On 05/05/15 17:52, Olivier wrote:
2. From personal experience, would you rate an IMAP migration as an
easy or as a difficult task ?
By IMAP migration, I mean changing from one IMAP software to another,
on the same or on an other box.
There is software called 'imapsync' which will sync mail
IMAP storage is still resilient to short network outages
(without IMAP replication).
Regards
2015-05-06 10:18 GMT+02:00 Gareth Blades
mailinglist+aster...@dns99.co.uk
mailto:mailinglist+aster...@dns99.co.uk:
On 05/05/15 17:52, Olivier wrote:
2. From personal experience, would you
You would need to give more information really.
Your sip.conf file listing the entries for the phones especially which
codecs are permitted.
A copy of the 'asterisk -rvvv' console output when you make the call.
On 27/03/15 17:05, Salaheddine Elharit wrote:
please no body has som with aastra
That is your issue.
You can enable a 'sip debug' and make the call again and get a trace of
the SIP message asterisk is sending to the phone.
We can take a look here to see if anything looks wrong.
If you could post a trace from a phone that can call that destination it
might be easier to
On 12/02/15 12:25, Stefan Viljoen wrote:
Hi all
Sometimes (about every three months) some of my Asterisk 1.8 boxes will
start running this message thousands of times in the CLI:
[Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack:
Call to timerfd_gettime() error: Invalid
On 06/02/15 07:54, Olli Heiskanen wrote:
My goal is to allow my users record their own queue announcements and
choose which announcements they want to use in each queue. I have
several Asterisk servers and a Kamailio server which dispatches call
traffic between the Asterisks. Question is, is
On 05/02/15 10:53, Sebastian Damm wrote:
Hi,
we have quite a few Asterisk machines running and try to keep them on
a current version of the Asterisk 11 branch. But since we upgraded to
11.14.0 a couple weeks ago, we have to restart the Asterisk process
every week because the load gets too
On 05/12/14 16:46, Olli Heiskanen wrote:
INVITE that Asterisk (at port 5070) receives:
PU.BL.IC.IP:5060 PU.BL.IC.IP:5070: SIP, length: 1046
INVITE sip:6...@testers.com
mailto:sip%3a...@testers.com;transport=UDP SIP/2.0
Record-Route: sip:PU.BL.IC.IP;lr=on;ftag=41030177
Via: SIP/2.0/UDP
On 04/11/14 15:11, Pat Collins wrote:
Hello group and thank you for the attention.
I'm using Asterisk 11.12 running on Ubuntu Server 12.04
We have an issue with channels remaining open after a SIP peer
unregisters.
It seems that if the peer goes away before manually hanging up a call,
the
On 30/10/14 19:40, Henry Fernandes wrote:
I’m trying to use Playtones to have a tone played periodically
throughout phone calls. Unfortunately, I can’t seem to get PlayTones
to work. I never hear the audio tones.
Here is the output on the Asterisk console.
-- Executing
In that case the only way I can think of doing it would be to place both
parties into a conference call and have an extension join it which just
plays a tone into the conference every so often.
On 31/10/14 16:40, Henry Fernandes wrote:
Unfortunately, the majority of my customers are using 1.8.
On 30/10/14 13:52, Jonas Kellens wrote:
Hello,
I read on the wiki :
Asterisk 1.8 will allow to read SIP response codes in the dialplan via
*${HASH(SIP_CAUSE,channel-name)}*. Additionally make sure you're
using the destination channel, not the source channel.
But when I use this in my
On 11/08/14 16:46, Farid Fadaie wrote:
Hello,
Full disclosure: my name is Farid Fadaie and I'm in charge of
BitTorrent Bleep (a private P2P SIP-based messaging application in
early alpha)
http://blog.bittorrent.com/2014/07/30/building-an-engine-for-decentralized-communications/
I have
On 15/05/14 16:28, Mike Leddy wrote:
Hi Russ,
I rebooted the machine loading dahdi_dummy in /etc/modules before
the /etc/init.d/dahdi.
Now dahdi_test shows a nearly perfect score:
# dahdi_test
Opened pseudo dahdi interface, measuring accuracy...
99.998% 99.990% 99.998% 99.996% 99.998% 99.998%
You would need to provide more information. Mobiles and landlines are
not SIP and yet you say calls are coming into your asterisk over SIP. So
what or who is doing the translation?
Initial thoughts are that it could be you are sending back SIP/180 with
no session progress and indicating
On 07/03/14 16:52, Johann Steinwendtner wrote:
Sorry, for the stupid question, but what happens if Kamailio fails ?
We have two copies on different servers which make use of keepalived to
provide a virtual IP address between them. We also have them connected
to two databases with
On 20/02/14 11:27, Brynjolfur Thorvardsson wrote:
Hi all
We have an Asterisk server that's been running for a few years now
without problems. We have IPTables running, as well as fail2ban and
have followed all the security recommendations we have found.
Every few weeks we get an attack
On 20/02/14 10:24, Igor Dvorzhak wrote:
Guys,
I am using
Asterisk 1.8.20.0 built by mockbuild @
buildvm-24.phx2.fedoraproject.org
http://buildvm-24.phx2.fedoraproject.org on a x86_64 running Linux
on 2013-01-18 19:52:25 UTC
How can I set variable in one context and then Redirect a channel
On 20/02/14 17:16, Paul Belanger wrote:
On Thu, Feb 20, 2014 at 10:40 AM, Tony Mountifieldt...@softins.co.uk wrote:
I haven't been able to find the answer online, and am not currently
able to conduct an experiment to find the answer...
I understand that in a SIP call where G729 has been
On 15/02/14 20:05, Jerry Geis wrote:
I have a confbridge in asterisk 11.
I am using an AGI to bring people in the conf automatically.
I want to speak a pre-recorded wave file message into the conf to
all users.
how might I do that?
Thanks,
Jerry
You could initiate a call which would
On 14/02/14 06:33, Daniel van den Berg wrote:
Hi All,
Lets say I want to setup a queue that will handle inbound calls to
dynamically added agents that are all mobile numbers. Now when I do this
setup it works, it loads the agents dynamically and if the mobile phone
is on and have reception it
On 11/02/14 18:45, Dave Platt wrote:
Just checking the transcoding on our Asterisk boxes and I get the
following results.
I have the g726, ilbc and lpc10 formats and codecs enabled in 'make
menuselect' so I dont understand why its showing as no translation path.
Any ideas?
Are the modules
Just checking the transcoding on our Asterisk boxes and I get the
following results.
I have the g726, ilbc and lpc10 formats and codecs enabled in 'make
menuselect' so I dont understand why its showing as no translation path.
Any ideas?
I am running certified-asterisk-11.2-cert2
Thanks
Gareth
On 04/02/14 18:56, Meadows Hoa wrote:
If SIP channel driver needs to connect to a remote GW over a dedicated
SIP trunk BUT the remote GW has a 'standby' in case of failure, how
can the sip configuration file be configured for the remote GW when
there are actually two IP addresses. If the main
We have a customer reporting poor quality calls when they come to us via
a particular provider. The SIP traces look perfectly normal both on the
ingress to us and egress to another telco. No additional sip messages
after the call has been answered until the BYE is received. However in
the
On 02/02/14 14:42, Markus Reschke wrote:
Hi!
My telco is Deutsche Telekom and they got about 30 SIP servers right
now. Currently I've set up a template for incoming calls in sip.conf
and added each SIP server by it's IP address like this:
[DTAG-in-1](DTAG-in-template)
host=217.0.16.103
...
On 28/01/14 15:01, Jerry Geis wrote:
I have been trying to get a feel for scaling or dimensioning using
asterisk 11.
if I desire to use something like a dell r320, hardware RAID, 2G
E5-2420, 4G RAM
and only SIP trunking using gsm (least bandwidth and no transcoding)
how many calls out can I
On 28/01/14 16:56, Steve McCann wrote:
Hello All,
I've asked this on the asterisk-dev list, so sorry for cross-posting.
So far I'm not sure how to accomplish this without looking at the
source code or looking at some other way to get around this issue.
I'm trying to have an automated call
On 23/01/14 23:37, Marek Cervenka wrote:
can someone confirm that mp3 is unsupported? is patch available?
what about patch for Opus?
uncle google doesnt know
MP3 is only supported for reading not writing. Its a patent issue as
Asterisk cannot distribute the software to write to mp3 under
On 23/01/14 13:38, Ishfaq Malik wrote:
Hi
Is there any way to change the preferred audio playback format in
asterisk (I'm using 1.8.25.0)
i.e. first check for gsm, if doesn't exits then check for slin?
It should pick whichever source format requires the least cpu to
transcode into the
On 23/01/14 15:21, Marek Cervenka wrote:
hi,
which file extensios are supported in mixmonitor application?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor
can i record to Opus?
core show file formats will give you a list of formats your system
supports
On 16/01/14 10:47, Tiago Geada wrote:
Hi folks,
We've been having a weird issue... It is happening more often in the
last few months...
Most inbound calls, we have in our dialplan before Queue():
Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL});
So when the call
2014 14:09, Gareth Blades
mailinglist+aster...@dns99.co.uk
mailto:mailinglist+aster...@dns99.co.uk wrote:
On 16/01/14 10:47, Tiago Geada wrote:
Hi folks,
We've been having a weird issue... It is happening more often in
the last few months...
Most inbound calls, we have
running it on asterisk box? I guess
only port 5060 is not too bad
On 16 January 2014 14:09, Gareth Blades
mailinglist+aster...@dns99.co.uk
mailto:mailinglist+aster...@dns99.co.uk wrote:
On 16/01/14 10:47, Tiago Geada wrote:
Hi folks,
We've been having a weird
On 16/01/14 15:29, Kevin Larsen wrote:
Not to derail the conversation, Gareth, but do you leave this running
full time on your asterisk boxes or just turn it on when you are
trying to track problems?
On average, how far back can you go for looking at problems?
Its normally running full time
On 15/01/14 09:39, Francesco Namuri wrote:
Hello James,
thanks for your answer, I supposed this too, but my provider answered me
that as
m=audio 43718 RTP/AVP 8 18 3 101
^ ^ ^ GSM proposal
^
You need to decide which codecs you are going to allow to be used on the
SIP side. As you are connecting to E1 then the standard codec would be
g711 alaw or ulaw. You could force the SIP side to use the same codec
but it uses about 100Kbps of bandwidth so quite a bit higher than other
codecs
Thats fine for calculating how many users a particular speed network
connection can cope with. 640 concurrent calls on a 100Mbps connection
is doable on a decent machine as long as you are not doing much codec
translation.
Once you get to the point where you start having hundreds of users
We have a machine with a quad core 'Intel(R) Xeon(R) CPU E5-1410 0 @
2.80GHz' running asterisk 11.2-cert with ingress and egress all sip.
Fastagi running as a daemon (written in perl) performing cdr updates at
call start, answer and call end together with a query when a call comes
in to get
On 27/11/13 14:12, James Bensley wrote:
What is the maximum delay RTP will tolerate one way (Does Asterisk
have a limit too)?
Can this be tuned (increased or decreased) within Asterisk (I'm
thinking of DSL customers where we may have this issue between our
PBXs and the customer)?
There isnt
On 19/11/13 16:44, Bas Rijniersce wrote:
Hi,
I just did a test install of Ast 11, and have trouble getting the same
logging information that Ast 1.x provided. I'm looking specifically
for the logging around call progress / dialplan actions.
In ASt 11 I've done the same thing that I did
On 07/11/13 11:20, Ishfaq Malik wrote:
Hi
We are using asterisk 1.8.23.1
We have a script that runs on a minute cron which polls the asterisk
server for 3 bits of information by using
asterisk -rx 'command'
which then gets pushed to a graphite server we have
99% of this runs smoothly.
On 11/10/13 18:43, Tiago Geada wrote:
Hi,
Seems a great workaround from Gareth Blades. Thanks I will try it.
Any way to make asterisk log a line in /var/log/messages ?
I normally have all the verbose output sent to the log file so anything
in the NoOp() line gets logged to the file so thats
On 13/10/13 20:06, CDR wrote:
I am quite surprised about the degree of surprise in the group. A few
days ago, somebody called a school and issued a threat, through my
network. The call came from China, but of course it was US caller. The
DA wants to know where call came from. The caller ID is
On 08/10/13 17:02, Eric Wieling wrote:
I have an Asterisk 1.4 box which is sometimes getting the message below. Here
is the weird part, the CNG is coming from ANOTHER ASTERISK SERVER.
209.220.119.19 is an Asterisk 11 box.
[Oct 8 11:59:27] NOTICE[20798]: rtp.c:849 process_rfc3389: Comfort
On 02/10/13 12:17, Johan Wilfer wrote:
Hi,
I have some servers that are dedicated to do meetme conferencing. From
some previous test i concluded that I need to use dahdi_dummy as it is
more accurate.
If I did use the core timers in dahdi (not loading dahdi_dummy) I got
bad quality in the
On 02/10/13 16:13, gincantalupo wrote:
Hi Garet,
ok but since the messages contain my own public IP with this method
I'm banning my public IP not the real attacker IP. Am I wrong?
Giorgio
No the asterisk dialplan entry is pulling the IP address out of the SIP
Contact: header which in the
On 01/10/13 15:44, gincantalupo wrote:
On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo
gincantal...@fgasoftware.com mailto:gincantal...@fgasoftware.com
wrote:
Hi,
I get a lot of these messages on my Asterisk CLI:
Failed to authenticate user
On 26/09/13 16:43, Rusty Newton wrote:
On Thu, Sep 26, 2013 at 10:08 AM, Gareth Blades
mailinglist+aster...@dns99.co.uk wrote:
On 26/09/13 14:59, Rusty Newton wrote:
Try the following:
extension = 6001,1,Set(CHANNEL(tonezone)=us)
same = n,Dial(SIP/6001,,r(ring))
The argument passed
We have an issue with a customer where when calls are sent to one of their
offices as soon as the call is answered the call fails.
We are performing remote bridging and switching the audio from the server which
initiated the call to our switch which is on the same network.
After the call is
On 27/09/13 14:15, Gareth Blades wrote:
Can anyone with a bit more knowledge about the SDP standard tell me if
what asterisk is doing is correct?
Or if it must be a bug with our customers equipment?
Reading RFC2327 it cais the c= line 'must' be present in all updates
while 'm=' media lines
On 27/09/13 14:36, Joshua Colp wrote:
Gareth Blades wrote:
We have an issue with a customer where when calls are sent to one of
their offices as soon as the call is answered the call fails.
We are performing remote bridging and switching the audio from the
server which initiated the call to our
On 27/09/13 14:28, Gareth Blades wrote:
On 27/09/13 14:15, Gareth Blades wrote:
Can anyone with a bit more knowledge about the SDP standard tell me
if what asterisk is doing is correct?
Or if it must be a bug with our customers equipment?
Reading RFC2327 it cais the c= line 'must
On 27/09/13 17:47, Phibee Network Operation Center wrote:
Hello,
I am looking to know if it is possible to modify the SQL query that is
on Realtime sip accounts.
I want multiple servers use the same sql table, so getting an extra
server field to indicate that the data is valid on the X
On 26/09/13 14:59, Rusty Newton wrote:
On Wed, Sep 25, 2013 at 6:45 AM, Gareth Blades
mailinglist+aster...@dns99.co.uk wrote:
We can use the Dial() command with the 'r' option in order to generate the
UK ringtone (as we are UK based the default is UK).
How do we generate a USA ringtone
On 26/09/13 15:25, Mauricio Tavares wrote:
So I have asterisk 1.8.23 and want to send my logs to rsyslog. I
tell asterisk to use syslog in addition to messages:
root@voip:~# tail -10 /etc/asterisk/logger.conf
;debug = debug
console = notice,warning,error
;console =
On 25/09/13 09:54, Kumar Shantanu wrote:
I am facing a strange problem on my asterisk box (using isdn lines
with pri card installed on it). Normal incoming/outgoing calls are
working perfectly fine.
When a user dial a wrong/out-of-service number they don't hear back
any such message like The
On 25/09/13 11:21, Kumar Shantanu wrote:
Thanks Gareth ,
Try calling Progress() just before the dial command. Without this
Asterisk wont send the SIP/183 Session Progress and send the inband
audio until the call is answered.
Do I need to change something in asterisk dial plan ? I am using
We can use the Dial() command with the 'r' option in order to generate
the UK ringtone (as we are UK based the default is UK).
How do we generate a USA ringtone for example?
I have tried setting the CHANNEL(language) and CHANNEL(tonezone) to 'us'
(and calling Progress() beforehand) and
On 25/09/13 13:57, Kumar Shantanu wrote:
Thank you Gareth,
It worked like a charm.
The only problem I am having is now, when I do some changes in my
freepbx and reload it just rewrites my dial play , I will try to fix
it though.
Thanks again
I did see in the console output it doing a
On 25/09/13 15:42, Andrew Colin wrote:
Hi Guys,
Anyone ever seen this before.
on asterisk 1.8 if i set one of my pabx extensions to show private
number and send a call over VoIP with g729 the call fails but with
alaw it works.
If i enable the callerid on g729 it also works
see error below
On 18/09/13 12:40, Kenny Watson wrote:
Hi,
Since opensips is not handling media (i presume) is the audio not already going
direct from asterisk to the other endpoint?
Thanks
Kenny
Opensips wasnt handling the media so the audio was between the original
caller and asterisk (with the
We have a system where calls are coming in from telcos via an opensips
server and then being redirected out to a regular sip destination.
There is no NAT, DTMF features, call recording, or codec translation
being performed so I would expect asterisk to issue a reinvite after the
call is
On 13/09/13 12:31, Henrik Westerberg wrote:
Hi,
I am running Asterisk 11.3 with both SIP and ISDN. When dialing out
(always over SIP) I want to keep track of who answered and of the
length of the call.
[outgoing-dev2]
exten = h,1,Agi(agi://localhost/ajpbxtest.agi?status=finished)
exten =
On 06/09/13 09:42, Dominique Haeber wrote:
Hi all,
is it possible that asterisk uses two proxies with SRV?
The enddevices are registered on one of the two Proxies (Kamailio).
The two proxies communicate with each other.
And asterisk can choose one of this proxies with SRV.
asterisk
| \
|
On 02/09/13 08:17, gpxctawjc...@irational.org wrote:
i am running Asterisk 1.6.2.5-0ubuntu1.4
and would like to know how to incorporate [default] encryption
can you point me to any guides please ?
do i need to upgrade ?
many thanks
There are two things :-
SIP TLS so that the invites are
On 28/08/13 19:34, Rusty Newton wrote:
On Wed, Aug 28, 2013 at 11:26 AM, Gareth Blades
mailinglist+aster...@dns99.co.uk wrote:
On 27/08/13 19:20, Asterisk Development Team wrote:
The Asterisk Development Team has announced security releases for
Certified
Asterisk 1.8.15, 11.2, and Asterisk
On 29/08/13 14:42, Olivier wrote:
Thanks for your very helpful reply.
1.My system prints out:
CLI core show application Hangup
-= Info about application 'Hangup' =-
[Synopsis]
Hang up the calling channel.
[Description]
This application will hang up the calling channel.
[Syntax]
On 27/08/13 18:16, Gregory Malsack wrote:
Hey All,
Growing call center. Currently at about 200 call center staff, running
about 1000 calls per hour. Gearing up to double that. Not too sure
that a single server will support that growth. So, I'm trying to come
up with ways to scale the system
On 27/08/13 19:20, Asterisk Development Team wrote:
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security
releases
are released as versions 1.8.15-cert2, 11.2-cert2, 1.8.23.1, 10.12.3,
On 22/08/13 15:43, Mordechay Kaganer wrote:
B.H.
Hello, i'm using AMI Originate action (with async=true) to send
outgoing calls to a SIP trunk (using asterisk-java library to connect
to AMI).
The problem is that in case of failed originate, OriginateResponse
event is returning only the
On 23/08/13 10:44, Mordechay Kaganer wrote:
In my experience with local channels, they cause a huge performance
problems, even without sipstorecause. We are dialing on about 100
channels in parallel and looks like this will kill my CPU :-)
Really?
We had Asterisk 1.8 with this setup and
On 20/08/13 17:48, Gergo Csibra wrote:
can I echo this variable ?
Like : exten = s,n,NoOp(${LAST_INSERT_ID()})
No, this is a mysql query, so:
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET C1=${ARG1},
C2=${ARG2}, timestamp=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
exten
On 20/08/13 09:29, Jonas Kellens wrote:
Hello,
how can I cut off the last character of the EXTEN-variable with
variating length ?
So I have :
112233#
123#
123456789#
I want to cut off the last character.
${EXTEN:-1} gives me #, but that is the character I want to cut off.
On 20/08/13 14:53, Jonas Kellens wrote:
Hello,
how can I obtain the inserted ID after having inserted a row with
MySQL in the dialplan ?
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET
C1=${ARG1}, C2=${ARG2},
timestamp=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
I need
On 14/08/13 18:48, Tony Mountifield wrote:
I have a system running CentOS 5.9 and DAHDI 2.6.2 with a 2-port E1 card
using the wct4xxp driver (also using Asterisk 11.5.0, but that isn't
relevant to the question).
With DAHDI and Asterisk started, the system appears to run normally, as
far as I
On 02/08/13 22:10, John Doe wrote:
Can you create dynamic bridges like in meemet?
Its the only way you can do it. Confbridge doesnt use a database or
config file to store the conference details and pin codes. You need to
manage all that yourself in the dialplan and then call confbridge to
On 05/08/13 12:38, Kamlesh Kumar wrote:
Hi,
Asterisk 1.6.2.9
PHP 5.3
Mysql 5.0
Can anyone suggest hardware specification for 500 hundred concurrent
SIP only calls, no codec transcoding, no IVR, no Voicemail or so. Just
plain switching. There is only one requirement is to execute one php
The last two companies I have worked for have both had this problem with
a BT ISDN30 line at some point. I also manage SS7 interconnects and its
not unusual for there to be issues with them either. So dont assume its
probably at your end :P
On 02/08/13 15:12, Andre Goree wrote:
I've
On 02/08/13 17:10, John Doe wrote:
Is there an easy way to have app_meetme create the recording in a temp
location and move it once the conference is over?
or should I just have a perl script run every minute to check for no
users in the conference room and then move it?
Asterisk 11
Thanks
On 31/07/13 15:32, Tony Mountifield wrote:
Most of my experience until recently has been in Asterisk 1.2, and I am
just starting to make use of Asterisk 11 for new systems.
I have a question about using SIP on a multi-homed machine.
I have a customer who wants an Asterisk box with two network
On 31/07/13 16:12, Tony Mountifield wrote:
Thanks. But I thought localnet= and externip= were for when the external
interface is going through NAT. In this case the ITSP is connected through
a real non-NATted public interface.
Is it possible to specify directmedia=no just for the SIP trunk? So
On 29/07/13 18:12, samuel wrote:
there's no dahdi installed.
Following debugging the issue, it looks like the astdb file is broken.
Whenever database show command is executed it loops over the same
results. The file itself is around 225K but dumping its content via
asterisk -rx 'database
On 30/07/13 05:35, Duncan Turnbull wrote:
E1 PSTN line interfaces are either unbalanced 75 ohm( and used to use
BNC connectors ) or a 120 ohm balanced twisted pair.
The other standard is T1 and digium cards can let you choose between
T1 E1 and definitely do 120 ohm
Telco's will usually
On 30/07/13 15:36, Andre Goree wrote:
/etc/dahdi/system.conf:
span=1,0,0,ESF,B8ZS
bchan=1-23
dchan=24
loadzone=us
The first '0' in your span line above indicated that asterisk is
generating the timing source. Normally the network operator provides
timing so I would expect this to be a '1'.
On 29/07/13 12:15, samuel wrote:
Hi folks,
Recently a customer of us moved his old asterisk installation, an
1.4.44 to a VMWARE infraestructure and has started having some weird
issues.
Asterisk started going slow and even refused to start up. After few
tests, it only loaded when
On 29/07/13 15:22, Eduardo Leones wrote:
Hello, working in a call center where we set up a structure in
asterisk. When my voip reaches 150 calls are with bad quality. We do
not transcode codec. What I realized using the top command server
(CentOS) processing is too high for the asterisk. But
On 29/07/13 16:28, Akib Sayyed wrote:
Dear asterisk users
I wanted to use E1 pri lines on my asterisk box but my provider
support only 120ohm on E1 line. I dont know how to set those values.
Please help me
Its done on whatever interface cards you have. Some may have a jumper
setting. I
On 26/07/13 16:32, Leandro Dardini wrote:
In my dialplan I'd like to send a 603 Declined message to the user
placing the call. I see the commands for the Busy and Congestion, but
not the one for the Declined. Any help?
Leandro
I dont think you can. Normally you would use the Hangup()
We have Asterisk 1.6.2.17.2 connected to a sangoma E1 card. The problem
we are having is that we have a calling card type application and when
people enter the number to be dialled we call the Dial application. It
gets back an indication that the number is incomplete (via PRI cause
code 28 I
Edwin Blommaerts wrote:
Hello everyone,
I have some questions regarding implementing asterisk and my-sql. I’m
no expert at asterisk but I’m going to list up my questions and
hopefully someone will be able to help.
-The people that setup our server made asterisk write the
Jonas Kellens wrote:
On 11/08/2010 09:50 PM, Jonas Kellens wrote:
Hello,
SIP DNS SRV records are not working.
My Grandstream uses the SRV records to find the first Asterisk server
to register to. This works.
But when I shut down the Asterisk proces on server 1 and I restart my
GXP
Jonas Kellens wrote:
On 11/09/2010 02:12 PM, Gareth Blades wrote:
Jonas Kellens wrote:
On 11/08/2010 09:50 PM, Jonas Kellens wrote:
Hello,
SIP DNS SRV records are not working.
My Grandstream uses the SRV records to find the first Asterisk server
to register to. This works
Use camailio or opensips as the registrar server so it accepts the sip
registrations. You can have copies running on a couple of boxes using
either a shared databases or a database on each server configured in
master-master replication mode. Opensips can be configured to use the
same database
Yes that is the way it is supposed to work. You do have to rely on the
sip devices you are using fully supporting SRV records though.
Jonas Kellens wrote:
Hello list.
When using SIP DNS SRV to define a production Asterisk server with high
priority and a backup Asterisk server with a
Something like http://www.audon.co.uk/udin.html UDIN-8R. It can only
control 750W so you will probably need to get it to control a more
powerfull relay as a heater is going to take a lot of current.
It can be controlled by a virtual serial port so you just program the
extension to make a
tell me for sure that when the production Asterisk server
becomes reachable again, the registration will go back to the production
server ??
Jonas.
On 10/18/2010 01:11 PM, Gareth Blades wrote:
Yes that is the way it is supposed to work. You do have to rely on the
sip devices you
Karim Davoodi wrote:
Hello,
I want to create channel bank in this case:
channel bank
|-|
| FXS,FXO-TDMoE--|--Asterisk
|-|
How can it?
Rizwan Hisham wrote:
Hi All,
Please refresh my memory. I am trying to install asterisk after 2 years.
I hav'nt used it since 2008 (version 1.4.2). Now I am trying to install
1.8.0-rc2 on centos 5.5 but getting the following errors.
app_mysql.c:33:25: error: mysql/mysql.h: No such file or
As the previous poster said use the sip software to make test calls.
Have the number it dials go out of the sangoma card and back into
another port via a crossover cable to an extension which answers and
plays back a file for a second or so before hanging up.
You can then make lots of calls
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