If you can figure out how to generate .call files from your DB
entries, you have it made.
Vicidial needs alot of work as far as I am concerned, for free it is
OK I guess. I think using meetme conference rooms for everything is a
kludgy hack, and the UI is less than nice (if you are into UIs).
I
Hello,
Thank you for the advice. I am sorry but I could not locate the problem
in the forum. Do you remember anything more specific about it? And was
it on asterisk-users? Do you remember year and month when it was seen?
Thanks a lot,
Roberts
On Tue, 2008-10-14 at 05:20 -0400, broadband Voice w
Hi everybody,
How can we add new contexts in asterisk realtime module? All I could figure
out after googling is that a new context HAS to be declared in
extensions.conf with 'switch => Realtime/@' under the context
name declaration. This works fine as long as we are adding extensions only
to this
> I did not know what I did but I bumped into something in the log that
says:
> [Oct 16 ...] ERROR[24536] res_config_mysql.c: MySQL RealTime: Ping
failed
> (2006). Trying an explicit reconnect.
> [Oct 16 ...] DEBUG[24536] res_config_mysql.c: MySQL RealTime: Server
Error
> (2006): MySQL server has
Hi,Here is the link to send free SMS to any mobile in India. I use it too :-) http://www.indyarocks.com/register_step1.php?invitor=MjEyMjkyMA==&emailencryp=YXN0ZXJpc2stdXNlcnNAbGlzdHMuZGlnaXVtLmNvbQ==.-Sunkara RaviPrakashPlease note: This message was sent to you by a user at Indyarocks.com. Click
look at Vicidial
ram
On Thu, Oct 16, 2008 at 4:46 PM, yavuz yildirim <[EMAIL PROTECTED]> wrote:
> hi everybody
>
> This is Yavuz YILDIRIM
>
> I am software developer.I have a some problems in asterisk.
> I am using mysql db. Realtime using asterisk modules. On db i am using
> calling hundred fie
Tzafrir:
Following the comments on your post, I started checking (after breaking my
head 'googling') the UDP ports in use, and found out that the script that my
Asterisk is running was using UDP connection too. This caused that ports
from 10,000 to 20,000 could not be used by Asterisk.
I change t
unfortunately I still see it in 1.6.0...
__Yehavi:
2008/10/17 broadband Voice <[EMAIL PROTECTED]>
> I am having a similar problem and I'm using Asterisk 1.4.19 and have that
> problem on some calls through our calling card platforms. Someone suggested
> we use 1.4.3 and ha
On Thu, Oct 16, 2008 at 7:25 AM, Rodolfo Alcazar Portillo
<[EMAIL PROTECTED]> wrote:
> Am Mittwoch, den 15.10.2008, 20:51 -0400 schrieb C F:
>> Being a Panasonic dealer and having more than 50 Asterisk system in
>> production, I can tell you that if this is your first Asterisk
>> project, then go w
Steve, I got to congratulate you on this one, very nicely written and
you make a lot of sense.
However to the OP my advice: As Steve has mentioned in his email "so
learn it prior to the demo" and you have indicated as well: "In some
point we must start this new tech". The ideal way would be to fir
On Thu, Oct 16, 2008 at 8:45 AM, Olivier <[EMAIL PROTECTED]> wrote:
>
>
> 2008/10/16 C F <[EMAIL PROTECTED]>
>>
>> * Live call screening - Yes there is a hack that can do it, but it's a
>> hell of a hack.
>> * Phones that can do most of the usefull features supported by the PBX
>> for a reasonable
Hi all,
After loading 1.6.0.1, I notice that I always have the "VOX" effect
on Meetme conferences whether I have the "o" option set in the dial plan
or not. Is anyone else seeing this?
Although I'm now running 1.6.0.1, I'm also seeing this on a system
still running 1.6.0beta9.
You generally don't need to enter the public IP of the router into the
Cisco, just setting nat_enable to 1 is almost always sufficient. * is
smart enough to realize that the IP of the packet is the public IP of
the phone.
Tony Mountifield wrote:
> I have used Grandstream phones for years, and
I am having a similar problem and I'm using Asterisk 1.4.19 and have that
problem on some calls through our calling card platforms. Someone suggested
we use 1.4.3 and have not tried it yet. Any comments from the group.
On Tue, Jul 29, 2008 at 1:19 AM, Yehavi Bourvine +972-8-9489444 <
[EMAIL PROTEC
Neal:
Try having on sip.conf:
srvlookup=no
Regards,
Juan
[EMAIL PROTECTED] wrote:
Hello,
Thanks for your replies.
We checked our sip.conf and we have canreinvite=no already. I agree
it could be a firmware issue. I will get another vendors phone hooked
up to the pbx before going crazy w
On Thursday 16 October 2008 13:59:46 Karl Fife wrote:
> On Thu, 16 Oct 2008 11:47:15 -0500, "Tilghman Lesher"
>
> <[EMAIL PROTECTED]> said:
> > If you could explain what ISN is, that might help.
>
> an ISN, stands for ITAD Subscriber Number, which in turn stands for
> 'Internet Telephony Administra
Gordon Henderson <[EMAIL PROTECTED]> writes:
> Hm. Drayteks are on my list of modems to turn any SIP ALG off on! You must
> have a goodun :)
Drayteks do indeed mess with SIP packets. If you keep STUN/ICE off on
the phone and let Draytek mangle the packets, there is a chance that
things will work
On Thu, 2008-10-16 at 13:59 -0500, Karl Fife wrote:
> On Thu, 16 Oct 2008 11:47:15 -0500, "Tilghman Lesher"
> <[EMAIL PROTECTED]> said:
> >
> > If you could explain what ISN is, that might help.
>
> an ISN, stands for ITAD Subscriber Number, which in turn stands for
> 'Internet Telephony Administ
On Thursday 16 October 2008 13:38:00 Ken D'Ambrosio wrote:
> Hi, all. This e-mail is a follow-up to an exchange I had several weeks
> ago. I've got an Asterisk box with a dual-span T1 card. I want to place
> it between the PSTN and my company's legacy PBX. I actually did do that,
> but internat
Jerry Geis wrote:
> -- Attempting call on DAHDI/1ww for
> [EMAIL PROTECTED]:1 (Retry 1)
> [Oct 16 14:36:42] WARNING[16408]: chan_dahdi.c:8132 dahdi_request:
> Unknown option 'w' in '1ww'
> [Oct 16 14:36:43] WARNING[16408]: chan_dahdi.c:1481 dahdi_enable_ec:
> Unable to enable echo
On Thu, Oct 16, 2008 at 1:09 PM, Wilton Helm <[EMAIL PROTECTED]> wrote:
>> having two NICs on the same subnet
>
> I'm trying to wrap my brain around that in the larger network picture. Two
> NICs in the same subnet (presumably on the same computer) would have access
> to the same other devices. T
On Thu, 16 Oct 2008, Tony Mountifield wrote:
> In article <[EMAIL PROTECTED]>,
> Gordon Henderson <[EMAIL PROTECTED]> wrote:
>> On Thu, 16 Oct 2008, Tony Mountifield wrote:
>>
>>> I have used Grandstream phones for years, and have just started testing
>>> a Cisco 7940 (with SIP firmware 7.4). I ha
On Thu, 16 Oct 2008 11:47:15 -0500, "Tilghman Lesher"
<[EMAIL PROTECTED]> said:
>
> If you could explain what ISN is, that might help.
an ISN, stands for ITAD Subscriber Number, which in turn stands for
'Internet Telephony Administrative Domain Subscriber Number'.
Essentially it is a very clever
-- Attempting call on DAHDI/1ww for
[EMAIL PROTECTED]:1 (Retry 1)
[Oct 16 14:36:42] WARNING[16408]: chan_dahdi.c:8132 dahdi_request:
Unknown option 'w' in '1ww'
[Oct 16 14:36:43] WARNING[16408]: chan_dahdi.c:1481 dahdi_enable_ec:
Unable to enable echo cancellation on channel 1 (No su
Hi, all. This e-mail is a follow-up to an exchange I had several weeks
ago. I've got an Asterisk box with a dual-span T1 card. I want to place
it between the PSTN and my company's legacy PBX. I actually did do that,
but international calls from the legacy PBX were having the "011" stripped
off
On Thu, Oct 16, 2008 at 04:49:37PM +0100, Julian Lyndon-Smith wrote:
> I seem to remember that there was a change to q931.c that meant a line
> did not drop immedately, and then that change was reverted ?
>
> I think that these are the lines of code:
>
> /* wait for a RELEASE so that sufficient
Olivier wrote:
> 2008/10/16 Torbjörn Abrahamsson <[EMAIL PROTECTED]>
>
>> You could use #exec statements in one of your config-files.
>
>
> Could you elaborate ?
> Which of /etc/asterisk files are thinking of ?
>
You can put it in any of the files, as far as I know. sip.conf may be a
good pla
Hi,
Am Donnerstag, den 16.10.2008, 09:37 +0800 schrieb GNUbie:
> Hello Karsten,
>
> On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer <[EMAIL PROTECTED]> wrote:
> >
> > Please post Your sip.conf.
> > Which IP-Address do You configure in the snom for Your asterisk? (eth0
> > or eth1)?
>
> The SN
In article <[EMAIL PROTECTED]>,
Gordon Henderson <[EMAIL PROTECTED]> wrote:
> On Thu, 16 Oct 2008, Tony Mountifield wrote:
>
> > I have used Grandstream phones for years, and have just started testing
> > a Cisco 7940 (with SIP firmware 7.4). I have found something puzzling
> > and don't know whet
> having two NICs on the same subnet
I'm trying to wrap my brain around that in the larger network picture. Two
NICs in the same subnet (presumably on the same computer) would have access
to the same other devices. This could potentially increase bandwidth
(maybe?) and offer redundancy (if NICS,
Am Donnerstag, den 16.10.2008, 11:54 -0400 schrieb Barry L. Kline:
> Rodolfo Alcazar Portillo wrote:
> > Does anyone knows how to trigger a phone call from a bash command?
> Yes.
> Do you mean that you need something more than:
> http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
Am Donnerstag, den 16.10.2008, 11:02 -0500 schrieb David A. Bandel:
> 2008/10/16 Rodolfo Alcazar Portillo <[EMAIL PROTECTED]>:
> > Hi.
> >
> > Does anyone knows how to trigger a phone call from a bash command?
>
> Two ways:
> 1. look in your asterisk source directory for a file called
> sample.ca
On Thursday 16 October 2008 10:46:51 Olivier wrote:
> Is Incomplete() application an acceptable work around for ISN ?
If you could explain what ISN is, that might help.
--
Tilghman
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On Thu, 16 Oct 2008, Tony Mountifield wrote:
> I have used Grandstream phones for years, and have just started testing
> a Cisco 7940 (with SIP firmware 7.4). I have found something puzzling
> and don't know whether it's just a limitation or something I haven't
> done correctly.
>
> The Asterisk s
Hi, I got a card from Digium TDM with 2 FXO modules (red ones). There is a
problem that has me quite upset and is that asterisk always detect tones
repeated two, three or more times.
i mean, if i press 123 on my phone. asterisk detects somethin like:
111223
or 112333
or things like that.
In my setup, I am using TDM Wildcards analog connections and the Asterisk PBX
box does the converting to my SIP Phones. I had similar problem, when Asterisk
could not recognize my DTMF tones, so I had to tune the FXO modules. Here is
the link to the page:
http://www.voip-info.org/wiki/view/A
2008/10/16 Rodolfo Alcazar Portillo <[EMAIL PROTECTED]>:
> Hi.
>
> Does anyone knows how to trigger a phone call from a bash command?
Two ways:
1. look in your asterisk source directory for a file called
sample.call. This will show you what you need to put into your spool
directory for * to plac
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Rodolfo Alcazar Portillo wrote:
> Does anyone knows how to trigger a phone call from a bash command?
Yes.
Do you mean that you need something more than:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
Bary
-BEGIN PGP SIGNATU
I seem to remember that there was a change to q931.c that meant a line
did not drop immedately, and then that change was reverted ?
I think that these are the lines of code:
/* wait for a RELEASE so that sufficient time has passed
for the inband audio to be heard */
if (pri->a
I have used Grandstream phones for years, and have just started testing
a Cisco 7940 (with SIP firmware 7.4). I have found something puzzling
and don't know whether it's just a limitation or something I haven't
done correctly.
The Asterisk server is directly on the Internet with a public IP.
The p
Is Incomplete() application an acceptable work around for ISN ?
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Hi.
Does anyone knows how to trigger a phone call from a bash command?
Thx!
--
Rodolfo Alcazar
Responsable red y datos
Deutsche Gesellschaft für
Technische Zusammenarbeit (GTZ) GmbH
Programa de Apoyo a la Gestión Pública Descentralizada y
Lucha Contra La Pobreza - PADEP
Av. Sánchez Lima 2226
On Oct 16, 2008, at 2:36 AM, [EMAIL PROTECTED]
wrote:
> I want to call an extension like 8 and invoke an external C
> program upon
> calling, pass an constant integer like 1 to the C program.
>
> What I have done is:
>
> /etc/extensions.conf:
> exten => 8,1,system(/usr/local/src/parall
On 16 Oct 2008, at 14:57, jonathan augenstine wrote:
> I am trying to build app_confcall and it is failing. Are there
> known build issues with this module. I am running Asterisk 1.6.0-
> beta9.
Ah yes. 'failing'. I bet that is all it says eh? its not like
compilers give descriptive errors
Hi
> Hi Everybody, I'm having a little problem with asterisk CLI, after the
> version 1.4.19 I'm not been able to see the CLI with colors anymore. I
> have a ubuntu box with asterisk 1.4.21 installed and I don't know how to
> enable the colors again. Of course I have the variable $TERM set to
> xt
I am trying to build app_confcall and it is failing. Are there known build
issues with this module. I am running Asterisk 1.6.0-beta9.
Jonathan
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asterisk-users mailing list
To UN
If you want to wow them with GUI stuff (and make it easy for you,
since the settings are generally correct out of the box) then download
and install EVB (Easy Box Box). Another FreePBX/Asterisk based GUI
with Webmin and lots of other good programs pre-installed. I would
not use it for a very larg
I think we wont see them in a long time, there was I bug isn't it ???
2008/10/16 Lucas Alvarez <[EMAIL PROTECTED]>
> Hi Everybody, I'm having a little problem with asterisk CLI, after the
> version 1.4.19 I'm not been able to see the CLI with colors anymore. I
> have a ubuntu box with asteris
Hi Everybody, I'm having a little problem with asterisk CLI, after the
version 1.4.19 I'm not been able to see the CLI with colors anymore. I
have a ubuntu box with asterisk 1.4.21 installed and I don't know how to
enable the colors again. Of course I have the variable $TERM set to
xterm-co
Am Mittwoch, den 15.10.2008, 21:03 -0400 schrieb C F:
> On Mon, Oct 13, 2008 at 11:54 PM, Jorge Mendoza <[EMAIL PROTECTED]> wrote:
> > Rodolfo Alcazar Portillo wrote:
> >> Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in
> >> a new office of ours: Panasonic or Asterisk. Aster
>
>> ** System Speed Dial on the display updated by the PBX
>
> This one is interesting.
> I can't see a way to do it.
> Ant idea ?
>
P-Asserted Identity ?
Most Business SIP phones support it.
At the moment, I think that Asterisk wouldn't update caller's phone screen
but hopefully, it should be on
Hi,
What is the difference between followme and substitution features ?
I would say both are the same.
Regards
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2008/10/16 C F <[EMAIL PROTECTED]>
>
> * Live call screening - Yes there is a hack that can do it, but it's a
> hell of a hack.
> * Phones that can do most of the usefull features supported by the PBX
> for a reasonable price with LED buttons, including the following
> features:
> ** Call recordin
(Im' answering cc the list, so the knowledge keeps there, and maybe some more
qualified
answers become).
Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario:
> Hey Rodolfo... Need some help from you ...
> I need to know what hardware do I need to make SIP calls if I set-up
>
Hello,
Does anybody use asterisk app mysql with stored procedure? I found article
here :
http://asteriskworld.ru/wiki/AsteriskAppMysql
But the patches are old ... maybe somebody has new patches , or the new
version of asterisk-addons have no problems with mysql stored procedures.
Thanks
--
Pa
Am Mittwoch, den 15.10.2008, 20:51 -0400 schrieb C F:
> Being a Panasonic dealer and having more than 50 Asterisk system in
> production, I can tell you that if this is your first Asterisk
> project, then go with Panasonic, you'll safe yourself lots of
> aggravation and have a happier customer.
Yo
hi everybody
This is Yavuz YILDIRIM
I am software developer.I have a some problems in asterisk.
I am using mysql db. Realtime using asterisk modules. On db i am using
calling hundred fields for use dial.
But i don't know how i can automaticly dial this fields on records
numbers. Who can help
2008/10/16 Steve Totaro <[EMAIL PROTECTED]>
> On Thu, Oct 16, 2008 at 2:57 AM, Olivier <[EMAIL PROTECTED]> wrote:
> > Hi,
> >
> > Using Asterisk 1.4, I would like to send a couple of SIP notifies or
> various
> > scripts whenever Asterisk restart.
> > My concern is also deal with restart from CLI
On 14 Oct 2008, at 18:05, Christian Victor wrote:
> Steven Howes schrieb:
>> Have created a system that involves using call files in the outgoing
>> spool folder. On some occasions it retries which is fine is there
>> any way to view calls waiting retries from the CLI? Using 1.4 btw.
>> Have go
2008/10/16 Torbjörn Abrahamsson <[EMAIL PROTECTED]>
> You could use #exec statements in one of your config-files.
Could you elaborate ?
Which of /etc/asterisk files are thinking of ?
> This would
> mean that they would be run on every reload. As you asked about running
> at every restart, not
2008/10/16 Tzafrir Cohen <[EMAIL PROTECTED]>
> On Thu, Oct 16, 2008 at 08:57:48AM +0200, Olivier wrote:
> > Hi,
> >
> > Using Asterisk 1.4, I would like to send a couple of SIP notifies or
> various
> > scripts whenever Asterisk restart.
> > My concern is also deal with restart from CLI but I don'
Dear All,
I have the following scenario:
My customer dial a DID number and it'll be forwarded to my asterisk server
by the below trunk defined in sip.conf:
[sip_proxy1]
type=peer
context=stations
host=81.201.82.112
disallow=all
allow=g729
allow=alaw
allow=ulaw
dtmfmode=RFC2833
relaxdtmf=yes
canre
I all, I'm trying to transfer a iax2 channel trought dialplan app
transfer to another extensions (IAX).
The variable TRANSFERSTATUS report SUCCESS but the call isn't trasfered.
I haven't other information, in console I see only hangup of a channel.
My scenario is 3 asterisk box connected with iax
On Thu, Oct 16, 2008 at 08:57:48AM +0200, Olivier wrote:
> Hi,
>
> Using Asterisk 1.4, I would like to send a couple of SIP notifies or various
> scripts whenever Asterisk restart.
> My concern is also deal with restart from CLI but I don't how Asterisk is
> restarted when using CLI.
What do you
On Wed, Oct 15, 2008 at 09:38:17PM +0200, Freddi Hansen wrote:
> Hi,
> I started to get some Zaptel compile errors after a 'make update'
>
> I did a clean zaptel install with:
>
> svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel
'make update' again, and it should be gone.
--
> Also i would suggest enabling full log, as it's one place you can see
> everything. Then use grep to search for realtime messages. Your
> logger.conf should already have commented line:
>
> full => notice,warning,error,debug,verbose
>
Yes, I did that.
> # tail -fn0 /var/log/asterisk/full | grep
You could use #exec statements in one of your config-files. This would
mean that they would be run on every reload. As you asked about running
at every restart, not reload, you would have to check if it is indeed a
restart. This could possibly be done by checking the uptime with
"asterisk -rx '
On Thu, Oct 16, 2008 at 2:57 AM, Olivier <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Using Asterisk 1.4, I would like to send a couple of SIP notifies or various
> scripts whenever Asterisk restart.
> My concern is also deal with restart from CLI but I don't how Asterisk is
> restarted when using CLI.
>
>
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