Hey everyone. I recenty installed a server at a datacenter offsite and
the thing is getting hammered with invalid ssh logins so I decided to
use some iptables.
I included my ruleset here. I was wondering if I could get some feedback
based on my ruleset from those of you using iptables in product
Details about externnotify and its arguments can be found here
http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf
M
RR wrote:
Hello all,
does anyone have a clever way of creating a customised email that goes
out as result of the voicemail notification. And I don't mean Editing
what
You can have your own external script to do whatever you want when vm is
left
from voicemail.conf:
; If you need to have an external program, i.e. /usr/bin/myapp
; called when a voicemail is left, delivered, or your voicemailbox
; is checked, uncomment this:
;externnotify=/usr/bin/myapp
M
RR w
Hello all,
does anyone have a clever way of creating a customised email that goes
out as result of the voicemail notification. And I don't mean Editing
what you want in the emailbody, emailsubject, serveremail etc
keywords. I mean custom in the sense that it has that info but the
email is stylise
Why not specify, several different ports, in your sip.conf?
You can have Asterisk listen on port 5060, 5061, 7080 etc as many as you
want; just make sure the port is not taken by some other application.
I have 8-phone lines (on sip.conf) and asterisk is listening each line
on a different port.
--
#
I'm actually only looking at the code at this point in
time (only in voicemail.c), hence my slightly
technical list of questions. How would the user find a
better way to change the password though? Isn't the
only way of changing the password by calling the
voicemail then selecting option 0 followed
The secret will still only be digits but encrypted
with Asterisk's public key. For a HW phone it is a
problem (unless there's a slot for a USB or CF card),
but for soft phones it shouldn't be a problem to
obtain the server's public key. Otherwise, the key may
just be retrieved from a central reposi
On Tue, Nov 28, 2006 at 09:12:19PM -0800, je . wrote:
> Encrypt voicemail password with Asterisk public key.
> Asterisk then decrypts the password and takes the hash
> of it and compares it with the hash stored in
> voicemail.conf. This way the real password is never
> stored in voicemail.conf
Earle,
I'm running Astlinux on a PIII 550 with 384 megs of ram. Booting from a
Compact Flash card. Non-Volatile storage on a USB Keydisk. I have three
SIP DID numbers in three different area codes here in Western Washington via
IPKall. I use a local SIP termination provider and also retain my
Cepstral sounds good and it's cheap. However, it still sounds like a
synthesized voice.
On 11/28/06, Hall, Eric M. <[EMAIL PROTECTED]> wrote:
I'm looking to set up asterisk to call customer 3 days before the app and
remind them we will be out to see them.
I'm looking for any ideas on good wa
Can you tunnel through a VPN connection?
On 11/28/06, Patrick <[EMAIL PROTECTED]> wrote:
On Tue, 2006-11-28 at 22:19 -0500, [EMAIL PROTECTED] wrote:
> We have many clients who live in third world countries where the ISPs
> purposely block traffic on port 5060.
>
> I know we could always change
So s,2 will only be executed if the user successfully
authenticated?
--- Luki <[EMAIL PROTECTED]> wrote:
> > Luki, thanks for the response. Could you give me
> an
> > example of the use of vmauthenticate in a very
> short
> > dialplan?
> >
> > Thanks
> > Jez
>
> *CLI>
> -= Info about applicati
Encrypt voicemail password with Asterisk public key.
Asterisk then decrypts the password and takes the hash
of it and compares it with the hash stored in
voicemail.conf. This way the real password is never
stored in voicemail.conf and there is no way to know
what the password is just by looking at
On Tue, 2006-11-28 at 22:19 -0500, [EMAIL PROTECTED] wrote:
> We have many clients who live in third world countries where the ISPs
> purposely block traffic on port 5060.
>
> I know we could always change the listening port in our Asterisk box.
> However, doing so will affect all our other users
I'm looking to set up asterisk to call customer 3 days before the app
and remind them we will be out to see them.
I'm looking for any ideas on good ways to do this. Also I think it would
be best to do some type of text to speech however I do not like the
sound of the free one . Any ideas?
Th
I am wondering the same, there must be a way to CORRECTLY bind on two ports.
On 11/28/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
We have many clients who live in third world countries where the ISPs
purposely block traffic on port 5060.
I know we could always change the listening port in
See the sample configfile, also its zapata.conf not zaptel.conf, the
channel=> acquires all the properties above it.
;callerid="Green Phone"<(256) 428-6121>
;channel => 1
;callerid="Black Phone"<(256) 428-6122>
;channel => 2
;callerid="CallerID Phone" <(256) 428-6123>
;callerid="CallerID Phon
We have many clients who live in third world countries where the ISPs
purposely block traffic on port 5060.
I know we could always change the listening port in our Asterisk box.
However, doing so will affect all our other users who use port 5060 with
no problems.
Is there any other solution? I gu
Everthing should be ABOVE the channel => line for that group of channels
On 11/28/06, Damon Estep <[EMAIL PROTECTED]> wrote:
Can anyone tell me if accountcode= should appear before or after the
channels definition that you want it to apply to?
I have several groups of channels (PRI) define
Can anyone tell me if accountcode= should appear before or after the
channels definition that you want it to apply to?
I have several groups of channels (PRI) defined in Zapata.conf, and wish
to specify accountcode by group.
If I put the accountcode= at the end of Zapata.conf it applies to
Ok, I am looking through the iax.conf file now and see 'guest' with no
password and tried to add another but none of these seem to let me log from
my softphone. I did restart asterisk each time.
Thanks!
On 11/28/06, blackwater dev <[EMAIL PROTECTED]> wrote:
I have asterisk installed and now w
I'm not looking for support for anything. I've alredy tried twice, going as
far as buying a Sun server (now running Linux) to get Solaris to work with
Asterisk and all I was able to use was the outdated packages from Mr.
Bendin, would not compile even using gmake + all the other dependancies.
Re
last I had heard, pretty much all FWD accounts that were created in the
past year or so no longer work with IAX. Still don't know why.
Timothy Parez wrote:
I've got the same problem here.
It can't register anymore --> timeout
Brian Capouch schreef:
I hadn't used FWD for quite a while. A cus
Andrew Joakimsen wrote:
> Solaris has poor support for anything in general.
maybe you are looking in the wrong places for support...
SOLARIS IS NOT LINUX
signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation provided b
Hi Peder -
Is the storage of actual voicemail messages in a database still limited
to ODBC? If so, why?
Yes. Why? Nobody has developed a voicemail solution that directly
connects to a *SQL database for message storage.
And is the use of mySQL and ODBC at the same time still a bad idea? I
Norbert Zawodsky wrote:
> RR wrote:
>
>> Mate, I can't say it with authority but I'm almost certain that the
>> only DB that a specific driver was written for is MySQL. I think if
>> you use res_mysql.o you should be able to talk to mySql directly
>> without needing ODBC.
>
>
>
> O.k., Nice to
Hello,
Can you recommend a good billing software for asterisk that supports
reseller accounts? Will be better if it haves opensource licence.
Best regards,
--
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular : +593 9 9
At 10:34 28/11/2006 -0700, Pavel Jezek <[EMAIL PROTECTED]> wrote:
I have done simple ael2 script, tak doing lookup in asterisk database
like: find full numer, if cidname isn't found, substract one digit from
right and try again, and so on
Thanks. If LookupCIDname doesn't come with its own
i had the same problem. the GUI stopped responding to configuration changes.
On 11/28/06, James Willing <[EMAIL PROTECTED]> wrote:
"Geoff Karl" <[EMAIL PROTECTED]> wrote:
> I just downloaded and installed the AsteriskNow appliance
> (http://www.asterisknow.org) . This looks like it has lots
hi,
i have tried to use asterisk now but it seems to me that it doesnt retain
configurations. have you experinced the same thing ??
On 11/26/06, Carlos Rojas <[EMAIL PROTECTED]> wrote:
Hello,
Anyone saw asterisknow, ?
Regards
___
--Bandwidth and C
Hi
I have the following setup to make outgoing calls:
X-Lite (build 34025) at home behind NAT -> Internet -> Asterisk at work
behind NAT -> Internet -> VoIP provider -> GSM gateway -> cellphone.
I just tried calling my own cellphone, but there is no sound either way.
Here's what I did on the
> > Yes, AgentCallbackLogin is deprecated, but it will not be removed
> > until after 1.4.
>
> Is there an isolated example somewhere of how to use existing dialplan
> logic and dynamic queue membership to simulate the current behaviour?
http://svn.digium.com/view/asterisk/trunk/doc/queues-with-
Are you sure you have configured the "dial plan" or "digit map" correctly in
the device? If you do a "sip debug" is the phone talking and sending INVITE?
On 11/28/06, Jerry Rasmussen <[EMAIL PROTECTED]> wrote:
I have an MG3 SIP ATA. This sip phone is registered and I am able to call
the phone
I have seen countless problems resolved by using "notransfer=yes" in
IAX.conf stuff like dropped calls, poor quality and even 1 way audio.
On 11/28/06, hugolivude <[EMAIL PROTECTED]> wrote:
Asterisk 1.2.7
RedHat 9.0
Hi,
I've run into some voice degradation problems with IAX2:
I frequently hav
Asterisk 1.2.7
RedHat 9.0
Hi,
I've run into some voice degradation problems with IAX2:
I frequently have calls come in on a DiD provided by an ITSP. I often
have to redirect these calls back out to the PSTN (i.e. to a cell
phone). When this happens, I don't want my server in the media path,
I
Hi for all,
I'm doing test with Asterisk and I have a question.
I've seen that exist differences between executing an AMI command (originate)
with channel ending in /n, but what's the meaning?
--
Nuria
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it was not a real code, but just a schema.
I can't write a more precise snippet of code, since I'm completely
unaware of your configuration.
But in any case, I would delegate all of the logic to your Ruby script
and keep the minimum in the extensions.conf.
AF.
Olivier Saulnier wrote:
> Anton F
I've got the same problem here.
It can't register anymore --> timeout
Brian Capouch schreef:
I hadn't used FWD for quite a while. A customer sent me an email last
week, "Is FWD broken when one tries to use it with IAX?"
I have been playing around, and indeed seems to be the case.
Is there a
Anton Frolov a écrit :
When registering the softphone:
OK, in which file do i do that??
SoftPhonesDB.insert("olivier", $ip);
Could you explain me what means "SoftPhonesDB.insert"?? It's not an AGI
commande, how can i use it??
In extensions.conf:
exten => 302,s,agi,script.rb,${EXTEN
Of course it does provide quite descent hangup notification (425Hz 200ms
200ms envelope for 40s), but it is asterisk/digium that in about 5%
fails to detect it. I have trie to alter Tx, Rx gain and I even change
the busy counter to value 2. Normaly it does detect hangup in about 2
seconds but f
IIRC, a "call" from the SIP perspective is any transaction or interaction
with a SIP device. So things that qualify as a "call" are things like
registration and qualification. Nothing to sweat about. You can suppress it
with "sip no debug" from the command prompt.
hth
-Original Message-
I didnt forgot the french translation, it's coming was just busy.
In 1-2 weeks i ll provide it to you
Regards
Le mercredi 25 octobre 2006 à 09:51 +0200, Alex a écrit :
> Hi all!
>
> We've released VoiceOne 0.4.0, a web-based and open source solution
> which allows to fully manage an Asterisk
I am using asterisk along with freepbx . When recording is enabled for a
extension the call record file made in /var/spool/asterisk/monitor contains
information like OUT(extension number)-(timestamp)-(uniqueid).wav . This can
be a big mess if there are more than 1000-2000 files in that folder and
On Tue, Nov 28, 2006 at 02:54:45PM +1100, Eric Bishop wrote:
> Hi all,
>
> We have Xeon-based system with only 1 (hyperthreaded) CPU (in a HP DL360).
> We are seeing high load on multiple meetme session as well as g729
> transcoding. My question is will putting an extra CPU help or does Asterisk
>
Looking at the CLI in Asterisk 1.2 it constantly reports "destroying
call" and gives an address of any of 6 sip phones connected to it.
However, there are no calls made by anyone in the last hour
so why is it destroying calls?
The phones are all on fixed IP - fixed by my firewall by the MAC
ad
On Tue, Nov 28, 2006 at 07:13:38PM +0100, Matic wrote:
> Hi,
>
> I have a small problem with tdm2400 with tone detection. About 5% of all
> calls doesn't end because Asterisk dosen't detect "busy" tone on remote
> hangup. I know that board does support current reversal and current off,
> but ho
On Wed, Nov 29, 2006 at 06:17:27AM +1300, kjcsb wrote:
> >>
> >>ln -s /usr/src/kernels/`uname -r` /usr/src/linux
> >>ln -s /usr/src/kernels/`uname -r` /usr/src/linux-2.6
> >
> >Unnecessary. Just install the relevant kernel-devel package. What
> >instructions are you following?
> >
> I already had k
On Tue, Nov 28, 2006 at 08:52:22AM -0800, je . wrote:
> I was wondering if we could protect against both.
> Sending a password encrypted would protect against
> eavesdropping. Once the password has been received,
> the hash of it is taken and compared with the hash of
> the password saved, so i
Hi,
Asterfex seems promising providing email to fax services.
With it, to send a fax to Mr Foo whose fax number is 123456789, an end user
just has to use its favorite email client and fill destination field (To:)
with [EMAIL PROTECTED]
I'm looking for an email client I could personnalize (config
I am using Zaptel for my Wildcard and got DTMF Tones in my conversations
since a long time.
To control this , i search for a possibility to control the detection of
DTMF and to control talkoff, called "dtmfthreshold".
As seen in mISDN* i am in need of an Patch to control the
"dtmfthreshold"
Hi,
I have a small problem with tdm2400 with tone detection. About 5% of all
calls doesn't end because Asterisk dosen't detect "busy" tone on remote
hangup. I know that board does support current reversal and current off,
but how do I configure asterisk to use tone detection first and if that
For the record, it does compile correctly, even on x86 Linux(if you
change the architecture flag) and it runs very reliably as well on the
machines I have run it on.
For the Sparc platform this is the only real option you have to run
Asterisk, so for that it's not bad at all, for anyone on x86 Li
On 28 Nov 2006, at 17:54, Doug Crompton wrote:
How does one process a return code in Asterisk?
Example...
exten => s,n,Playback(/tmp/podcast/${CALLERIDNUM})
exten => s,n,System(rm "/tmp/podcast/${CALLERIDNUM}.gsm")
If the caller hangs up on the playback command the file remove System
statem
How does one process a return code in Asterisk?
Example...
exten => s,n,Playback(/tmp/podcast/${CALLERIDNUM})
exten => s,n,System(rm "/tmp/podcast/${CALLERIDNUM}.gsm")
If the caller hangs up on the playback command the file remove System
statement after it never gets executed. The playback comm
why don't you want to manage everything in Ruby?
I would call the script with just the number called
exten => 302,s,agi,script,${EXTEN}
and then make all of the decisions based on the phone number inside of Ruby.
When registering a softphone, you could store its IP in a database. Then
your script
Solaris has poor support for anything in general. The Solaris version of
asterisk is outdated, and doesnt even compile correctly. Is one individual's
flawed testing your basis to use Asterisk on Solaris?
On 11/28/06, Frank Tarczynski <[EMAIL PROTECTED]> wrote:
I'm looking to build the zaptel dr
On 29 Oct 2006, at 11:30, Darko wrote:
Hello List,
We are deveoping apication/system based on PHP5, Postgre,Ajax,ect..
It should be compleate sistem for realystate agensy and road worers
(agents)
and it will be distributed system.
We made very good inplementation based on asterisk and OSP fo
Hello,
i would like to use AGI and Ruby for communicate with a softphone. I
would like send the IP adress of the softphone, directly for the
extensions.conf file, as:
exten => 302,1,agi,/ruby/ipphone.rb|ip_adress
I know, with extensions.conf file, that i work on the softphone 302.
But, how c
ln -s /usr/src/kernels/`uname -r` /usr/src/linux
ln -s /usr/src/kernels/`uname -r` /usr/src/linux-2.6
Unnecessary. Just install the relevant kernel-devel package. What
instructions are you following?
I already had kernel-devel installed and was still getting the message "You
do not appear to
I write my AGI's in C and use the standard command line parser
"getopt_long()" to parse the options passed to the AGI. It looks a little
"wordy" but there is little doubt as to what is going on:
exten = s,n,agi(play-path,--debug,--path=${PROMO-PATH})
"getopt_long" does all of the heavy
I was wondering if we could protect against both.
Sending a password encrypted would protect against
eavesdropping. Once the password has been received,
the hash of it is taken and compared with the hash of
the password saved, so it also takes care of a local
attacker.
I could certainly use SSL/TL
On Tue, 28 Nov 2006 10:27:27 -0600 (CST)
Jason Parker <[EMAIL PROTECTED]> wrote:
> Yes, AgentCallbackLogin is deprecated, but it will not be removed
> until after 1.4.
Is there an isolated example somewhere of how to use existing dialplan
logic and dynamic queue membership to simulate the curren
I have an MG3 SIP ATA. This sip phone is registered and I am able to call the
phone from another softphone. However, I am unable to place a call from the
phone.
In addition, after calling the SIP ATA phone the sip phone does not see to hang
up the call in complete. Can anyone shed some light on
Yes, AgentCallbackLogin is deprecated, but it will not be removed until after
1.4.
- Original Message -
From: Miguel Paolino <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Sent: Monday, November 27, 2006 7:19:44 AM GMT-0600 US/Central
Subject: [asterisk-users] AgentCallbackL
I would say that whatever the solarisvoip svn repository has, is the latest
version of Zaptel for Solaris. If you didn't require Zaptel, you would be able
to use Asterisk 1.4 (stock). It's a trade off for now, I guess. The main
problem here, is that for kernel modules, the interface to the ke
Hello -
Desperately hoping someone has seen this before or can help:
In my Queue, Asterisk seems to be coming up with SIP 486 responses:
Called SIP/trainer2
-- Got SIP response 486 "Busy Here" back from 221.219.11.71
-- SIP/trainer2-08fae890 is busy
These are *definitely* not coming fro
I also use ldapadmin, but for many common tasks, I use custom command
line scripts that wrap standard ldap commands. I also wrote a couple
simple CGI's that allow users to change their password, select their
preffered shell, update GECOS, and a few other options. If you are
managing Asterisk users
Am Dienstag, den 28.11.2006, 17:23 +0200 schrieb Jean-Marc Salsa:
> Hi,
>
> I have tried to use the Record Command in Asterisk,
>
> Here is the configuration :
> exten => record,1,Answer
> ...
> exten =>
> record,n,Record(/var/spool/asterisk/record/${CALLFILENAME}:WAV)
> exten => record,n,Playb
Max,
How did you fix it? It seems like knowledge that could be shared, even if it
was basic oversight.
John
On 11/25/06, Max Bergmann <[EMAIL PROTECTED]> wrote:
Max Bergmann schrieb:
>
>
> How can i programming a Cisco 7961 to be used as busy lamp field?
>
> my configs :
>
> sccp.conf :
>
>
On Tue, 28 Nov 2006, Marnus van Niekerk wrote:
Klaus Darilion wrote:
Thanx, but AFAIK that can only be used with their service, not asterisk or
other SIP server.
AFAIK there is no "generic" sip-client for Symbian that you can use to
connect to any other sip-service than the service that prov
I second the vote for ldapadmin.
You can extend it with custom templates for your asterisk specific
attributes.
-ejay
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of marvin horst
Sent: Tuesday, November 28, 2006 6:52 AM
To: Asterisk Users Mailing List - Non-Commercial
In 2.6.15 kernels and higher, you can use "taskset" to pin a task to a
certain CPU. Here's a way to set httpd to the 2nd processor in a 4 way
system:
HTTPDPID=`ps -A | grep -a -A0 "httpd"`
taskset 0x0002 -p ${HTTPDPID:0:5}
-Original Message-
From: Don [mailto:[EMAIL PROTECTED]
Sent
Eric Bishop wrote:
You can't hanup channels with a call file you can only create them no?
*snipped
actually you could hangup a call using a call file
example
Channel: Tech/Dev-occurance
Application: Hangup
Data: somecausecodevar or digit equiv
_
Hi,
I have tried to use the Record Command in Asterisk,
Here is the configuration :
exten => record,1,Answer
...
exten => record,n,Record(/var/spool/asterisk/record/${CALLFILENAME}:WAV)
exten => record,n,Playback(vm-goodbye)
exten => record,n,system(/usr/local/bin/send-recording.pl --to ${EMAILA
Klaus Darilion wrote:
sillyant.com
Thanx, but AFAIK that can only be used with their service, not asterisk
or other SIP server.
M
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Have you looked at QueueMetrics?
http://queuemetrics.loway.it/
There are also several call center packages for Asterisk out there
that have all of the reports built into them that you want:
http://www.voip-info.org/wiki/view/Predictive+dialer
MATT---
On 11/28/06, Hernany Oliveira <[EMAIL PROTEC
The code from there is stable, and it has a lot of bugs fixed on it
that are fixed in later versions of the Asterisk 1.2 tree. The reason
it was forked is that the Asterisk developer community doesn't test
each release on Solaris and they put a lot of things into the code
that just plain won't wor
On Tue, Nov 28, 2006 at 08:30:55AM -0500, Frank Tarczynski wrote:
> I'm looking to build the zaptel drivers on a Solaris 10 X86 box. I've
> found the driver source code on
> https://svn.sunlabs.com/svn/solaris-asterisk but this source is posted
> along with Asterisk 1.2.7.1 Does anyone know
Does anyone have an xml interface to comedian mail setup for the aastra 480i
CT?
The old interface to comedian mail on the ADSI phones was a nice feature
although kind of slow loading. I thought I'd try to duplicate it on the SIP
phones, but didn't want to reinvent the wheel.
Marv Horst
RR wrote:
> Mate, I can't say it with authority but I'm almost certain that the
> only DB that a specific driver was written for is MySQL. I think if
> you use res_mysql.o you should be able to talk to mySql directly
> without needing ODBC.
O.k., Nice to hear. But I'm not sure *how* to "use res
Is it free, if not what's the cost and where can I get it from?
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"Geoff Karl" <[EMAIL PROTECTED]> wrote:
> I just downloaded and installed the AsteriskNow appliance
> (http://www.asterisknow.org) . This looks like it has lots of
> promise.
> Anyone know what the secret is to being able to actually login to the
> root console?
Yes, as I found out (rather pai
You could use Eric's solution to minimize glare, but in my experience having
the circuit set to CO Yields causes a ton of problems. This is only from my
experience with a couple of my customers and also my old PRI at the office.
Just about the time I figured out the problem with my circuit was gl
Hi!
I once tried Asterfax but it is really difficult to install and
configure, and running openoffice in an xvfb session, together with the
need for java is IMO overkill. I'm now using hylafax+iaxmodem with
virtual printer devices for windows clients and hy-email2fax for clients
which prefer
I'm looking to build the zaptel drivers on a Solaris 10 X86 box. I've
found the driver source code on
https://svn.sunlabs.com/svn/solaris-asterisk but this source is posted
along with Asterisk 1.2.7.1 Does anyone know of a fresher version? Is
this code considered "somewhat ready for prime t
I need something like Call Manager.
I need to know how many agents is logged in, how many calls are on queues,
transfer calls, hang up calls, reports and so on.
Everything related to a Call Center operation.
I have been looking for and I did not find anything.
-Mensagem original-
De: [E
Marnus van Niekerk wrote:
Anybody know of a SIP/IAX softphone for Symbian Series 60? (Apart from the
builtin Nokia one!)
sillyant.com
Tx
M
--
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
Thomas Alva Edison - Inventor of 1093 patents,
inc
On 11/27/06, Anil Ramsingh <[EMAIL PROTECTED]> wrote:
We use a windows ldap browser & editor called ldapeditor from
http://www.ldapeditor.com . Its the best free browser but it only runs on
m$ windows.
On 11/27/06, Steven Baker <[EMAIL PROTECTED] > wrote:
> Hello All,
> we are using asterisk+o
Anybody know of a SIP/IAX
softphone for Symbian Series 60? (Apart from the builtin Nokia one!)
Tx
M
--
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
Thomas Alva Edison - Inventor of 1093 patents,
including the light bulb, phonogram and moti
Hello List,
We are deveoping apication/system based on PHP5, Postgre,Ajax,ect..
It should be compleate sistem for realystate agensy and road worers(agents)
and it will be distributed system.
We made very good inplementation based on asterisk and OSP for distributed
offices and it will be part of
On Tue, 2006-11-28 at 07:18 -0500, Barry Fawthrop wrote:
> Hi all
>
> Is the use of a VPN between IP-PBX and VoIP Provider a useful tool?
> Since the QoS and general traffic of the Internet can never be
> predicted, would the implementation of a VPN between Client and VoIP
> Provider increase vo
Hi all
Is the use of a VPN between IP-PBX and VoIP Provider a useful tool?
Since the QoS and general traffic of the Internet can never be
predicted, would the implementation of a VPN between Client and VoIP
Provider increase voice quality and/or security or is the converse true ?
Thanks
Barry
Hi Friends,
I am facing a strange problem with DISA. I have installed and configured
Trixbox. I've created a secret extension i.e., 555 and called this extension in
Digital Receptionist using custom extension i.e., created in
extensions_custom.conf file.
When I call from my mobile phone to my
YES!!
It works ;-))
Best regards,
Olivier S.
Anton Frolov a écrit :
you should use a separator between the arguments as well. try
exten => 0470022762,2,agi,/ruby/ring.rb|192.168.0.10|5010
--
Olivier Saulnier
STEGANUX
1er étage DIAMECANS
BEL AIR
03410 St-Victor
T: 04.70.02.27.62
On 28 Nov 2006, at 08:47, Olivier Saulnier wrote:
Hello,
I want to use AGI for give some information for a softphone, as:
exten => 0470022762,2,AGI(/ruby/ring.rb 192.168.0.10 5010)
We use Ruby langage.
The line doesn't worksin as this, but works with shell command.
Also, if i modify my ruby sc
On 28 Nov 2006, at 03:01, Eric Bishop wrote:
I am trying to do it with FOP and Calling Circles. Both have closed
code. Anyway to do it from Asterisk?
You could use the 'Local' channel as the argument to the originate
command
and then set it in the dialplan.
Tim Panton
www.mexuar.net
w
Olivier Saulnier wrote:
> hello,
>
> I try:
> exten => 0470022762,2,agi,/ruby/ring.rb|192.168.0.10 5010
>
> And it doesn't works again...
> best regards,
> Olivier S
you should use a separator between the arguments as well. try
exten => 0470022762,2,agi,/ruby/ring.rb|192.168.0.10|5010
AF.
___
hello,
I try:
exten => 0470022762,2,agi,/ruby/ring.rb|192.168.0.10 5010
And it doesn't works again...
best regards,
Olivier S
Anton Frolov a écrit :
bonjour, Olivier
I think that your line should be something like:
exten => phone,2,agi,script.py|args
--
Olivier Saulnier
STEGANUX
1er ét
Olivier Saulnier wrote:
> Hello,
>
> I want to use AGI for give some information for a softphone, as:
> exten => 0470022762,2,AGI(/ruby/ring.rb 192.168.0.10 5010)
> We use Ruby langage.
> The line doesn't worksin as this, but works with shell command.
> Also, if i modify my ruby script for give
I corrected the problem. I think it was a problem with the bootrom not
being new enough, not sure because when you upgrade the bootrom it formats
the phone so maybe it was configs. I dont think so though because the
configs i used where the same even after the bootrom but thats not to say
tha
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