dbruce wrote:
The solution to the IRQ issue is IO-APIC.
Recompile your kernel with IO-APIC support (most new motherboards support
IO-APIC)... If your board supports it, It WILL solve all your IRQ issues...
If your board does not support it then it will make no difference.
I've never heard
Karl S. Katzke wrote:
Next time, I'd actually buy a Dell SC420 or something similar. :-P
Honestly, I'm really scared of Dell hardware. Some of the most messed up
computers I've worked on have come from Dell. I hear their servers are
better than their laptops and desktops, but having only
snom supports GSM.
CS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Rich Adamson
Sent: Thursday, August 18, 2005 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] codec gsm and cisco
I
Im using a TE110P as a trunk to a Panasonic KD-500 everything works
well.but Im having this problem where one of the channels becomes
blocked with a partial phone number after about two days. So if the
channel that becomes blocked is channels 23 no calls can get in. If
the
This smells of the You never get fired for buying IBM (replace
with Cisco, etc) quote. Sure you can take the small gamble that a tier
1 platform will meet your needs, however, nothing beats a full battery of
tests including burn in, capacity and failure mode tests. If you don't
know
Hi;
Do you know where the addmailbox script goes? Now in CVS, it shows
revision 1.3
date: 2005/08/02 11:57:06; author: markster; state: dead; lines: +0 -0
This script is now useless...
So any new tool to create voicemail folder?
Thanks
Kun
For canreinvite=yes
to work, I think I need to remove the t argument in the Dial(SIP/ext|60|t)
application. Otherwise, Asterisk will allways stay in the middle. I don't want
that, so I removed the 't' argument. That works. Now, when two UA are calling,
Asterisk gets out of the RTP stream.
jennyw wrote:
I've never heard about IO-APIC before, so I just did a Google search.
The articles I found say that it's an Intel thing, and, since I have an
AMD processor w/ ASUS motherboard, it's unlikely it'll work, right?
Even so, it sounds interesting. But does it apply to
Asterisk cvs-head (up to date) keeps core dumping on me. I finally
tracked it down to my register command for Vonage in the sip.conf file. If I
remove the username and password from the register command, it won't core
dump, but of course won't register either... This is odd. Any
Tim Connolly wrote:
Asterisk cvs-head (up to date) keeps core dumping on me. I finally
tracked it down to my register command for Vonage in the sip.conf file. If I
remove the username and password from the register command, it won't core
dump, but of course won't register either... This
I am using TE110P E1 Card and using the wcte11xp
driver.
Here is my ztcfg -vvv result: -
Zaptel Configuration
==
SPAN 1: CAS/HDB3 Build-out: 0 db (CSU)/0-133 feet
(DSX-1)
Channel map:
Channel 01: Individual Clear channel (Default)
(Slaves: 01)
Channel 02: Individual Clear
On 8/22/05, Massimo De Nadal [EMAIL PROTECTED] wrote:
Forget RPM.
I agree.
First of all read:
http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+SuSE
we did
then download
http://www.junghanns.net/downloads/bristuff-0.2.0-RC8n.tar.gz
explode the tarball, read the file INSTALL
On Tue, 23 Aug 2005, Voicomm User wrote:
Hello,
We have 4 'Onramp-2' Telstra ISDN BRI services operating on Asterisk
Server with Eicon 4BRI card. For most part the service is okay.
However, we are are having problems with passing callerID to internal
extensions.
This is the set of
hi, this is the q931 error we got, once the
setup msg sent, e got a release complete msg back saying swithc is
Circuit/channel congestion (34), any onegot this before,
thanks in advance.
Protocol Discriminator: Q.931 (8)
len=47 Call Ref: len= 2 (reference 35/0x23) (Originator) Message
hello
i m using asterisk-1.0.9. i want to connect to db
through odbc. isql is working. but asterisk is not
getting user information from this table. can any one
pls check this
/etc/asterisk/extconfig.conf
[settings]
sipusers = odbc,mysql1,sip_buddies
sippeers = odbc,mysql1,sip_buddies
sip.conf
Title: Faxing help
Hi, I have still had no luck with faxing and am getting a couple of pages of the following debug message
Changed from phase 1 to 4
DIS:
Prefer 256 octet blocks
Can receive fax
Supported data signalling rates: V.27ter and V.29
R8x7.7lines/mm and/or
Dear Asterisk experts and users.
Let's assume I purchased the call number, let it be 2200
for simplicity. So when we call 2200, our call will be
processed with Asterisk, and come in, say, [incoming] context.
Well, then I'd like Asterisk to process all calls that a
*begin* with 2200, i.e.
Hello Lars,
Have you got kernel sources installed ?
I think that are mandatory for Zaphfc.
Regards
Tuesday, August 23, 2005, 10:37:37 AM, you wrote:
LD On 8/22/05, Massimo De Nadal [EMAIL PROTECTED] wrote:
Forget RPM.
LD I agree.
First of all read:
Hi,
I use Grandstream HT486 with Asterisk. I dial 9 to get an FXO line, then
hangup and wait 1-2 secs. Then, I dial immediately 9 again to get another
outbound channel and hangup again. Guess what happens, the two outbound
lines are connected indefinetly.
The cause of this issue is that I
Mike Hansford wrote:
If Asterisk is not able to function as a SIP proxy, how do I re-write
and/or route messages? Can Asterisk fake these processes or will I
require a proxy like SER to do it for me?
Please read
http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20not-proxy
/Olle
Title: Message
I have
no idea about that error message sorry.
But I
recently installed 1.0.9 on two FC4 boxes (one was a brand new Shuttle which
impressed me cos I was sure I was going to have driver
issues).
On the
first I installed Asterisk without doing any updates first and the second
My context in zapata.conf is default
But still cannot dial via Zaptel (TE110P)
--- Craig Guy [EMAIL PROTECTED] wrote:
Hi rootlinux,
I'm in Australia where we also uses crc4 on the span
line, could you also
show the relevant section of your zapata.conf?
Looking at your
Hello All,
I have this strange problem,
I can dial out with my sip phone and it seems to work relatively well, but
when I call in, the line just rings and rings, I get no indication in
asterisk that it's detecting an incoming call.
The strange thing is that in ztmonitor 1 -vv the rx volume goes
The IO-APIC may originally have been a feature introduced on jigh end
servers running dual processor, but in the past several years it has become
a more universally supported feature. The IO-APIC/APIC is part of the
chipset, not of the processor, so should be processor type nuetral.
Doing a qucik
David Sampson wrote:
Hello
My single line extension
users (connected via channel banks)
need to be able to hang up faster. If they just flash the hook it
doesnt
disconnect right away. Any ideas on how to resolve this?
Thanks,
Dave
In zapata.conf put this
First off... go through your zapata.conf and zaptel.conf files and actually
set your configuration for your specific hardware and desired results.
The obvious is that you are using E1 signalling (set in zaptel.conf), but in
zapata.conf you specify switchtype=national. This probably not work
___
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Sorry, sent with wrong accountread
below
From: Sherwood McGowan
[mailto:[EMAIL PROTECTED] Sent: Tuesday, August 23, 2005 8:34
AMTo: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: SIP DEADLOCK
Anyone using a
CVS-HEAD pulled later than 8/13? We're runnign a
show application dial will show you the options supported by your
version of Asterisk.
Joseph wrote:
On Mon, 2005-08-22 at 14:32 -0400, William Suffill wrote:
I'd suggest Dial(trunk/1800555,30,D(1wwww2)
I'm using *-1.0.8 and reading the documentation Option D: is in CVS
since
Wei Kun wrote:
Hi;
Do you know where the addmailbox script goes? Now in CVS, it shows
revision 1.3
date: 2005/08/02 11:57:06; author: markster; state: dead; lines: +0 -0
This script is now useless...
So any new tool to create voicemail folder?
They are now
O come on now! Nothing? Not even a No idea! Good Luck! or anything?
Weak :)
Just kidding. Thanks just the same.
- Don
[EMAIL PROTECTED] 8/22/2005 11:09 AM
Hello,
I am having an issue with hangups being handled within Asterisk. Right now,
when an inbound call hits
the Asterisk
On Monday 22 August 2005 17:07, jennyw wrote:
This is for an office -- I figured that running hardware RAID would be
the most likely to avoid downtime if a hard drive failed. How do most
people handle this?
Linux software RAID1. Don't piss away your money on anything else. Hell on a
Hi All,
I have installed chan_unicall and MFC/R2 successfully, and is runnign fine.
But I noticed that once unicall is installed, asterisk CPU usage as reported by
'top', jumps to 99% every few seconds.
I have no incoming calls, and I have even removed the E1 lines from card and I
tried almost
Does anybody have an example follow-me configuration web page code
written in either php or perl that can write out the follow-me config
into the asterisk files?
I'd like to setup something on our office voip server that I can
change as needed via a web page rather than writing the script by
I have asterisk connected to a PSTN line with FXO interface cards from Digium.
When I first start Asterisk it sounds fine after a few days of
running I begin to hear alot of echo on calls... restart asterisk and
it is fine again.
Any thoughts what this might be?
Eric Wieling aka ManxPower wrote:
John Novack wrote:
Or, in the example below, wait before dialing?
exten = s,1,Dial(ZAP/g1/${ARG1},360) ; ARG1 is the number to be dialed
If you are using analog ports, yes. Dial(Zap/g1/ww15551212).
exten = s,1,Dial(ZAP/g1/ww${ARG1},360) should work
John Novack wrote:
Eric Wieling aka ManxPower wrote:
John Novack wrote:
Or, in the example below, wait before dialing?
exten = s,1,Dial(ZAP/g1/${ARG1},360) ; ARG1 is the number to be dialed
If you are using analog ports, yes. Dial(Zap/g1/ww15551212).
exten =
The main regret I have about hardware RAID is that the card is sharing
an IRQ with one of the Digium cards. This whole IRQ thing is driving me
crazy ... I disabled everything I could in BIOS and that freed up some
IRQs, but there's no way to assign a particular IRQ to a particular
device. I
I would like to do the same thing, and the easiest would be to use MySql
and a web connector :..
I can help.
Ben
Does anybody have an example follow-me configuration web page code
written in either php or perl that can write out the follow-me config
into the asterisk files?
I'd like to
I have a proposition for US based CLEC(s) and would like to speak with
any that read this list offline. In short I am looking for US DIDs for
high volume traffic. If there are any CLECs out there, please contact
me offline via email.
Thank you,
Bret McDanel
[EMAIL PROTECTED]
--
Trixter
Hi!
First I have to say, that I'm not very familiar with CVS and patching.
I tried to patch compile CVS-HEAD.
First I checked out zaptel, libpri and asterisk with this command: cvs
co zaptel libpri asterisk
But the latest patch sipsubscribe-20050812.rev806v2.txt from
BUG/SYMPTOMS:
1.Under certain circumstances, octoBRI (and most likely quadBRI) ISDN
cards (Junghanns/CologneChip) severely distort certain ISDN payload.
2.Although these claims relate to the bristuff patch, the problem might
not be limited to bristuff and in fact be rather
Make sure you have your Alcatel's analogue port setup for fxo so that
you can plug the X100P's fxs port (line) into it.
As for the ringing, that's artificially gernerated by the Alcatel.
You do have the Zaptel drivers compiled and loaded right? What do you
get when you do zap show channels?
You and a whole bunch of other guys, including me:
http://www.google.ca/search?hl=enq=tdm+restart+site%3Alists.digium.commeta
=
Cron a reboot every morning at 4 in the morning. There's a script floating
around also to shutdown Asterisk when convenient, unload the driver reload
it restart
Hi,
This is what I want to do:
1. Asterisk to answer calls via DID's, currently using SIPGATE
2. Provide a menu, and allow users to dial out.
3. According to the country and area they dial, the call should connect via
one of up 4 carriers depending on cost.
4. If the carrier is busy it should
Why don't you post YOUR config files, then you might get some replies as
to what is wrong.
What you are trying to do can be done.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Huw Morgan
Sent: Tuesday, August 23, 2005 8:33 AM
To:
Ronald Voermans wrote:
For canreinvite=yes to work, I think I need to remove the t argument in
the Dial(SIP/ext|60|t) application. Otherwise, Asterisk will allways
stay in the middle. I don't want that, so I removed the 't' argument.
That works. Now, when two UA are calling, Asterisk gets out
Kamran Ahmad wrote:
hello
i m using asterisk-1.0.9.
Come on people. Pay attention.
What does the very first opening paragraph say:
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime
-Matthew
___
Asterisk-Users mailing list
On Tue, 2005-08-23 at 07:57 -0500, Eric Wieling aka ManxPower wrote:
show application dial will show you the options supported by your
version of Asterisk.
Yes, it shows support for: 'D([digits])' -- Send DTMF digit string
*after* called party has answered but before the bridge. (w=500ms sec
i have 2 iax servers one with analoge line plug into
the TDM card and the second server is do dial from,i
have made the connection between the 2 servers
sucessfuly and when it goes to TDM it dials the number
and the call is done ,but in bad and noisy sound
quality and with an echo. does anyone has
Hello,
with regard to the description of testing asterisk + USB ISDN TA on
OpenSlug:
http://www.nslu2-linux.org/wiki/OpenSlug/Asterisk
and
NSLU2 running Debian
http://peter.korsgaard.com/articles/debian-nslu2.php
I've tested the same thing (asterisk as VoIP/PSTN
Remember in the good ol days when answering machines were smart enough to
know when there was a message on the machine, and it would pick up after 2
rings rather than 4? (that is, if you knew how to turn it on - that
required to know how to set the time on your VCR to avoid the flashing
12:00:00)
Matthew Boehm wrote:
Umm.. DUH! If you remove the RTP stream from asterisk, asterisk
can't send audio (the rtp stream) to the phones.
Umm. DUH! Yes it can.
When a SIP endpoint is placed on hold, Asterisk will re-INVITE the audio
stream back to itself for precisely that reason.
I found many mailing list threads and one wiki webpage with ideas and
questions related to failover and high availability solutions.
Is there any webpage or wiki page that summarizes all these ideas?
What I have found:
- case 1: two identical Asterisk boxes with one acting as hot-failure
Anyone able to get me a comp/highly discounted ticket to this?
$150 just to visit the exhibition halls sounds crazy?
Dean
-Original Message-
From: Jeff Pulver [mailto:[EMAIL PROTECTED]
Sent: Tuesday, 23 August 2005 11:47 AM
To: mailinglist1
Subject: Register Today for Fall 2005
since delete is a reserved word, what do you name a column in your
voicemail options table to allow setting of the delete option for realtime
voicemail? Anyone?
Sherwood McGowan
ViaTalk
Level 2 Support
VOIP System Engineer
___
Asterisk-Users
Chris,
Thank
you for your answer. By the way, my * server won't be a PSTN gateway. The SER is
connected to another SIP gateway provided by our Telco. Would you be so kind to
give me some more details on this:
- Say
I have * server A with extensions 100, 101, 102, and * server B also with
I have a TDM4xx card with two (3 and 4) interfaces for my land lines. I
have a basic setup working with them and one VoIP provider. Questions:
1. How do I determine which Zap line the incoming call is on so I can
handle it differently? One line is my home phone and the other is my
work line.
Kevin P. Fleming wrote:
Matthew Boehm wrote:
Umm.. DUH! If you remove the RTP stream from asterisk, asterisk
can't send audio (the rtp stream) to the phones.
Umm. DUH! Yes it can.
When a SIP endpoint is placed on hold, Asterisk will re-INVITE the audio
stream back to itself for
Hello, I'm trying YAACID ( http://www.shatterit.com/opensource/yaacid/ )
for incomming call notification on PC (and open url with callerid), but
it does not display/pop anything :-(
my config is very simple...
(yaacid is successfully registered as manager in asterisk)
thanks
PJ
* dialplan:
Hi,
i've been discovering app_sms and it states that it can act as an smsc
for landline sms. Receiving SMS from my Gigaset Phones is no problem and
the SMS are stored as files on by * box. So far so good.
Let us assume that i have a couple of phones which should be able to
receive SMS
hello
i m using asterisk-1.0.9. i want to connect to db
through odbc. isql is working. but asterisk is not
getting user information from this table. can any one
pls check this
odbc connection is working properly is there some
thing required
/etc/asterisk/extconfig.conf
[settings]
sipusers =
Hello,
In our company we are using Asterisk-cvs-Head with realtime. I am not
able to figure out a issue wherein the sip channel stays active
indefinitely.This happens when a call is in progress and the person
who called doesn't hangup normally but in between the conversation his
ATA gets
All,
I'm having a heck of a time getting hdlc to work on kernel
2.6.11.9 .. I compiled hdlc, hdlc_gen, hdlc_cisco, hdlc_raw, into the
kernel (note into, and not 'modules').
System comes up, I configured zaptel.conf
span=1,0,0,esf,b8zs
nethdlc=1-24
modprobe wct4xxp
ztcfg
sethdlc hdlc0
I found the problem. The ztdummy wasn't loaded. So it had no timer
there. When the RTP stream was going through asterisk, I think * used
the stream for timing.
Ronald
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Matthew Boehm
Verzonden: dinsdag 23
Hello List,
This is my first message herein. I was playing around with System() and
AGI() and found out something I cound not determine my configuration
error. I added before.agi and after.agi to the agi-bin dir. Tried to
make before.agi get run before the dial call and after.agi be run
Hi,
We are currently running our own equipment to break calls out in a
location I need to balance the calls out between two sites so that one
site doesn't keep getting hit again and again.
So currently we have something like this:
exten = _1.,1,Dial(IAX2/pop1/${EXTEN})
exten =
On Tue, 2005-08-23 at 01:30 -0600, Colin Anderson wrote:
Good luck and please keep posting so everyone can learn from your
experience.
Hmmm... we too moved our telephone system at work to asterisk just a few
days back and since then, I have been following this thread.
A few snippets from my
We are building asterisk clusters using mysql replication. All the
configuration and cdr data is stored using the res_mysql module.
Replication creates identical servers. Then, the phones register to each
server using DNS SRV records. If any server goes down, all the phones
registered to that
hi christian,
Christian Wengel schrieb:
But the latest patch sipsubscribe-20050812.rev806v2.txt from
http://bugs.digium.com/view.php?id=3644 didn't worked,
maybe you like to try the latest patch i created a view hours ago...
sipsubscribe-20050823.rev813.txt on http://bugs.digium.com/view.php
Hi
I've seen some posts on the list regarding integrating Nokia's PTT (nokia 6020
and nokia 6230i) with asterisk
And use * as a PTT server..
So far I was able to have mobile register itself , send an invite to it, and get
SIP error 603 (DECLINED) back from it.
And ofcourse the PTT sign
Hello There,
I'd like to define multiple providers for one dial prefix , like , i
want if my one trunk gateway is filled the call is transfered to other
ip, how can i achieve it with areskicc.Kindly Help.
cheers
Thanks
Junaid Uppal
___
I have had an idea of using two identical servers: Server A with IP
x.x.x.a and server B with IP x.x.x.b. Server A is live while server B
sits in the background monitoring server A. Server B rsync's asterisk
config files daily with server A.
In the event of server A going down, server B
HI,
In Asterisk 1.0.9...
I'd like, as a part of the dialplan, to retrieve and play the voicemail
name.
It would be played to the user before the password prompt.
1. User dials voicemail
2. Test ${CALLERIDNUM} for proper range
3. based on ${CALLERIDNUM} play the voicemail name
4. run
Hi Guys,
I have couple of SIP trunks, everyone of them with different IP address.
Is there a way to create some kind of SIP group trunks, like I can do with
ZAP channels.
With sip I have to do example below, in order to jump to next SIP trunk if
first one is busy.
Is there another, nicer
Matt Schulte wrote:
All,
I'm having a heck of a time getting hdlc to work on kernel
2.6.11.9 .. I compiled hdlc, hdlc_gen, hdlc_cisco, hdlc_raw, into the
kernel (note into, and not 'modules').
System comes up, I configured zaptel.conf
span=1,0,0,esf,b8zs
nethdlc=1-24
modprobe wct4xxp
Hi all,
I replaced a TE410P Rev C 1st Generation Firmware with a TE411P
without any cfg changes (zaptel/zapata).
As a result Asterisk crashes on outbound from PRI4 going to PRI1 calls:
Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a
UA, but i'm in state 1
Aug 23
I don't think this will work but it's worth a try.
Fall VON 2005 http://von.com is happening September 19-22, at the BCEC in Boston.
As usual, we have a special offer for members of the pulvermedia community, which is valid for the month of June only. Register using priority code JUNE and
[EMAIL PROTECTED] wrote:
I have had an idea of using two identical servers: Server A with IP
x.x.x.a and server B with IP x.x.x.b. Server A is live while server B
sits in the background monitoring server A. Server B rsync's asterisk
config files daily with server A.
In the event of server
List,
I purchased 2 g729 licenses but I can't get it to answer a g729 call
from a cisco router with a vwic card. In the debug output below you
will see that asterisk thinks it only supports: (gsm|ulaw|alaw|h263)
when it should support g729 according to the config also listed below.
The real odd
I'm currently working out the config bugs on my * box and I'm noticing
that the meetme is very scratchy. As in not usable scratchy tho I can
hear the audio it sounds like when you talk through a fan.
Anyone have any ideas? Linux 2.6 with RTC installed. Using stable
release and SIP devices.
Recompile zaptel with
- MMX enabled
- Enable the AGGRESSIVE_SUPPRESSOR with MARK2
Excellent suggestion, I had forgotten about that. Note to those that try:
Enabling MMX in Zaptel will bugger up SpanDSP, your faxes won't recieve
correctly. Why? Dunno. Just my experience; although I've only
On Tuesday 23 August 2005 13:25, VaibhaV Sharma wrote:
* 2 SATA Hdds with H/W RAID (RAID mainly because we plan to do a lot of
recording on conference calls + fault tolerance)
What good does RAID give you on writes? None whatsoever. RAID only helps
performance on reading. Fault tolerance
Colin Anderson wrote:
Recompile zaptel with
- MMX enabled
- Enable the AGGRESSIVE_SUPPRESSOR with MARK2
Excellent suggestion, I had forgotten about that. Note to those that try:
Enabling MMX in Zaptel will bugger up SpanDSP, your faxes won't recieve
correctly. Why? Dunno. Just my
Ok, thanks for the info.. What about the other problem(s) I'm having?
Any thoughts?
Matt
-Original Message-
From: Kristian Kielhofner [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 23, 2005 1:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
We're currently working on a SIP load-balancing system using ipvsadm and so
far it seems to work pretty well.
We load balance the port 5060 registration (tracking the connection for a
little more time than the registration retry so that it goes back to the
same server) using tunneling.
So a
What good does RAID give you on writes? None whatsoever. RAID only
helps performance on reading.
Come again? Writing to multiple hard drives in parallel is way faster
than writing the same file to one HDD.
You should Google the words RAID and Write Performance.
I assume you must have meant
I found the script for anyone interested
#!/bin/sh
# This script tell asterisk to stop when there are no active calls,
# waits for it to actually stop, then reloads the wctdm module
# and restarts asterisk.
/usr/sbin/asterisk -rx stop when convenient
while /bin/ps ax | /bin/grep
On Tuesday 23 August 2005 14:37, Colin Anderson wrote:
Excellent suggestion, I had forgotten about that. Note to those that try:
Enabling MMX in Zaptel will bugger up SpanDSP, your faxes won't recieve
correctly. Why? Dunno. Just my experience; although I've only tried it on
two different boxes
Title: Message
All,
wondering if you can
help, I had a perfectly working Mandrake 9.2 box running on a via Mini ITX
5000/classic. Asterisk (zaptel and libpri)was built from CVS head around
22nd July 2005. I decided now was a good time to ghost it upalthough
humorous for you all suffice
Call Digum. They support the license codec install.
Matthew Schumacher wrote:
List,
I purchased 2 g729 licenses but I can't get it to answer a g729 call
from a cisco router with a vwic card. In the debug output below you
will see that asterisk thinks it only supports: (gsm|ulaw|alaw|h263)
Interesting. What version are you running? I may try to update to the
latest.
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 23, 2005 12:56 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Small office setup/using analog lines
That username password combination is referenced elsewhere for
different models of ATA's as well. I believe it is somewhat a Vonage
standard.
On 8/23/05, Steve Gladden [EMAIL PROTECTED] wrote:
There is a fee, but I believe you can call Vonage and get a box
unlocked after you are done with
I'd rather unload and reload but what you are saying then is that
I need to unload the driver and reload the driver?Is that really
neccessary? I've always been able to fix it just restarting
asterisk... so this is a known (unknown) bug?
On 8/23/05, Colin Anderson [EMAIL PROTECTED] wrote:
Michael D Schelin wrote:
Call Digum. They support the license codec install.
Matthew Schumacher wrote:
List,
I purchased 2 g729 licenses but I can't get it to answer a g729 call
from a cisco router with a vwic card. In the debug output below you
will see that asterisk thinks it only
On Tue, 2005-08-23 at 15:16 -0400, Douglas Logan wrote:
That username password combination is referenced elsewhere for
different models of ATA's as well. I believe it is somewhat a Vonage
standard.
one of the things about the vt1000 is that the provider can dynamically
change your pw. That
That did the trick! Make clean allowed me to recompile.
On a related note, is there a way to complete remove asterisk and all
installed files? (automatically)
Thanks,
Bob
-Original Message-
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Monday, August 22, 2005 2:56 PM
To:
SNIP
Ok, I figured it out, * was not using the config under
the [router]
context in the config file. Once I enabled g729 in
[general] it worked.
So the question is why does * ignore this config for the
192.168.77.254
endpoint?
in sip.conf:
[router]
type=friend
context=default
How do you do monitoritng? How Server B knows that Servar A is down? I just
do a rsync and MySQL Replication, but I try to do a C program that monitor
Server. If you know how can I do this monitoring I will be pleasant with
you.
regards,
srsergio
-Mensaje original-
De: Senad J
[EMAIL PROTECTED] wrote:
How do you do monitoritng? How Server B knows that Servar A is down?
I just do a rsync and MySQL Replication, but I try to do a C program
that monitor Server. If you know how can I do this monitoring I will
be pleasant with you.
1. use heartbeat for failover between
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