I dont think this is existed.
However, its easy to build a script in php or perl or any other language
which check time from file or database and generate call file which execute
paging in asterisk. Just put this script in cron. Thats it...
Regards,
Bharat Lalcheta
On Thu, Dec 27, 2012 at
I disabled all logger channels but still it logs to /var/log/messages.
Any hints?
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New to Asterisk? Join us for a live introductory webinar every Thurs:
please refer logger.conf under /etc/asterisk
and stop messages log for full.
On Thu, Dec 27, 2012 at 2:43 PM, [Digital^Dude] ®
millennium@gmail.comwrote:
I disabled all logger channels but still it logs to /var/log/messages.
Any hints?
--
I'm trying to get T.38 passthrough working on Asterisk 1.8.18.1. It isn't
working. Calls go through and are answered, but the fax machines are unable
to communicate. I checked the value of CHANNEL(t38passthrough) and it seems
to always be 0. One side is Level 3 T.38 TN and the other side
Hi!
I am installing asterisk with my ISP but he give me a Cisco 887M router to use
for SIP conection. My problem is that I dont know how to link Asterisk with
this device because I dont have user/pass to use.
Anybody has a cluee to use CISCO 887M with Asterisk ?
Thks!
On Thu, Dec 27, 2012 at 9:10 AM, Edwin Quijada
listas_quij...@hotmail.comwrote:
Hi!
I am installing asterisk with my ISP but he give me a Cisco 887M router to
use for SIP conection. My problem is that I dont know how to link Asterisk
with this device because I dont have user/pass to use.
Shouldn't be difficult. You're just setting up the Cisco box as a SIP
gateway. Here's a link to get you started.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/b06gtw
ay.html
From: asterisk-users-boun...@lists.digium.com
If it is writing to /v/l/m, then it is coming from somewhere else. All
Asterisk messages go to /v/l/asterisk.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Thursday, December 27, 2012 3:32 AM
To: Asterisk
I would set up 5 shell files called pray1.sh, pray2.sh, etc and then set up
5 entries in /etc/crontab to run them at the specified time daily. The file
pray1.sh should look something like this:
#!/bin/sh
cp /pray1/*.call /tmp
mv /tmp/*.call /var/spool/asterisk/outgoing
the entry in
Setting directmedia=no does not help. The calls still go through and the fax
still fails after switching to T.38.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Thursday, December 27,
Thanks Matt. The suggestion helped. No more slip erros.
Dave
Original Message
Subject: Re: [asterisk-users] dahdi timing source multiple cards
From: Matthew Fredrickson cres...@digium.com
Date: Fri, December 21, 2012 3:41 pm
To: asterisk-users@lists.digium.com
You
On Thu, 27 Dec 2012, [Digital^Dude] ® wrote:
I disabled all logger channels but still it logs to /var/log/messages.
Any hints?
What version of Asterisk?
What does 'logger show channels' show?
Any chance your startup script pipes output through the logger shell
command?
--
Thanks in
We are offering $100 (paid via paypal or check) to the first person who assists
us in successfully sending and receiving faxes in the setup described below.
Offer expires Dec 31. We are a direct customer of Level 3, there is no other
carrier involved.
What we want to work:
Level 3 T.38
Have you configured the canreinvite=yes in sip peer?
I am currently off work for two days, but a 100% fail means a configuration
problem for sure.
Leandro
2012/12/27 Eric Wieling ewiel...@nyigc.com
We are offering $100 (paid via paypal or check) to the first person who
assists us in
We have set directmedia=yes as well as directmedia=no. There is no NAT
involved.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini
Sent: Thursday, December 27, 2012 1:08 PM
To: Asterisk Users
directrtpsetup=yes in sip.conf?
On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling ewiel...@nyigc.com wrote:
We have set directmedia=yes as well as directmedia=no. There is no NAT
involved.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
We have directrtpsetup=no because the comments in the sample config indicates
it does not work in all situations.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher
Harrington
Sent: Thursday,
True, but it should bypass Asterisk when possible for SIP streams and may
solve your problem.
On Thu, Dec 27, 2012 at 12:16 PM, Eric Wieling ewiel...@nyigc.com wrote:
We have directrtpsetup=no because the comments in the sample config
indicates it does not work in all situations.
I am using voiceglue to record voice.
VXML :
record name=rec finalsilence=4s maxtime=30s beep=true
dtmfterm=true
filled
submit enctype=*multipart*/*form*-*data* next=http://domain/getVxml/
namelist=rec method=post
/filled
/record
I can take rec parameter but it is not file. rec is
It does not appear to make any difference. Calls are still failing.
-Original Message-
From: Christopher Harrington [mailto:ch...@acsdi.com]
Sent: Thursday, December 27, 2012 1:20 PM
To: Eric Wieling
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
If VoiceGlue is the software making the recording, then that's where you
need to look for support.
Try https://github.com/voiceglue/voiceglue/issues and
http://www.voiceglue.org/mailing-list/ .
On Thu, Dec 27, 2012 at 12:27 PM, ulvi cesur uce...@gmail.com wrote:
I am using voiceglue to record
Last thing to check, just for sanity's sake:
t38pt_udptl=yes in sip.conf? It defaults to off.
On Thu, Dec 27, 2012 at 12:32 PM, Eric Wieling ewiel...@nyigc.com wrote:
It does not appear to make any difference. Calls are still failing.
-Original Message-
From: Christopher
We are using t38pt_udptl=yes,redundancy,maxdatagram=400 Without the
maxdatagram we get errors in the CLI. We also tried using FEC instead of
redundancy, but no difference.
-Original Message-
From: Christopher Harrington [mailto:ch...@acsdi.com]
Sent: Thursday, December 27, 2012 2:23
This past holiday weekend has resulted in some real groaners when it
comes to bugs in our dialplan, making obvious the need for some changes
in our procedures.
First, our hours of operation for Christmas Eve, Christmas, Boxing Day
and New Year's Eve had changed with little to no notice. Okay,
Ernie Dunbar wrote:
This appears to boil down to always remember to test it at the time
that it becomes relevant. But if I was the kind of person who always
remembered to do things at the right time, then there would never be a
need for computers to do jobs like this in the first place.
I
The simplest way to address this kind of change is to test it a week
(month) or so in advance on your test machine (we have VM's set up to mock
our live machines). A protection against last minute changes is to have
this kind of thing controlled by variables so you can possibly even avoid
I'm a fan of your method. I haven't had good luck with GotoIfTime in the
past.
A lot of my dialplan is actually handled by an AGI script. I've always
found that to be the easiest.
On Thu, Dec 27, 2012 at 2:00 PM, Doug Lytle supp...@drdos.info wrote:
Ernie Dunbar wrote:
This appears to boil
We bypass this problem by storing business hours and holidays in a
database table. We use an ODBC call to return whether or not to play
the day or night greeting based on the database. We also store the
name of a custom greeting file to play.
The database is fairly easy to manipulate with
I would say that the database method has the advantage over GotoIfTime in
that it should stay the same between releases. More headache on the front
end, but easier once it is up and running.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On Thu, Dec 27, 2012 at 12:46 PM, Ernie Dunbar maill...@lightspeed.cawrote:
This past holiday weekend has resulted in some real groaners when it comes
to bugs in our dialplan, making obvious the need for some changes in our
procedures.
First, our hours of operation for Christmas Eve,
Do you have reinvite allowed? That was an issue on one of my
installations if I am remembering correctly. Any debug, logs, confs
that would help?
Thanks,
Steve Totaro
On Thu, Dec 27, 2012 at 12:15 PM, Eric Wieling ewiel...@nyigc.com wrote:
Setting directmedia=no does not help. The calls
On 28/12/2012 1:55 AM, Eric Wieling wrote:
We are offering $100 (paid via paypal or check) to the first person who assists
us in successfully sending and receiving faxes in the setup described below.
Offer expires Dec 31. We are a direct customer of Level 3, there is no other
carrier
On 28/12/2012 4:59 AM, Larry Moore wrote:
On 28/12/2012 1:55 AM, Eric Wieling wrote:
.
snip
.
directmedia=no
t38pt_udptl=no
snip
Hmm, the t38pt_udptl will need to be set to yes, this was set to no for
non T.38 capable devices
I had set faxdetect=no in the peer's configuration for the
#1 I assume you have spandsp installed
#2 I'm guessing you got some hints from this thread -
https://issues.asterisk.org/jira/browse/ASTERISK-18394 ?
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
I was not aware you needed SpanDSP for T.38 passthrough.. How will that work
with the UDPTL packets not going through Asterisk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday,
Not certain that you actually do. I do know that T.38 can be a dental
experience with Asterisk, but some folks have succeeded with it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent:
Friends,
Curious if others have run into this scenario, and can shed further light
on it. I am working with an installed base of systems using PRI circuits
from several carriers, and the symptoms I relate occur across the board.
Most carriers are restricting CALLING Number ID to be one of the
udptl.conf settings:
[general]
udptlstart=4000
udptlend=4998
udptlchecksums=no
udptlfecentries = 3
udptlfecspan = 3
use_even_ports = yes
T38FaxUdpEC = t38UDPRedundancy
T38FaxMaxDatagram = 400
sip.conf settings:
directmedia=yes
faxdetect = no
t38pt_udptl=yes,redundancy,maxdatagram=400
On Thu, Dec 27, 2012 at 2:41 PM, Barry D. Hassler
barry.hass...@gmail.comwrote:
Friends,
Curious if others have run into this scenario, and can shed further light
on it. I am working with an installed base of systems using PRI circuits
from several carriers, and the symptoms I relate occur
On Thu, Dec 27, 2012 at 3:45 PM, Eric Wieling ewiel...@nyigc.com wrote:
sip.conf settings:
directmedia=yes
I know you've said you tried it both ways, but consensus seems to be that
directmedia needs to be =no when using UDPTL.
--
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone:
I usually set directmedia=yes with good results...
Leandro
2012/12/27 Christopher Harrington ch...@acsdi.com
On Thu, Dec 27, 2012 at 3:45 PM, Eric Wieling ewiel...@nyigc.com wrote:
sip.conf settings:
directmedia=yes
I know you've said you tried it both ways, but consensus seems to be
On 28/12/2012 5:45 AM, Eric Wieling wrote:
udptl.conf settings:
[general]
udptlstart=4000
udptlend=4998
udptlchecksums=no
udptlfecentries = 3
udptlfecspan = 3
use_even_ports = yes
T38FaxUdpEC = t38UDPRedundancy
T38FaxMaxDatagram = 400
sip.conf settings:
directmedia=yes
faxdetect = no
We process a lot of t.38 with Level 3 and our Asterisk servers. We do not
prefer Adtran end points for t.38.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
From: Danny Nicholas da...@debsinc.com
Sent: Thursday, December 27, 2012 4:51
Hello,
I'm trying to connect a Cisco AS5300 has Asterisk, but I have a problem.
Sound from POTS - Asterisk does not work. (In the sense Asterisk - POTS
it works!!)
The problem lies in two directions (call initiated from the Asterisk or
POTS)
I have no firewall between Asterisk and Cisco. (it's a
Thanks a lot for your kindly reply and help.
Really I did not understand why you need to place them in the
/var/spool/asterisk/outgoing?
Regards
Bilal
---
I would set up 5 shell files called pray1.sh, pray2.sh, etc
and then set up
5 entries in /etc/crontab to run them at the
On 27/12/2012 3:14 PM, Carlos Alvarez wrote:
On Thu, Dec 27, 2012 at 12:46 PM, Ernie Dunbar maill...@lightspeed.ca
mailto:maill...@lightspeed.ca wrote:
This past holiday weekend has resulted in some real groaners when
it comes to bugs in our dialplan, making obvious the need for some
Please don't top-post.
On Thu, Dec 27, 2012 at 1:29 PM, bilal ghayyad bilmar...@yahoo.com
wrote:
How can I have Paging on Asterisk to call for pray?
The pray is 5 times in a day and there is a timing for pray (actually it
can be existed in a text file or database for the next 2 or 5 years).
On 12/27/2012 07:36 PM, Ron Wheeler wrote:
On 27/12/2012 3:14 PM, Carlos Alvarez wrote:
On Thu, Dec 27, 2012 at 12:46 PM, Ernie Dunbar maill...@lightspeed.ca
mailto:maill...@lightspeed.ca wrote:
This past holiday weekend has resulted in some real groaners when
it comes to bugs in our
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