Wolfgang S. Rupprecht wrote:
Have the spa3k use an "S0" dialplan:
PSTN Line:
...
Dial Plan 8:(S0 <:> )
...
PSTN Caller Default DP: 8
In the little bit of testing I did with an SPA-3000, I could not get it
to automatically send the call on to the Asterisk server
Joseph wrote:
In my test it seems to work fine for a little and than soon the phone
looses its time. At first the status shows clear, and then it appears to
get confused about the ntp time source and the time goes away on it.
I don't have that problem on the 10 or so phones I've updated to 7.4.
___
Anyone try the new Cisco firmware for the 79xx sip phones?
In my test it seems to work fine for a little and than soon the phone
looses its time. At first the status shows clear, and then it appears to
get confused about the ntp time source and the time goes away on it.
No features, just bug fixe
> > Well, incoming call handling on SPA-3000 kind of sucks at the
> > moment... but I don't see how it could be configured to ring a bunch
> > of phones anyway. At best it can deliver the call to a single
> > gateway/proxy, and even it really wants to answer the line first and
> > present a second
Very cool ...
Have you tried to compile it against Mono?
-JB Hawaii
Thorben Jensen wrote:
Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] På vegne af Henry Devito
Sendt: 16. marts 2005 16:17
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [Asterisk-Users] IPS
For me this version has issues with DHCP, and registrations. Although the
look of it is something of an improvement. I wish they would get their
existing feature set to work well before they make more changes.
-Nate
-Original Message-
From: Rod Bacon [mailto:[EMAIL PROTECTED]
Sent: Wed
Thomas Dingermann wrote:
Hi all,
i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with
Cisco ATA-186 3.1.1 atamgcp
We are used to make an special ;) blind transfer like
(Flash)Number(Hangup before anyone answers or ring).
Then * crashes (see below) if the man in the middle is an
cis
I have that on my HandyTone 286, and it's got some issues. At least
mine does. The page loading in FireFox very rarely ever completes, and
this means that I can't really provision this thing very well at home
(I only run Linux). I brought it to work and it loads up fine in IE
here.
I engaged their
> >Is that with channels recording ? ;)
> >
> >>
> We are running 40-50 simultanious calls at the call center here, and
> recording everycall in and out, with no problems
> On a Pentium 3ghz with 1gig ram.
Can you share with us what type of system this is (or motherboard
model if not a commercial
[EMAIL PROTECTED] (Kevin P. Fleming) writes:
> Well, incoming call handling on SPA-3000 kind of sucks at the
> moment... but I don't see how it could be configured to ring a bunch
> of phones anyway. At best it can deliver the call to a single
> gateway/proxy, and even it really wants to answer th
> I've posted this question twice without a single reply. Does that mean no
> one knows the answer, or no one cares to answer?
>
> I've been having an issue with an IAX2 trunk setup in Asterisk. Setup the
> trunk fine and it registers and works fine. I'm able to make outgoing calls
> from any e
Hi All,
I'm trying to hook up asterisk (CVS-HEAD-02/09/05-13:44:11 ) to a ADIT
600 via MGCP. Got it working really nice but now have a pretty bad problem:
1. When I perform a flash on the telephone, I usually get a second
dialtone, but when I dial, dialtone doesn't break. If I flash back and
fo
On Wed, 2005-03-16 at 04:18 -0600, Eric Wieling wrote:
> David Zanetti wrote:
> > But * won't bring up chan_zap at all:
> >
> > ERROR[2215]: Signalling requested is PRI Signalling but line is
> > in Unknown signalling 896 signalling
> > ERROR[2215]: Unable to register channel '1-30'
>
> > Agreed 100%. Think about how one might config a spa3k to accomplish
> > everything noted, plus some. :)
>
> Well, incoming call handling on SPA-3000 kind of sucks at the moment...
> but I don't see how it could be configured to ring a bunch of phones
> anyway. At best it can deliver the call
Guys.. I just noticed that my grandstream handytone 286 ata are having
problems with voice cutoffs... We can listen to the person on the zap
channel (x100p cards) without problems but they sometimes listen to us with
cutoffs.. like "He ...lo. ow...r.. you" and it comes and goes.. this
doesnt ha
"Giudice, Salvatore" <[EMAIL PROTECTED]> writes:
> Regardless, I would not call the database deficient because it
> truncates your data to 100 characters and doesn't warn you with an
> error. Get real. It is not as if this behavior is unexpected or some
> sort of a surprise.
Quoting the SQL/92 st
I use 1.0.5.22
Can't fault it.
Don't be afraid of upgrading to a newer version, you can always downgrade
again.
- Original Message -
From: "Paul Fielding" <[EMAIL PROTECTED]>
Sent: Tuesday, January 18, 2005 2:34 AM
Subject: [Asterisk-Users] Best Grandstream firmware to use?
I've seen l
I'm running [EMAIL PROTECTED] distro 0.6 which I think is the latest.
I've had a look in the iax.conf file: "bindaddr = 0.0.0.0" Assume that's
what you were referring to?
I don't think I lose any network settings on reboot or internet restart.
Just the registration constantly times out. I've g
I am trying to make *8 pickup extension to work.
if i get a call from-pstn or from anywhere i can get that call by dialing *8
in my IP phone,.
but if i dial *8 from an analog phone while ringing another phone of my
group..
Is there something else i have to configure...
please help me..
this is
I am trying to make *8 pickup extension to work.
if i get a call from-pstn or from anywhere i can get that call by dialing *8
in my IP phone,.
but if i dial *8 from an analog phone while ringing another phone of my
group..
Is there something else i have to configure...
please help me..
this is
As stated by John N., The Wiki can be pretty incomplete but it is a
great place to start.
In iax.conf. The bind parameter. However, this is only relevent if the
IP on the * box is changing. Since it is not, you should be fineas long
as it matches your static IP or is 0.0.0.0.
> I don't think t
ADCOM Corp wrote:
Does anyone know a good place to find a BRI S/T and U card for north
america?
Good luck; they are very rare. Even if you can find a card, most of them
do not offer US NI-1/2 firmware, so they can't be used on the network
here anyway.
The few that do work are listed on the wiki,
Hello Everyone,
I am trying to find a single port isdn pci card in the usa for asterisk,
but it seems everything is abroad.
Does anyone know a good place to find a BRI S/T and U card for north
america?
Thanks,
Greg
___
Asterisk-Users mailing list
Asteri
I have been trying for days to contact ABP to get the SIP firmware for our
MP-108-FXO to no avail. Would someone be kind enought to help me by sending me
the firmware. Thanks.
J
<>___
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Asterisk-Users@lists.digium.com
http:/
Great!! Something to work with.
Yes, I've searched the wiki and everything else I can think of, including
these archives (which to be honest are very hard to search for this sort of
thing). I've probably spent about 8-10 hours and hence the reason I've
given up on the personal research and start
I have just published a mini manual for "IPSwitchBoard for Asterisk" on the
Wiki:
http://www.voip-info.org/wiki-IPSwitchBoard+BETA
IPSwitchBoard is a FREE Windows.NET application that will:
. Organize all your extensions (automatically retrieved from Asterisk)
. Monitor all extensions
. Monit
Wiley Siler wrote:
Please check the Wiki (www.voip-info.org) and the list archive by
Googling site:lists.digium.com
IMHO, the Wiki information on IAX , especially IAX communication
between two Asterisk machines, is poorly written and seems quite out of date
Certainly little there addresses h
Paul Dugas wrote:
Maybe this would be enlightening:
http://www.samag.com/documents/s=9408/sam0411b/0411b.htm
I believe there's more that one way to connect Opterons; some better than
others.
Yes, that's an excellent article; thanks for the link.
In my previous messages I was certainly assuming that
Is that with channels recording ? ;)
___
We are running 40-50 simultanious calls at the call center here, and
recording everycall in and out, with no problems
On a Pentium 3ghz with 1gig ram.
Kyle
___
hi gary,
Brett, Gary wrote:
(can I just
use a single cat5 straight through cable between them ?? and cant the Digium
e1 cards operate ok in both modes?)
you need a crossover cable (not the same as a ethernet x-over)
take a look at: http://www.voip-info.org/wiki-crossover+T1+cable
frank
___
-Original Message-
I am new to Asterisk and currently work mainly with Cisco Callmanager.
With Callmanager I can setup partitions and call search spaces to
determine where a given phone can and can't dial. Does Asterisk offer
this type of functionality, and if so how?
-when you set up a
Yes this helps a lot. thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kristian
Kielhofner
Sent: Wednesday, March 16, 2005 12:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Capabilities
Park
Please check the Wiki (www.voip-info.org) and the list archive by
Googling site:lists.digium.com
Also, please include some more info. That is probably why you got no
answer...
Is your machine sitting behind a router or is it directly connected to
your broadband (assuming)?
If the machine is beh
On Wed, March 16, 2005 12:56 pm, Kevin P. Fleming said:
> If you are referring to the memory bus and/or the bus used to talk
> between the CPUs, that is exactly the reason that I suggested Opteron.
> Nothing out there in x86/x86-64 land even comes close to HyperTransport
> without spending large su
Hi,
I've posted this question twice without a single reply. Does that mean no
one knows the answer, or no one cares to answer?
I've been having an issue with an IAX2 trunk setup in Asterisk. Setup the
trunk fine and it registers and works fine. I'm able to make outgoing calls
from any extensio
Parker, Blake (MIS) wrote:
I am new to Asterisk and currently work mainly with Cisco Callmanager.
With Callmanager I can setup partitions and call search spaces to
determine where a given phone can and can’t dial. Does Asterisk offer
this type of functionality, and if so how?
***Blake Parker*
Rich Adamson wrote:
Agreed 100%. Think about how one might config a spa3k to accomplish
everything noted, plus some. :)
Well, incoming call handling on SPA-3000 kind of sucks at the moment...
but I don't see how it could be configured to ring a bunch of phones
anyway. At best it can deliver the c
Hi All,
Somebody has installed shadydial?, shadidial is used for predictive dialers,
I need to know the following: when a call is answered, this call es sending
a to agent, but I need that the agent when receive the call in the desktop
appear all information of this number.
Somebody have experi
mattf wrote:
In our experience having more processors doesn't really matter on the x86
platform because of the limitations of the motherboard bus.
The "motherboard bus" is not very specific.
If you are referring to the PCI bus, then most dual/quad/etc. Opteron
boxes have multiple independent PCI b
Parker, Blake (MIS) wrote:
I am new to Asterisk and currently work mainly with Cisco Callmanager.
With Callmanager I can setup partitions and call search spaces to
determine where a given phone can and can’t dial. Does Asterisk offer
this type of functionality, and if so how?
In Asterisk those
> > To avoid legal issues down the road, I'd suggest handling it via a
> > local pstn line (one way or another), and install a Red Phone with
> > a normal pstn line for emergency use. (The pstn line for the Red
> > Phone 'could' be used for incoming faxes as well, and when combined
> > with somethi
Ok, we all get it, some people prefer mysql, some people prefer postgres.
Now can we all just get on with our life or at least create a mailing list:
[EMAIL PROTECTED]
so that those people who think that mustard tastes better than ketchup have
somewhere more appropriate to argue.
Thanks,
I want to use a cordless (radio) phone as an extension on Asterisk.
When I ring the extension, there is no indication of ringing.
The ringing goes out to the base (I plugged a phone into the "phone" jack on
the base), but the base does not register ringing.
When I connect the base to PSTN, it works
Title: Asterisk Capabilities
-Original Message-From: Parker, Blake (MIS)
[mailto:[EMAIL PROTECTED]Sent: Wednesday, March 16, 2005
12:41 PMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] Asterisk Capabilities
I am new to
Asterisk and currently work mainly w
check linphone. AFAIK there is a linphonec for command line. I think
there is also a joshua (sample application) for osip which is CLI based.
regards,
klaus
Olle E. Johansson wrote:
Klaus Peras wrote:
Hey there,
does anybody know a CLI SIP Client für Linux?
I think you may find one in Vovida.org
Hi there!
I'm trying to set up a Asterisk server.
I'm having some problems with the music I should listen during
on hold.
Here a piece of extensions.conf I used to test this feature:
exten => 98,1,Answer
exten => 98,2,MusicOnHold()
Title: Asterisk Capabilities
I am new to Asterisk and currently work mainly with Cisco Callmanager. With Callmanager I can setup partitions and call search spaces to determine where a given phone can and can’t dial. Does Asterisk offer this type of functionality, and if so how?
Blake Park
Hello all,
I want to set up a time limit for IAX register.
Anyone knows how could I do that?
Any parameter?
Any clue will be wellcomed.
Thanks for you time.
Ismael.___
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http://lists.digium.c
Vladyslav wrote:
Is that with channels recording ? ;)
If you have a fast disk subsystem, yes. Recording calls is not CPU
intensive, only transcoding is (at that call volume, anyway).
___
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http://
On Wednesday 16 March 2005 16:52, Steven Critchfield wrote:
> On Wed, 2005-03-16 at 15:04 +, Andy and Jayne Slim wrote:
> > I'm sure this is a pretty basic problem, unfortunately I am a telecomms
> > rather than a Linux person so any suggestions would be most appreciated.
> > I have successful
Rich Adamson wrote:
To avoid legal issues down the road, I'd suggest handling it via a
local pstn line (one way or another), and install a Red Phone with
a normal pstn line for emergency use. (The pstn line for the Red
Phone 'could' be used for incoming faxes as well, and when combined
with somethi
I was able to get Asterisk working with the demo on
FreeBSD 5.3 without crashing, but not the music on hold, so I just have that
disabled for now, but I'm ready to get some IP hardware working.
So I picked up a Cisco 12sp+ IP phone
(mistake?) and am having difficulty finding any truly helpf
Hello,
We tried a Dual Processor AMD system last year and were greatly
dissapointed. A single P4 system was much cheaper and actually outperformed
the Dual AMD.
Is anyone actually running an octal AMD system out there?
In our experience having more processors doesn't really matter on the x86
p
Matt wrote:
How exactly does Asterisk provide E911 service??
Could you ask a slightly more open-ended and ambiguous question next
time? This one might actually have some real answers...
Asterisk does not provide _any_ service, the user configuring Asterisk
makes that happen. Asterisk can be used
On Wed, 2005-03-16 at 18:31, Kevin P. Fleming wrote:
> [EMAIL PROTECTED] wrote:
>
> > If we want a box that can perform 60 calls. What would be apoproximate
> > budget
> > for that using AMD x86-64 ?
>
> 60 calls can easily be done on a 3.4GHz Pentium 4 box, no special
> hardware is required.
> How exactly does Asterisk provide E911 service??
It doesn't do anything with 911. You tell * what to do when someone
dials 911 via your dialplan.
To avoid legal issues down the road, I'd suggest handling it via a
local pstn line (one way or another), and install a Red Phone with
a normal pstn
) in new
> >stack
> >-- Executing DBput("SIP/103-dfb6", "RepeatDial/103=901") in new
> >stack
> >-- DBput: family=RepeatDial, key=103, value=901
> >-- Executing DBget("SIP/103-dfb6", "recv=Record/103") in new stack
>
Hi,
I am playing with conferencing, but might have hit a bug... Any use who
wants to hang up or leave the conference should press the # key, after
which they get a "goodbye" message and the call gets disconnected.
However, this does not happen. whatever keys are pressed by whichever
party gets
Jason,
>>exten => s, 4, Dial(${VOICEPULSE}/011${ARG1}, ${LONGTIMEOUT}, Tt)
When I removed T and t options from dial command, the DTMF digit
recognition started working. Working line is below
exten => s, 2, Dial(${VOICEPULSE}/011${ARG1}, ${LONGTIMEOUT})
I will not change the features.conf, unles
> I believe the driving factors for this are the ability to commercially
> license Mysql for product integration over PostgreSQL's BSD license,
This is a ridiculous FUD statement. Are you actually trying to suggest that
one cannot commercially license PostgreSQL?
That's simply FALSE.
The primar
On Wed, 2005-03-16 at 15:04 +, Andy and Jayne Slim wrote:
> I'm sure this is a pretty basic problem, unfortunately I am a telecomms
> rather
> than a Linux person so any suggestions would be most appreciated. I have
> successfully downloaded and installed the various Asterisk packages.
>
> -Original Message-
> From: Giudice, Salvatore [mailto:[EMAIL PROTECTED]
> As for your 'artist license with your data' comment, put it into some
> context. I would blame a programmer for trying to insert a
> string of 255
> characters into a field only 100 character wide. Maybe you could
Klaus Peras wrote:
Hey there,
does anybody know a CLI SIP Client für Linux?
I think you may find one in Vovida.org
/O
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Once you run Dial from an AGI script, you lose control of the call via
the AGI script.
Jean-Michel Hiver wrote:
(obviously if you do other magic in your dialplan this needs to be
adjusted. The important part is the 'g' flag to Dial (go on after
hangup), and the NoOp which echos the dialstatus
How exactly does Asterisk provide E911 service??
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[EMAIL PROTECTED] wrote:
If we want a box that can perform 60 calls. What would be apoproximate budget
for that using AMD x86-64 ?
60 calls can easily be done on a 3.4GHz Pentium 4 box, no special
hardware is required.
___
Asterisk-Users mailing list
Ast
ting DBget("SIP/103-dfb6", "recv=Record/103") in new stack
-- DBget: varname=recv, family=Record, key=103
-- DBget: set variable recv to on
-- Executing GotoIf("SIP/103-dfb6", "1?7:9") in new stack
-- Goto (from-sip,901,7)
-- Executing SetVar(&qu
Use whichever you want. Go get your own benchmarks. I'm sure you will
find benchmarks all over the web based on different conditions. The fact
remains that enterprises are deploying MySQL 4:1 over postergreSQL. I
believe the driving factors for this are the ability to commercially
license Mysql for
> I changed the sequence first disallow and than allow. After
> restarting * it is working now!
> I am sure I copied the table and did not change it, ... somewhere it
> must have the wrong order.
>
>
>
> Thanks for your patient with me!
Glad we got it working.
-Matthew
(obviously if you do other magic in your dialplan this needs to be adjusted.
The important part is the 'g' flag to Dial (go on after hangup), and the NoOp
which echos the dialstatus and hangupcause variables to the console.
How would you do this in an AGI script? Basically what I have at the
Thanks Kevin for this info,
If we want a box that can perform 60 calls. What would be apoproximate budget
for that using AMD x86-64 ?
µSelon "Kevin P. Fleming" <[EMAIL PROTECTED]>:
> Erick Perez wrote:
> > And what people are using to deploy super servers with astersik?
> > Itanium with linux? c
tDial, key=103, value=901
-- Executing DBget("SIP/103-dfb6", "recv=Record/103") in new stack
-- DBget: varname=recv, family=Record, key=103
-- DBget: set variable recv to on
-- Executing GotoIf("SIP/103-dfb6", "1?7:9") in new stack
-- Goto (from-sip,901
Where did you get 1.05.23 from? The doc is available on the grandstream
site but not the actual firmware.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: Tuesday, March 15, 2005 11:48 PM
To: asterisk-users@lists.digium.com
Subject: Re: [A
Erick Perez wrote:
And what people are using to deploy super servers with astersik?
Itanium with linux? clusters of itanium with linux? or some RISC
processor with some *nix? cause it seems asterisk is only 100%
supported on Linux/Intel
or am i totally wrong?
The highest-performing "standard" hardw
And what people are using to deploy super servers with astersik?
Itanium with linux? clusters of itanium with linux? or some RISC
processor with some *nix? cause it seems asterisk is only 100%
supported on Linux/Intel
or am i totally wrong?
On Wed, 16 Mar 2005 05:51:18 -0600, Rich Adamson <[EMAI
> Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] På vegne af Henry Devito
> Sendt: 16. marts 2005 16:17
> Til: Asterisk Users Mailing List - Non-Commercial Discussion
> Emne: Re: [Asterisk-Users] IPSwitchBoard BETA
>
> I installed this and it seems to be working great. Good jo
Steve can you post your Cisco configs? Can you post the configs from your *
box that pertain to your issue?
- Original Message -
From: "Steve Blair" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, March 16, 2005 9:35 AM
Subject: [Aste
Hi Vladyslav,
Use 0.0.2pre1, but add the line
fax.verbose = TRUE;
just after
fax_init(&fax, calling_party, NULL);
That will turn on the detailed logging.
Is the listing you posted the entire log? It looks like there should be
more.
One common mistake people make - Did you use the "|
Hello:
Has anyone on this list had to configure hairpinning on a Cisco
gateway running IOS 12.2 or 12.3 and using a PRI for connectivity
to the PSTN? If so could you tell me how it is done? I'm told this
is the source of my call transfer problems and yet I cannot find
clear instructions for how the
Hello:
Has anyone on this list had to configure hairpinning on a Cisco
gateway running IOS 12.2 or 12.3 and using a PRI for connectivity
to the PSTN? If so could you tell me how it is done? I'm told this
is the source of my call transfer problems and yet I cannot find
clear instructions for how the
Matthew Boehm wrote:
*CLI> Urgent handler
-- SIP Seeding peers from Astdb: '3044' at [EMAIL PROTECTED]:64718 for
120
Urgent handler
-- Registered SIP '3044' at 64.XX.XX.XX port 17524 expires 120
Codecs : 0x10c (ulaw|alaw|g729)
Codec Order : (g729|ulaw|alaw)
Using the following table:
C
Hi Ppl.
Once, couple weeks ago when I have updated * from CVS-HEAD something
happen and I could not send a fax anymore.
After that I have tried previous * CVS versions with different versions
of spandsp (0.0.1, 0.0.2pre4, 0.0.2pre10) but without any changes.
I have tried that on Fedora Core 2 with
I installed this and it seems to be working great. Good job. Just one
question though, What is the shared extensions file?
- Original Message -
From: "Thorben Jensen" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Wednesday, March 16, 2005
On Tue, Mar 15, 2005 at 08:38:04AM -0600, Matthew Boehm wrote:
> I'm not a PRI expert and therefore don't know what this debug stuff means
> for PRI, so if anyone can help me here...
> I'm running the latest libpri and zaptel from CVS.
> Keep in mind that everything works fine when using the STABLE
On Tue, 15 Mar 2005 14:13:40 -0500, Kanuri, Seshu (Company IT)
<[EMAIL PROTECTED]> wrote:
> atxfer => *2 ; Attended transfer
Remove attended transfer capability and then you will be able o enter *2XXX
Jason
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Asterisk-Users mailing
On March 16, 2005 07:12 am, pixer wrote:
> Unfortunately I have already also tried this, without results.
> I do not know what to do any more..
Was it an entirely different motherboard (different manufacturer)? If so,
it's time to call Digium and open a ticket. It sounds like the card is DOA.
Dear All,
I've setup got a Asterisk and pgSQL combi that works fine. I'm about to
perform the migration deployment when I noticed a issue which I need some
expert advise here.
When user connect to Voicemail, the CPU Load of the machine will shoot up
to around 50 - 60%, and its causing sound disto
I'm sure this is a pretty basic problem, unfortunately I am a telecomms rather
than a Linux person so any suggestions would be most appreciated. I have
successfully downloaded and installed the various Asterisk packages.
However, when I try to start Asterisk, I immediately get a message saying
Dan, Thanks for the time helping me out. I figured everything out except
for the patch.
7. cd to asterisk/apps and run patch -p0 <
path-to/apps-meetme-cbmysql.txt
When I do this step it errors out and asks for the file to patch.. When I
look at the apps-meetme-cbmysql.txt It shows the file na
*CLI> Urgent handler
-- SIP Seeding peers from Astdb: '3044' at [EMAIL PROTECTED]:64718 for
120
Urgent handler
-- Registered SIP '3044' at 64.XX.XX.XX port 17524 expires 120
Codecs : 0x10c (ulaw|alaw|g729)
Codec Order : (g729|ulaw|alaw)
Using the following table:
CREATE TABLE cust
Matthew Boehm wrote:
Ronald Wiplinger wrote:
[mysql1]
dsn => astconf
username => root
password => MyPassword
pre-connect => yes
You are not using the ODBC drivers. You can remove that [mysql1] stuff
from your res_mysql.conf
Removed, but still no codecs
br
__
This Postgres vs. MySQL business is ultimately just a religious debate, like
PC vs. Mac, Ford vs. Chevy, or Kirk vs. Picard. They both work; they both
have their plusses and minuses; and debates about which are better never
convince anyone to change their preconceived ideas. It's also about as
on
Gianluca,
Did you install the .59r. Version of mpg123? The most common problem I
have seen for this is that people keep installing the 59q or 59g version
of mpg123. 59r is the way to go.
http://www.voip-info.org/wiki-mpg123
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
Thanks Steven, that was really a simple solution I overlooked. I added
appropriate context=siphones-superuser in the user settings in sip.conf,
commented out the includes under default and all inbound/outbound security
accounts are routed as I intended.
You were right, even unregistered SIP
I have * running with sipgate.de so that works fine. However, if all
you want is to use * as a softphone, you'd be better off using an actual
softphone -- * would be overkill for that, and it still wouldn't be as
easy to use as a proper softphone.
> -Original Message-
> From: Christian Sc
Martijn van Oosterhout wrote:
> On Wed, Mar 16, 2005 at 03:25:17PM +0800, Ronald Wiplinger wrote:
>> Mar 16 15:13:45 DEBUG[29502]: Raw Hangup 69.73.19.178:4569, src=14,
>> dst=1259 Mar 16 15:13:45 DEBUG[29502]: MySQL RealTime: Update SQL:
>> UPDATE sip_buddies SET name = '621' WHERE allow = 'g729'
Nicolas,
>> I have setup flash pannel, ... looks nice, but so far I could not
>> configure it to get more than 4x7 buttons.
>> I tried to make the buttons smaller, but than just the entire picture is
>> smaller.
>
>What did you change in op_style.cfg? You can have literally hundred of
>buttons per
Hi,
I also wrote a PHP scripts that generate op_style.cfg. You specify how many
rows x cols and the icons/buttons/text alignment are properly scaled.
(i.e. you defined a 5 x 20 for 100 buttons, button height will be small so
"Line", "CallerID", "Timer" position will be "adjusted")
Script not 10
Ronald Wiplinger wrote:
> vpbx*CLI> realtime update sippeers allow g729 name 621
> Failed to update. Check the debug log for possible SQL related
That is the wrong format of the command. Notice the incorrect SQL that
was queried? Type "realtime update" by itself to see an example.
> That is
On Wed, 2005-03-16 at 14:20 +, Razza wrote:
> Chris Blake wrote :
>
> -%<-
> >If anyone can help I`ll send the call file to you, or is it ok to
> clutter the list with it ?
> -%<-
>
> 'Clutter' the list I'd be interested and at least it is pertinent to *
> ;o)
I am almost sur
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