190 without using
DHCP? In the web interface, I specify a path to the Snom firmware, and it
works, except I have to physically press OK to get the update to download. I
need to do it remotely...
___
Hi All,
I just purchaced a Cisco uBR924 and was under the
assumption that it did SIP.
Being somewhat new to Asterisk, is there anyone
willing to supply a working config that will get me started on configuring these
items.
Best Regards
___
Hi guys,
Can somebody help me on some questions please ?
If I have a VoIP network with my Asterisk platform in Europe, what do I
need to interconnect my VoIP network to another network in the USA in order
to my customers in Europe be able to call to customers in the USA network ?
The
On Monday 11 July 2005 16:51, Angel Diaz wrote:
Hi guys,
Can somebody help me on some questions please ?
If I have a VoIP network with my Asterisk platform in Europe, what do I
need to interconnect my VoIP network to another network in the USA in order
to my customers in Europe be able
Hi List,
I have slapped together a no-frills yet functional prepaid framework for
Asterisk. It supports concurrent calls and has been built with
robustness, simplicity and billing accuracy in mind.
You can find some docs and the code on the following page:
http://ykoz.net/intl/grobill/
My job is to combine video phones of SIP and h323 on a * box.
Which H323 and how to setup?
bye
Ronald Wiplinger
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE
Hello,
I'm trying to find out if Asterisk will support plain G729 or G729b.
I've read all over that it supports G729, but I can't seem to find any
explicit remarks regarding the specific versions of the codec Asterisk
will support. I noticed that Digium allows Asterisk users to register
and
Hi Folks;
I just bought a Polycom SoundPoint 500 off of ebay after having spent
way too much time trying to get updated sip images for our cisco phones.
The phone I bought didn't have an AC power adapter; Could someone
please tell me the volts amps that the dc plug that comes with the
Mine says 12VDC @ 400ma , tip +
Tim
Michael Jones wrote:
Hi Folks;
I just bought a Polycom SoundPoint 500 off of ebay after having spent
way too much time trying to get updated sip images for our cisco phones.
The phone I bought didn't have an AC power adapter; Could someone
please
1 - 100 of 138 matches
Mail list logo
