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===YXN0ZXJpc2stdXNlcnNAbGlzdHMuZGlnaXVtLmNvbQ==.-Sunkara RaviPrakashPlease note: This message was sent to you by a user at Indyarocks.com. Click
here in
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 gone
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/@databasetable' under the context
name declaration. This works fine as long as we are adding extensions
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
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
** Call Fwd by PBX with LED indication (not phone based callfwd which
sucks).
Some IP phones support this
Which ones?
With Thomson ST2030, using telnet, you can for instance :
- check current forwarding status (is it forwarded ? toward which number ?),
- and change forwarding
Tzafrir Cohen wrote:
; Allow inband audio (progress) when a call is RELEASEd by the far
; end of a PRI
;
;inbanddisconnect=yes
What does this mean about the default value?
The default value is 'no', to make the behavior be the same as previous
versions of libpri/chan_zap/chan_dahdi.
--
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
Hi,
2008/10/16 Torbjörn Abrahamsson [EMAIL PROTECTED]
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
Is anyone seeing it in 1.4.22? I plan to upgrade to that version, my fear is
zaptel compatibility.
On Fri, Oct 17, 2008 at 1:08 AM, Yehavi Bourvine
[EMAIL PROTECTED]wrote:
unfortunately I still see it in 1.6.0...
__Yehavi:
2008/10/17 broadband Voice [EMAIL PROTECTED]
Hi Duncan,
yes I have a tftp server (I use also Cisco 7941G that use tftp server for
upload configuration) and I know this function, but now my problem is that
the phone is stopped on the initial screen that show 'upgrading' and MAC
address and the process not continued.
Thanks.
--
On Fri, Oct 17, 2008 at 01:24:35AM -0400, Juan Rodríguez wrote:
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
Hi Salvatore
Have you checked the tftp logs in any event? Its important to check the
tftp logs and see if anything is being requested.
I have had this before but usually its still trying to grab its first
couple of files, and from that you can get an idea of where its getting
stuck. If it
manager.conf seems to be read whenever Asterisk restarts (in 1.4).
As at the moment, my requirements are to reboot a couple of hardphones, I
think an #exec statement in manager.conf, plus a script that would wait a
bit before rebooting phones should fit the bill.
Thanks for tip, again.
On Fri, Oct 17, 2008 at 01:48:09PM +0200, Olivier wrote:
manager.conf seems to be read whenever Asterisk restarts (in 1.4).
As at the moment, my requirements are to reboot a couple of hardphones, I
think an #exec statement in manager.conf, plus a script that would wait a
bit before rebooting
Am Freitag, den 17.10.2008, 13:22 +0800 schrieb Cindy Tan:
HI this is cindy... i am still a student... i want to learn more
things about asterisk from you. can i ask you something?
Yes. CC to the list, expecting qualified answers :)
actually, i am thinking how live messager can works on
You are argueing with things like I can do it with panasonic, but it's
not documented anywhere, documentation is a mess but not poor, sorry
for underestimate your abilities. Sorry but I do [completely
understanding Panasonic PBXs]. Not technical. Worst even: that's the
propietary software culture.
At 5:46 PM +0200 2008/10/16, Olivier wrote:
Is Incomplete() application an acceptable work around for ISN ?
It is impossible to determine the full sequence of digits for an ISN
number ahead of time (well, I shouldn't say impossible because one
could create a really nasty hack...) because the
On Fri, Oct 17, 2008 at 4:17 AM, Freddi Hansen [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.
Given the following SRV records:
_sip._udp.tollfree.sip-happens.com. 38400 IN SRV 10 0 5060
sometimes.sip-happens.com.
_sip._udp.tollfree.sip-happens.com. 38400 IN SRV 20 0 5070 ares.sip-happens.com.
Why is asterisk (1.4.17) not honouring the priority and not failing over
to using other records
--
jand. more than just a group
Asterisk AGI Command GET DATA is usually of this form
GET DATA timeout max_digits
When I execute this command, I get only a single digit, regardless of
what the value of max_digits is,
Also the script quits Immediately after the press of the digit
regardless of
In 1.4 manager.conf is parsed on every manager connection, right?
I wouldn't swear at all ...
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Hi,
Whenever Asterisk restarts or reboots, what should happen to ongoing
subscriptions (MWI, Dialogs, ...) ?
Should hardphones discover by themselves Asterisk has restarted so phones
should renew Subscriptions or shall Asterisk send a Notify or another SIP
message telling phones something special
Dear All,
i have the following context defines in etensions.conf:
[a2billing]
exten = _X.,1,Gotoif($[${EXTEN} = 111] ? 21)
exten = _X.,2,DeadAGI,a2billing.php
exten = _X.,3,Wait,2
exten = _X.,4,Hangup
exten = _X.,21,Playback(AR_GetGiveToID)
exten = _X.,22,Wait(2)
exten =
On Friday 17 October 2008 01:09:18 Lee, John (Sydney) wrote:
So, at this stage, my res_config_mysql.c is still not writing anything
into table queue_log despite having: a) correct res_mysql.conf b)
extconfig.conf c) mysql up and running d) res_config_mysql.c start up
okay
I believe that it
Hello,
are there any plans in including SRTP into Asterisk?
The patches in http://bugs.digium.com/view.php?id=5413 are pretty old
and do not work with asterisk 1.6.0.
Thanks, Artem
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2008/10/17 John Todd [EMAIL PROTECTED]
At 5:46 PM +0200 2008/10/16, Olivier wrote:
Is Incomplete() application an acceptable work around for ISN ?
It is impossible to determine the full sequence of digits for an ISN
number ahead of time (well, I shouldn't say impossible because one
exten = _+X.,1,Goto(${EXTEN:1},1)
michel freiha wrote:
Dear All,
i have the following context defines in etensions.conf:
[a2billing]
exten = _X.,1,Gotoif($[${EXTEN} = 111] ? 21)
exten = _X.,2,DeadAGI,a2billing.php
exten = _X.,3,Wait,2
exten = _X.,4,Hangup
exten =
Artem Makhutov wrote:
are there any plans in including SRTP into Asterisk?
Yes.
The patches in http://bugs.digium.com/view.php?id=5413 are pretty old
and do not work with asterisk 1.6.0.
Correct. There is still work to be done, but it's getting much higher
on our list of things that need
Dear All,
I tried to put + before the x like _+X but when making a call i got the
following error:
[Oct 17 15:08:58] WARNING[17532]: ast_expr2.fl:407 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected '+', expecting $end;
Input:
+9613089187 = 111
Regards
On Fri,
Brian J. Murrell wrote:
Given the following SRV records:
_sip._udp.tollfree.sip-happens.com. 38400 IN SRV 10 0 5060
sometimes.sip-happens.com.
_sip._udp.tollfree.sip-happens.com. 38400 IN SRV 20 0 5070
ares.sip-happens.com.
Why is asterisk (1.4.17) not honouring the priority and not failing
On Fri, Oct 17, 2008 at 9:29 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
I have mutihomed boxen on many different networks as well, this has
never been an issue.
Let's put aside why would you or there is no reason, and then think
about it again. Let's just say you wanted two NICs on the same
On Fri, 2008-10-17 at 10:32 -0500, Andres wrote:
Because Asterisk does not support that.
Which is just another way of saying Asterisk is broken then. SRV
records have requirements for their correct use. If those requirements
are ignored, that is a broken implementation.
The only thing that
There are a few options.
He should probably start on the wiki.
http://www.voip-info.org/wiki/view/Predictive+dialer
Steve Totaro wrote:
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
I started this at 4pm yesterday, its 10am and the handsets still say they are
in progress?
Is that normal?
Thanks!
jlc
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On Friday 17 October 2008 10:32:30 michel freiha wrote:
I tried to put + before the x like _+X but when making a call i got the
following error:
[Oct 17 15:08:58] WARNING[17532]: ast_expr2.fl:407 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected '+', expecting
It should be fairly easy to write an AGI script that does the SRV query,
do whatever you want with the response, set a channel variable with
the results and use that in your dialplan.
Brian J. Murrell wrote:
On Fri, 2008-10-17 at 10:32 -0500, Andres wrote:
Because Asterisk does not support
On Fri, 2008-10-17 at 11:18 -0500, Eric ManxPower Wieling wrote:
It should be fairly easy to write an AGI script that does the SRV query,
do whatever you want with the response, set a channel variable with
the results and use that in your dialplan.
Maybe. If I were an AGI hacker. But
On Fri, Oct 17, 2008 at 8:02 AM, Rodolfo Alcazar Portillo
[EMAIL PROTECTED] wrote:
You are argueing with things like I can do it with panasonic, but it's
not documented anywhere, documentation is a mess but not poor, sorry
for underestimate your abilities. Sorry but I do [completely
Brian J. Murrell wrote:
On Fri, 2008-10-17 at 11:18 -0500, Eric ManxPower Wieling wrote:
It should be fairly easy to write an AGI script that does the SRV query,
do whatever you want with the response, set a channel variable with
the results and use that in your dialplan.
Maybe. If I
On Friday 17 October 2008 10:15:22 Olivier wrote:
2008/10/17 John Todd [EMAIL PROTECTED]
At 5:46 PM +0200 2008/10/16, Olivier wrote:
Is Incomplete() application an acceptable work around for ISN ?
It is impossible to determine the full sequence of digits for an ISN
number ahead of time
On Fri, 2008-10-17 at 11:35 -0500, Eric ManxPower Wieling wrote:
If you fight Asterisk's oddities then you will have a depressing and
miserable life. If you embrace Asterisk's oddities then you will have a
joyous and enlightened life. 8-)
I just want something that works. :-)
I agree
The wiki says it should take about 20 minutes per handset.
Joseph L. Casale wrote:
I started this at 4pm yesterday, its 10am and the handsets still say
they are in progress?
Is that normal?
Thanks!
jlc
Yes, the problem is there as well...
__Yehavi:
2008/10/17 broadband Voice [EMAIL PROTECTED]
Is anyone seeing it in 1.4.22? I plan to upgrade to that version, my fear
is zaptel compatibility.
On Fri, Oct 17, 2008 at 1:08 AM, Yehavi Bourvine
[EMAIL PROTECTED] wrote:
On Friday 17 October 2008 11:46:09 Brian J. Murrell wrote:
On Fri, 2008-10-17 at 11:35 -0500, Eric ManxPower Wieling wrote:
If you fight Asterisk's oddities then you will have a depressing and
miserable life. If you embrace Asterisk's oddities then you will have a
joyous and enlightened
My guess is that if you had two NICs on the same subnet with
different IPs the kernel route table and ARP cache would get pretty
confused. This seems so incredibly broken to me I've never tried
That was my guess and point to begin with. I was not aware of or thinking
about bonding.
The wiki says it should take about 20 minutes per handset.
yeah I just found that, and so I called tech support
and they said to reset the gateway, and if needed to pull
the battery out of the phones and power them on. I have done
this and they restarted the firmware download so I will wait
and
Our M3's also got stuck somewhere in the middle, so we rebooted the base by
pressing the reset button. After this the handset moved along and continued
with the upgrade. This happended on all our 5 handsets, connected to three
different bases, one with fw 1.01 and two with 1.07, upgrading to 1.16.
On Fri, 2008-10-17 at 12:11 -0500, Tilghman Lesher wrote:
Have you considered upgrading to 1.6?
Not to this point, no. 1.4 does everything I want and if it ain't
broke, don't fix it. Well, now it's broke I guess. Still, Ubuntu still
uses 1.4 and I don't like having to maintain my own
I do, I am planning to have little more than 1000. Right now I had
managed little more than 700 SIP channels + 100 IAX channels.
Do you think this can cause any problem?? --I mean, having this RTP
ports range--
Tzafrir Cohen wrote:
On Fri, Oct 17, 2008 at 01:24:35AM -0400, Juan Rodríguez
On Fri, Oct 17, 2008 at 03:11:17PM -0400, Juan E. Rodríguez wrote:
I do, I am planning to have little more than 1000. Right now I had
managed little more than 700 SIP channels + 100 IAX channels.
Do you think this can cause any problem?? --I mean, having this RTP
ports range--
If you
On Thu, 16 Oct 2008, GNUbie wrote:
Hello,
On Tue, Oct 14, 2008 at 12:07 AM, Jeff LaCoursiere [EMAIL PROTECTED] wrote:
A packet trace will probably show exactly what is happening. Try:
tcpdump -nlXs 8192 -i eth0 port 5060
You should be able to see the SIP information going back
Here is my hardware configuration
TELCO --- PRI1 --- PBX --- PRI2 --- Asterisk
The PBX is a Siemens Hicom 200 EX (Model 80)
We are connecting between the PBX and Asterisk using QSIG switch type.
What I want to do is the following:
1. Call comes from TELCO via PRI1 and enters PBX
2. PBX Routes
I am using Asterisk and an X101P card as a glorified answering machine.
We have a residential PSTN line with about six phones connected to it.
Like an answering machine, I want Asterisk answer the line *only* when
an incoming call is not answered after four rings.
This mostly works. My
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jack Bates
Sent: Friday, October 17, 2008 4:48 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] anoyingly answers already in use pstn line
I am using Asterisk and an
GNUbie wrote:
What particular configs are you looking for? Below is my current setup
and scenario:
[snom] ==LAN== [asterisk] ==FXO/POTS == [analog_telephone/mobile_phone]
SNOM is using the 192.168.101.102 IP address
Asterisk is using 192.168.101.1 IP address for its eth1 interface
FXO port
Here is my hardware configuration
TELCO --- PRI1 --- PBX --- PRI2 --- Asterisk
The PBX is a Siemens Hicom 200 EX (Model 80)
We are connecting between the PBX and Asterisk using QSIG switch type.
What I want to do is the following:
1. Call comes from TELCO via PRI1 and enters PBX
2. PBX Routes
On Fri, Oct 17, 2008 at 5:24 PM, Ron Joffe [EMAIL PROTECTED] wrote:
Here is my hardware configuration
TELCO --- PRI1 --- PBX --- PRI2 --- Asterisk
The PBX is a Siemens Hicom 200 EX (Model 80)
We are connecting between the PBX and Asterisk using QSIG switch type.
What I want to do is the
On Fri, Oct 17, 2008 at 05:04:32PM -0400, Gleim, Jason wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jack Bates
Sent: Friday, October 17, 2008 4:48 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
On Oct 17, 2008, at 5:14 PM, Paul Douglas Franklin wrote:
When off site, our IP phones lose contact after a few minutes of
inactivity. They no longer receive calls, though they can call out.
Asterisk acts as if it is ringing the phone, but the phone does not
ring.
The phones are behind a
On Wed, Oct 15, 2008 at 7:57 PM, Stephen Reese [EMAIL PROTECTED] wrote:
I've searched around and found a few similar situations where the
phone will call out when using a Asterisk server but not receive
inbound calls. My issue is a little stranger. If I call out from the
phone then the phone
It is likely a NAT timeout issue. When you call outbound, you
'reactivate' the SIP session in your NAT device, allowing calls to come
in until it expires (default on many devices is 60 seconds). You may
also receive inbound calls when the phone reregisters regularly. Try
'qualify=yes' in your
Sorry, I missed the Cisco router bit.
As a last resort (if qualify doesn't help), you could enter this
(global) to increase the timeout on UDP translations:
ip nat translation udp-timeout 300 (or greater if you prefer)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
I am running Asterisk and would like to add SER to register my (sip) DID and
connect it to asterisk;
but I'm not sure if this is the correct forum.
I have as DID, sip account with one VoIP provider; currently Im using just
stand alone SIP phone and register with the VoIP provider via:
On Friday 17 October 2008 17:38, Steve Totaro wrote:
I would engineer the system so that Asterisk is in the middle rather
than the far end. Is there a reason why you don't want to or cannot
do that?
TELCO --- PRI1 --- Asterisk --- PRI2 --- PBX
Steve,
I have also done this same method in
SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do.
On Fri, October 17, 2008 9:36 pm, Joseph wrote:
I am running Asterisk and would like to add SER to register my (sip) DID
and connect it to asterisk;
but I'm not sure if this is the correct forum.
I have as DID, sip
On 10/17/08, Alex Balashov [EMAIL PROTECTED] wrote:
SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do.
Slight clarification: Kamailio (formerly OpenSER) / OpenSIPS is the
thing to do.
--
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
I'm using Gentoo and the only package I was able to find in portage was SER;
I could compile manually but it is harder to upgrade and keep track of
dependencies.
--
#Joseph
On 10/17/08 22:42, Alex Balashov wrote:
SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do.
On
On 10/17/08 23:23, Kristian Kielhofner wrote:
On 10/17/08, Alex Balashov [EMAIL PROTECTED] wrote:
SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do.
Slight clarification: Kamailio (formerly OpenSER) / OpenSIPS is the
thing to do.
I would gladly go with any of the
As far as I'm aware SER (and it's derivatives) cannot initiate
outbound registraitions. They can do the opposite and act as a SIP
Registrar. For outbound registrations you should be able to use
Asterisk.
Regards,
Greyman.
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On Sat, Oct 18, 2008 at 9:20 AM, Joseph [EMAIL PROTECTED] wrote:
On 10/17/08 23:23, Kristian Kielhofner wrote:
On 10/17/08, Alex Balashov [EMAIL PROTECTED] wrote:
SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to
do.
Slight clarification: Kamailio (formerly
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