Get portsip ( www.portsip.com ) its realtively easy to configure ( just
push in user/password and server name at startup ) .. there might be NAT
issue so make sure you have nat=yes in ur asterisk's sip.conf for the peer
definition . If it still doesnt work then you need to find a iax phone like
Asterisk is not a sip proxy but it *can* partly act as a sip proxy if
reinvites are enabled ( canreinvite=yes in sip.conf ) only then asterisk
connects 2 end points directly and does signalling between them .
Asterisk is a PBX now suppose u need to record all calls ..do conferencing
stuff then
In your case it will send calls without registering to softswitch . Btw what
does your softswitch expects from asterisk ? like is it configured to
authenticate by username alone , user/pass or ip address ?? People here can
help you better if you post that info .
On 24/07/07, bilal ghayyad
Idefisk/zoiper softphone is for IAX2 and it works fine almost everytime .
However there is more variety in sip softphones . I think zoiper is much
better than other iax2 softphones .
On 25/07/07, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi All;
Thanks for all replies :) -
But that means,
Idefisk is now renamed to zoiper . http://www.zoiper.com/ :)
On 26/07/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Mittwoch, den 25.07.2007, 12:13 -0700 schrieb bilal ghayyad:
Hi BaharatSamaria;
Thanks for your kindly email.
Are (Xlite and phoner) IAX or SIP? From where I can
Btw are the phones behind NAT ?? Also you can try some softphone and make
sure that this problem is caused by snom phones or some other factors ..
On 25/07/07, Michael J. Liberatore [EMAIL PROTECTED]
wrote:
I thought it was the fios service but now I realize it's the snom 360!
It doesn't hang
in ur sip.conf under the device definition you can set it
for example device name is asterisk is pap2
[pap2]
username=pap2
secret=blabla
type=friend
disallow=all
allow=g729
Then asterisk will only use g729 for incoming as well as outgoing calls on
this device .
On 27/07/07, Matt [EMAIL
http://www.voip-info.org/wiki-Asterisk+config+sip.conf
*
call-limithttp://www.voip-info.org/wiki/edit.php?page=Asterisk+sip+call-limit
* = number : Number of simultaneous calls through this user/peer
On 27/07/07, Nicholas Blasgen [EMAIL PROTECTED] wrote:
I'm running Asterisk without FreePBX
Is this a web hosting forum or mailing list ?
On 31/07/07, Asterisk guy [EMAIL PROTECTED] wrote:
1and1 dedicated server's service has been down for a few hours , unable
to reach them by phone or email. do anyone know what is going on there ?
Mario
Do you have proper version of zaptel installed corresponding to your
asterisk version ?
On 31/07/07, Knud Müller [EMAIL PROTECTED] wrote:
Alex Balashov schrieb:
On Mon, 30 Jul 2007, Knud Müller wrote:
what does your modules directory contain? Can you find a file
Chanspy() app allows spying live channel but you will get 2 way voice in it
. I dont think any other app allows to spy on one side of call .
On 03/08/07, nik600 [EMAIL PROTECTED] wrote:
Hi
is it possible to spy (not record, spy) partially on a channel?
for exaple, i'd like to listen only
Yes, since IAX2 only uses one port, this is correct. Another thing to
keep in mind is to set a low qualify value in Asterisk since some
routers will tear down the connection pretty quickly. The qualify acts
as a keep-alive and prevents the router from closing the port and losing
the map.
sock=/tmp/mysql.sock
Is this path for socket correct ?
In some distro it is /var/lib/mysql/mysql.sock . Type locate mysql.sock in
shell . Also remove uncomment port=3306 if using socket to connect .
On 07/08/07, Alessandro Russo [EMAIL PROTECTED] wrote:
Hi, try to login as asteriskcdruser
When you make calls then context=xxx of the peer you are using ( your
extension ) will matter , the context=yyy line of your trunk wont matter .
If you dont specify a context= for a peer then it is considered to be in
[default] context .
On 07/08/07, Jared Smith [EMAIL PROTECTED] wrote:
On
This should be configured in phone system instead of asterisk :) .
On 08/08/2007, Michael Rice [EMAIL PROTECTED] wrote:
This is part f the phones dial plan. Our aastra phones do the same
thing. Most phones allow you to configure the dial plan on them.
satish patel wrote:
i have only one
Please stop advertising your forums/services on every single chance u get on
users list .
On 08/08/07, Al Bochter [EMAIL PROTECTED] wrote:
That is why you need to start posting info about the providers at
http://www.bochterservices.com/phpbb/
so everyone knows
This is a FREE SERVICE
google for ASTERISK CMD CHANSPY and follow voip-info link in search results
.
On 08/08/07, satish patel [EMAIL PROTECTED] wrote:
Dear all
i need this feature in asterisk whn 2 party calling that
time i pickup call and listen conversation of that party spoofing like is it
Enable mysql loggin of cdr's by installing asterisk-addons and use
asterisk-stat
http://www.areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54
On 13/08/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Aug 13, 2007 at 08:49:11AM -0500, Jeremy Mann wrote:
Does anyone have
I made same thread few months ago and many people said that they dont have
such records in plain asterisk install ( no freepbx ) . I was also using
freepbx when i had this problem . Heres mine :
mysql select count(*) from cdr where billsec duration;
+--+
| count(*) |
+--+
|
/08/07, *Edoardo Serra* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I noticed that fpbx calls ResetCDR on call hangup (don't know why
this
choice)
Could it be related to that ??
Tnx
E.
Jaswinder Singh ha scritto:
I made same thread few
What i actually do is make asterisk listen on some other port like 5097 and
redirect port 5060 to it with iptables like this
/sbin/iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5060 -j DNAT --to
YOURIPHERE:5097
This works very well . If i make asterisk listen on 5060 and redirect say
5097
Sin you have sangoma card , it will act as timer . You need to install
meetme ( app_conference is not very stable last time i read ) .
On 01/09/07, fateme fatah [EMAIL PROTECTED] wrote:
Hi:
I want to have conference call service and I use A102d sangoma's card.Do I
should install ztdummy or
You can use asterisk realtime which can read sip config from database (
mysql/pgsql) . Your application can just write info to database and asterisk
will read it and make peers . You can also include a custom config file
within sip.conf and make your application write peer settings to that file
'show channels' shows only running calls while 'sip show channels' shows
all running sip sessions including phones trying to register .
On 09/09/2007, ram [EMAIL PROTECTED] wrote:
Hi all
what is the difference between
show channels
sip show channles
i see the difference in both
show
I prefer centos , debian/ubuntu are also a good option . It just depends on
which distribution you are comfortable with . We also have asterisk running
very stable on slackware .
On 12/09/2007, Gordon Henderson [EMAIL PROTECTED] wrote:
On Wed, 12 Sep 2007, Euler Pereira wrote:
Hey all!
Sep 21 20:31:49 ERROR[3774]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql:
Unknown connection error:
(2006) MySQL server has gone away
This part is more like mistake in /etc/asterisk/cdr_mysql.conf . Check it
once and relaod asterisk , then you can type cdr mysql status in cli to
check if it
A2billing is very versatile and good solution for asterisk prepaid/postpaid
billing .
On 22/09/2007, Apa Minerala [EMAIL PROTECTED] wrote:
You should make sure you know how to install it yourself.
And you should also test it very very VERY carefully.
I can't underline very enough.
And if
Here you go http://www.voip-info.org/wiki/view/Asterisk+firewall+rules .
You can also set your rtp.conf properly and open very few rtp ports instead
of all 1-2 udp ports .
On 22/09/2007, Guenther Sohler [EMAIL PROTECTED] wrote:
Hallo,
I'd like to correctly set up my firewall in my
since asterisk is only using operating system's routing ability , you can
always set static routes using route command in linux .
On 26/09/2007, Jeremy Mann [EMAIL PROTECTED] wrote:
Why did you waste time with this reply? You do realize some users don't
have control over their Exchange
Also many people using softphone turn's on mic boost in windows xp which
also makes echo if it is set to very loud .
On 30/09/2007, Philipp Kempgen [EMAIL PROTECTED] wrote:
http://linux.sgms-centre.com/misc/netiquette.php#threading
http://linux.sgms-centre.com/misc/netiquette.php#toppost
SCNR
See chanspy in asterisk 1.4 , it also has a whisper mode and you can talk to
one party http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy . But i
dont know how to play a recorded file in it .
On 08/10/2007, Girts Graudins [EMAIL PROTECTED] wrote:
Hello everyone,
I'm looking for a
astrundir = /var/run
Change this to astrundir = /var/run/asterisk on 1.4 server and chmod
/var/run/asterisk to 777 . make sure u create that directory as well .
On 20/10/2007, Al lists [EMAIL PROTECTED] wrote:
this message is basically tells you asterisk is not running.
can you check and see
iax.conf doesn't take canreinvite=no . It's only for sip . For iax2 its
transfer=yes/no (1.4 ) or notransfer=yes (1.2) , there's one more parameter
in 1.4 with which u can transfer only audio stream . Check voip-info wiki
for all options .
On Thu, Mar 13, 2008 at 7:48 AM, Gonzalo Servat [EMAIL
I am using asterisk 1.4.18 (server A ) and have it store records in
mysql database . One of my client uses predictive dialer ( asterisk
1.2.26 based and server B ) which makes many calls . B registers with
A over sip and there is no nat involved If i re-invite rtp from
server B to my carrier (
That's strange , i am able to see the *url* in Martin's reply .
On Sun, Mar 23, 2008 at 6:14 PM, Eric Wieling [EMAIL PROTECTED] wrote:
The only messages I have EVER seen Digium remove from the mailing list
archives are discussions about this unlicensed codec.
Martin wrote:
Download an
When g729 phone calls another g729 phone and you are not recording
calls or doing meetme with them then license is not required ... g729
phone calling g711 will require a license to transcode the g729 side (
no license for g711 side of call ) . In short anytime u need to
convert g729 into some
Check dns server entries in asterisk box . /etc/resolv.conf . Put
opendns servers ip there just to test . opendns ip's are
208.67.220.220 and 208.67.222.222
On Tue, Jul 15, 2008 at 2:19 PM, map [EMAIL PROTECTED] wrote:
Hi Giorgio,
RE my point 2:
You should test a sip client, whatever you
[442033553]
user=442033553
type=pusers
secret=1234
host=dynamic
context=users
nat=yes
make it context=stations , i am assuming this is how your DID provider
is sending u calls ?
Let us know if your DID provider is just sending calls to your ip
address or you are registering asterisk server with
Why not use a asterisk specific live cd distribution like www.astlinux.org
? It is also installable on usb . You can copy your whole dialplan and
settings ( all files in /etc/asterisk ) on a pendrive .
On 25/04/07, Khaled Chehab [EMAIL PROTECTED] wrote:
Dears its too urgent
Can anyone guide
Try ilbc if the phone supports (free) or g729 ( better but your asterisk
will need license if you want to transcode calls from g729 to other codecs
or want to record calls ) . Also check your phones config if its support
multiple codecs . .
On 02/05/07, Rob Schall [EMAIL PROTECTED] wrote:
I was just going through my call records ( stored in mysql database
by cdr_MYSQL module ) and saw a record having duration = 0 and billsec
of more than 50 seconds . I did a query on cdr where duration
billsec and saw that there were infact some 250 records with duration
less than billsecond (
Someone in -biz list pointed out that this could be a freepbx problem
so i think i will go check there .
@ Salvatore Giudice:
how can i intentionally do it ? Damn i need a app that can make sure
customer phone doesnt hangup for the time i specify .. even if
customer breaks his phone . lol
try soft hangup sip channel name
On 02/05/07, Ken Williams [EMAIL PROTECTED] wrote:
I posted about this problem last week and thought it was a combination of
SIP/ZAP causing issues in Asterisk. Since then I've realized it's only the
SIP channel that's hanging. When this happens a call can
I have figured out a way to include dialed number in recorded
voicefile in freepbx . You have to edit
/var/lib/asterisk/agi-bin/recordingcheck
add this lines after $agi=new AGI()
$temp= $agi-get_variable(DIAL_NUMBER);
$agi-verbose(Number to be dialled is -{$temp[data]});
After this you can use
No python code needed . Check .call files at
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
On 23/05/07, Brad Sumrall [EMAIL PROTECTED] wrote:
Can anyone guide me to a how to on automating a call?
I know a little piece of code (normally python) has to be place some where
change conf = 222
to conf = 222
( remove | )
I had same problem as freepbx always put | removing it fixed the problem
On 29/05/07, Khaled Chehab [EMAIL PROTECTED] wrote:
I am using asterisk 1.4.4 now and facing a problem with meetme,the code I
was using with asterisk 1.2 is not functioning
I think its a fair decision . 1.2 is very stable and they are not
closing it all together , security issues will still be fixed . They
need to concentrate more on 1.4 to make it bugfree .
On 29/05/07, Stephen Bosch [EMAIL PROTECTED] wrote:
John covici wrote:
I have an install using Rhino cards
Well i guess you just need a good look on logs for why and when you
are getting core dumps . We are having few servers running .1.2.18 and
it has turned out to be most stable in whole 1.2 branch ( had some
issues with 1.2.13 and 14 ) .
Except that for some users 1.2.18 is NOT stable. I've had
What you say might be true for small business or home pbx systems .
But if you have a production server handling sip/iax trunks over
internet then you need to upgrade to avoid security related bugs and
exploits that are released .
You seem to miss the idea here. You work with a version
Well if you are out of luck with asterisk .. How about its fork
callweaver ? I am highly awaiting its stable release to see if it
holds upto what its wiki says .
On 30/05/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, May 29, 2007 at 01:06:54PM -0500, Eric ManxPower Wieling wrote:
Michael
Is it over iax and there are lot of outgoing channels ? If yes then
you are not the only person having this ..
On 30/05/07, ram [EMAIL PROTECTED] wrote:
Hi
i have 20 people calling agents calling
when ever they calling i get this below error
May 30 00:46:57 WARNING[2840]: channel.c:785
I dont think asterisk supports this . You can have host=dynamic and he
can send calls from different servers . Problem will arise when you
need to call him ( if registrations are enabled then latest
registration will be getting call from you or you can directly send
calls to his ip . )
On
Can you post some output from asterisk cli output while you make call ?
On 30/05/07, BSumrall [EMAIL PROTECTED] wrote:
after 18 hours, over 200 pages of reading, a complete reinstall of asterisk
I am down to this.
extensions.conf
[globals]
CONSOLE=Console/dsp
IAXINFO=guest
Do you have 'r' in ur dial command ? 'r' makes asterisk produce ring .
On 30/05/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
when a user dials any number, asterisk automatically generates ringing which
caller can hear, and after 2 - 3 rings asterisk detects that the called user
is busy,
You just have a 1 call limit on your account on net2phone side .
Making 10 trunk wont let you make 10 account its restriction on your
account not ip . Just change your provider .
On 01/06/07, Salah Eddine ELMRABET [EMAIL PROTECTED] wrote:
Hi,
Any help regarding Net2Phone poblem?
BR
On
Well freepbx runs fine with asterisk 1.4.4 ( atleast for me ) i think
some changes was introduced in 1.4 ( 1.4.4 ?) for some backward
compatibility... like show channels now work in 1.4.4 instead of
core show channels but it gives a notice that 'show channels' is
deprecated bla bla .Freepbx
independently install each rpm via rpm command :-/
On 04/06/07, Khaled Chehab [EMAIL PROTECTED] wrote:
I have 2 servers, one connected to internet and the other is on a private
lan have no access to internet.
On the first server I update the kernel by yum update
And installed asterisk
Asterisk by default uses the codec preferred by other device/client .
Asterisk 1.2 ( dunno abt 1.4 specifically) is not intelligent enough
to check if it can avoid transcoding by forcing same codec on other
side of conversation . If both sides prefer g729 then asterisk does
not do transcoding
It just might be that your carrier is not sending ring . You can use
'r' in asterisk dial command in extensions.conf to generate ring from
asterisk .
On 31/05/07, dima [EMAIL PROTECTED] wrote:
Hello, everyone.
Could anyone explain me how does ringback detection works in asterisk.
Sometimes,
Are you trying to record the conversation as well ?
On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote:
I installed a hardware g729 codec card in my asterisk, and I'm getting the
following error when calling from a g729 sip extension to a SIP trunk also
set to g729. The call goes through just
(number)=14073844200)
exten = _1NXXNXX,7,MixMonitor(${CALLFILENAME}.wav49)
exten = _1NXXNXX,8,Dial(SIP/[EMAIL PROTECTED],,wW)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent: Wednesday, June 06, 2007 4:28 AM
To: Asterisk Users
Yep its down for me tooo .
On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote:
Is anyone else having trouble going into voip-info.org today?
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Just read somewhere that you can use extension as g729 even in
mixmonitor so it will record g729 stream and later you can convert it
to mp3 or wav using sox . If this fails then try monitor application .
On 06/06/07, Jaswinder Singh [EMAIL PROTECTED] wrote:
I think asterisk first converts
patched here and there by one person or another,
but does anyone know if any of these patches to make CODEC negotiation
actually, you know, negotiate a CODEC will ever make it into the core
src?
Jaswinder Singh wrote:
Asterisk by default uses the codec preferred by other device/client .
Asterisk
I think there is a patch for sip over tcp in asterisk but not sure if
its stable or not
try this http://bugs.digium.com/view.php?id=4903
You can also install openser as sip proxy . it supports sip over tcp .
On Wed, 6 Jun 2007 19:14 +0300, Yehavi Bourvine +972-8-9489444
[EMAIL PROTECTED]
Its up and working now .
On 06/06/07, Compnet Bobby [EMAIL PROTECTED] wrote:
Same in southern cali!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent: Wednesday, June 06, 2007 8:48 AM
To: Asterisk Users Mailing List - Non
In sip.conf it should be bindport=5062
On 06/06/07, Crazy Boy [EMAIL PROTECTED] wrote:
Hi Friends,
I want to use 5062 port for SIP protocol. I made the below modifications in
my server to use 5062 port.
Polycom phone: port=5062
Trunk settings: port=5062
sip.conf: bindaddr=5062
Extension
In your no-ip client set it to update ip every 2 minutes or so . and
/etc/asterisk/dnsmgr.conf set refresh interval as 30-40 by defualt
its 300 ( 5 minutes)
On 10/06/07, Ronaldo Z. Afonso [EMAIL PROTECTED] wrote:
Hi Matt,
Every time I do that, IAX stop sending the POKE messages (necessary for
.
Jaswinder Singh wrote:
In your no-ip client set it to update ip every 2 minutes or so . and
/etc/asterisk/dnsmgr.conf set refresh interval as 30-40 by defualt
its 300 ( 5 minutes)
On 10/06/07, Ronaldo Z. Afonso [EMAIL PROTECTED] wrote:
Hi Matt,
Every time I do that, IAX stop sending the POKE
Remove Answer() and try .
On 12/06/07, Rosalinda Trevino Cadena [EMAIL PROTECTED] wrote:
I'm using the Dial application in the extensions file with the G option
to execute an AGI script after the Dial (I need to track the call status) as
follows:
exten = _X.,3, Dial({DIAL_STRING},,G(_X.^4))
What does sip show peers output ? Also set a timeout in millisec like
qualify=200 instead of qualify=yes
On 14/06/07, randulo [EMAIL PROTECTED] wrote:
I totally puzzled by this situation. I have asterisk 1.4.4 behind NAT.
All SIP peers are working properly to place or receive calls.
Any SIP
Please do not post same thing again and again . It wont help you get better
replies , Post you asterisk cli output while call is in progress and when it
disconnects prematurely .
On 18/06/07, Don Kelly [EMAIL PROTECTED] wrote:
I am attempting to establish SIP peering between Asterisk and an
You can use bindaddr=0.0.0.0 to bind to all interfaces in sip.conf and
iax.conf .
On 23/06/07, Jordan Novak [EMAIL PROTECTED] wrote:
I have a simliar problem as the port binding question.
I have a four port parelell processing NIC, I would like to team them
together. Can I do this in
exten =*76,1,Answer
exten = *76,2,Chanspy(|qb) ; q for quiet and b for only bridged calls
exten = *76,3,Hangup
Now you can spy on any call ,. All you need to do is press * again and again
to change calls . Like if 3 calls are going then you can switch between
calls by pressing * and #
This is due to changes in cdr in asterisk 1.4.5 so in all outgoing
dial from macro it shows 's' in cdr . Could this be a bug ? I doubt if
it was intended to be that way .
On 25/06/07, Troy - Purple Oranges [EMAIL PROTECTED] wrote:
I am using VoiceOne http://voiceone.it/ as my management
It was due to changes in cdr in asterisk 1.4.5 previous version does not do
it .there is a fix on bugs.digium.com or you can wait till next release or
use asterisk 1.4.4
On 28/06/07, Rob Schall [EMAIL PROTECTED] wrote:
I currently have about 50 polycom 501 phones on my asterisk setup. The
Sorry i didnt read your mail properly . I thought your problem is with
cdr's. Here's link to cdr problem :)
http://lists.digium.com/pipermail/asterisk-dev/2007-June/028085.html
see the next message for patch .
On 29/06/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Rob Schall wrote:
I jumped into asterisk 1.4 and its pretty stable .. i never got a core dump
but it did halt while reloading a few times . I am back on asterisk 1.2 now
but i think asterisk 1.4 is stable .
On 29/06/07, Bruce Reeves [EMAIL PROTECTED] wrote:
While I have not jumped all my systems to 1.4, there
Are you sure calls were dropped with change in IP ?? I think it should let
current calls run and use new IP for new connections . However if
destination serv drops calls then it's a different story .
On 03/07/07, Henry J. Cobb [EMAIL PROTECTED] wrote:
Asterisk 1.4.5 full log:
[Jul 2 09:31:16]
Think about voicesense which will sense what you are talking and pop in a
*relevant* voice ad to spice up conversation :P .
On 03/07/07, Dean Collins [EMAIL PROTECTED] wrote:
Ooops did Google just become a carrier :)
http://googleblog.blogspot.com/2007/07/all-aboard.html
I hear stocks
Yes that is write order . libpri then zaptel then asterisk . Remember that
zaptel compilation is not required if you are using asterisk for voip only
environment .But it's always good to install it before asterisk if you want
to use conferencing abilities of asterisk .
Regards,
Jaswinder Singh
extension/peer ( please correct me if i am wrong ).
Regards,
Jaswinder Singh.
On 05/07/07, Eugene Prokopiev [EMAIL PROTECTED] wrote:
Hi,
Is it possible to filter messages on asterisk console, which was started
with -, to see messages only for one extensions? By default there
are all messages
Yes just download new version of asterisk,zaptel,libpri . make install
for all 3 ( first libpri , then zaptel, then asterisk ) . It is recommended
to stop asterisk b4r doing make install of new version . Do not do make
samples or it will overwrite you config's . After installing newer zaptel
do
Asterisk is poor with codec negotiation . It does not check if it can avoid
transcoding by forcing codec available to both sides .. instead it will
read it's config file and will select first allowed codec that is also
available on other device on each leg of call and happily transcode between
If you manage to get everything working with canreinvite=yes ( i suppose u
figure out nat issues ) then you cant play music on hold , can't record
calls , and can't do most of pbx stuff asterisk is capable of .. but dont
worry asterisk doesnt disable all this features if canreinvite=on .. like if
ITS Open source related .
On 22/03/07, Bill Hackensack [EMAIL PROTECTED] wrote:
On 3/22/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello
P.S The program that I am using is open source, of course
( www.phpsurveyor.org)!
What part of the survey is running Asterisk?
[outgoing]
exten = 1234,1,Dial(SIP/[EMAIL PROTECTED])
Whats the dialplan number to ring to userA on server (ONDO) ?
if u know that try
exten = 1234,1,Dial(SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED]
)
since ur sip.conf has [*192.xxx.xxx.xxx*-out].
i am not sure why you use Sip/test to call to
You can use qualify=(time in ms) option in sip.conf but its phone's
configuration that should register to asterisk everytime its reconnected .
On 26/03/07, cimsi [EMAIL PROTECTED] wrote:
Hi,
I've noticed that if I disconnect or reconnect a phone from the net,
Asterisk take long time to realize
disable voicemail for that extension .. apply settings .. re-enable
voicemail .. re-apply settings . this helped me once before.
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Wow i need a tftp client to download it now .
Nice April 1 joke :P .
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I registered few days back and got a DID. Maybe this is temporary ?
On Mon, Apr 6, 2009 at 7:05 PM, Steve Howes st...@geekinter.net wrote:
On 6 Apr 2009, at 14:32, Dean Collins wrote:
None of their pages apart from the front page seem to work though
Here's what fail2ban service caught
The IP 89.111.184.221 has just been banned by Fail2Ban after
80 attempts against ASTERISK.
On Wed, Apr 8, 2009 at 7:01 PM, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com wrote:
On Tuesday 07 April 2009 11:28:52 Tilghman Lesher wrote:
The recent
There is also GNUdial but i would prefer VICIdial anyday over it ( personal
opinion :) ) .
On Wed, Jun 10, 2009 at 9:43 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com
wrote:
Thank you Jose.
Interesting suggestion!
Is there any other?
On Wed, Jun 10, 2009 at 11:21 AM, Jose P. Espinal
If you plan it right from the start, FreePBX can save hell lot of time.
Instead of fixing in include files, you can also create custom contexts from
within the GUI now, i am sure there is a module for that as well. As said
above, either stick fully to GUI or fully to manual configurations. Ugly
Latest version of X-lite does not have GSM codec . Downgrde your versiona dn
you will get gsm codec . I read on their forums that next version will
again be including GSM codec .
On 03/11/2007, Julio Tejera [EMAIL PROTECTED] wrote:
Latest version of X-Lite does not
support GSM codecs any more
asterisk -rx module load codec_g729.so or module load codec_g729 and
shhow translation recalc
On Nov 27, 2007 2:36 AM, Fernando Berretta [EMAIL PROTECTED]
wrote:
Dear Mindaugas,
Thanks for your promt response
I've already tried this but.. it's not working,, what file do you think I
should
:) .
On Nov 27, 2007 3:33 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Nov 27, 2007 at 03:00:19AM +0530, Jaswinder Singh wrote:
asterisk -rx module load codec_g729.so or module load codec_g729 and
shhow translation recalc
You ca't really fix a typo without intrducing a new one, eh
Its pretty clear from netstat that asterisk is listening on udp 5060 . It
might be firewall configuration in server thats blocking it . Also you might
have scanned for TCP port 5060 from outside and hence u find it closed ?
On Nov 28, 2007 5:57 AM, Nick Brown [EMAIL PROTECTED] wrote:
Zaheer,
Can you post the part of your dialplan which causes this behaviour ?
On Dec 17, 2007 11:19 PM, Roger Schreiter [EMAIL PROTECTED] wrote:
Hi,
some months ago, I had the problem with an asterisk-1.4.x-
Version, that some calls (but not all) were interrupted
64 seconds after connect (a call
call-limit is to set number of alternate calls . and L is to limit
duration of each call .
On Dec 22, 2007 2:54 PM, Pezhman Lali [EMAIL PROTECTED] wrote:
Dear
I am using this function with L
for example in the dbase.
app=Dial
appdata=SIP/[EMAIL PROTECTED]|60|L(10)
it means dial 1
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