Hello,
I face a problem on some dahdi incoming calls. Hardware is Xorcom with
Elastix 1.6.2.27/Asterisk 1.4.36/DAHDI 2.4.9-svn-r9328 inside. Setup is
3 incoming BRI (euroisdn), ringing phones are 3xSNOM320, 4xSNOM300 and
4xFXS phones.
On this calls, phones are ringing and when picked up,
Le 13/01/2012 14:32, Jonas Kellens a écrit :
On 01/13/2012 02:23 PM, Doug Lytle wrote:
Jonas Kellens wrote:
I have the following in dialplan :
[TrunkAccounts]
dialplan show TrunkAccounts
Make sure the sort order is what you're expecting.
Doug
Hello,
The order is correct for as far as
Le 27/12/2011 16:04, Tim Nelson a écrit :
- Original Message -
On Mon, Dec 26, 2011 at 11:54 PM, virendra bhati virbh...@gmail.com
wrote:
Hi list someone is trying to hack my server . Is there any way by
whcih I can stop hacking of my server except iptables ?
[...]
Odd nobody
Le 06/12/2011 10:16, Harel Cohen a écrit :
Hello Everyone,
Hi Harel
I didn’t get a reply to my problem below so I’m posting again just in
case someone who might be able to help missed my previous post.
Thank You…
Please take a look at issue ASTERISK-18875
Le 01/12/2011 13:44, Olivier a écrit :
[...]
I still can explain myself why a PoE switch (a Linksys SRW224P) would
succeed or fail to deliver power to a plugged IP phone, given that
only a couple of Polycom phones are using this switch a power source.
I think your switch deliver a max value of
Hi list,
something crazy here. 2 asterisk on 2 different place (1.4 and 1.8) both
having an extension [115], one as type peer (caller side 1.4) and one as
friend (callee side 1.8). Phones from both location connect to Asterisk
from LAN. Router are Linux boxes.
Connection between the 2 sites
Le 16/11/2011 10:23, Faraj Khasib a écrit :
Hi all,
I tried making a video SIP call using Asterisk But it didnt workonly
voice call works?
Hi Faraj,
Asterisk support H261, H263, H263+ and H264. Video calls are working
since at least 1.4 version. You have to activate it by setting
Hi all,
I have a question: we have few customers asterisk servers runing 1.4 1.6
or 1.8 asterisk version under Debian Lenny or Squeeze. No one of this
computer has telephony card, so we use dahdi_dummy for timing. Asterisk
and dahdi always compiled ourself (*)
Last week we face quality
Le 07/11/2011 10:19, Tzafrir Cohen a écrit :
How can we get dahdi_dummy compiled on those machines?
You no longer need to. Merely loading the module dahdi provides timing
and pseudo channels for conferences if no DAHDI hardware is available.
Well:
output of 1.6.20 without dahdi_dummy
, aso), no luck :-) I don't now
if it's
solved for him.
If someone had a solution on this, would be great to share ;-)
Regards
--
Daniel
Le 07/10/2011 15:01, Administrator TOOTAI a écrit :
Hi,
my asterisk 1.8.7 is working well with phones (SNOM, Gigaset
of the proposed ideas where working (reverse permit/deny, tried
with only permit=0.0.0.0/0.0.0.0, aso), no luck :-) I don't now if it's
solved for him.
If someone had a solution on this, would be great to share ;-)
Regards
--
Daniel
Le 07/10/2011 15:01, Administrator TOOTAI a écrit :
Hi
on this, would be great to share ;-)
Regards
--
Daniel
Le 07/10/2011 15:01, Administrator TOOTAI a écrit :
Hi,
my asterisk 1.8.7 is working well with phones (SNOM, Gigaset 620 and
GrandStream) connected from the lan
I now want to connect a snom320 from outside but it failed, having always
[Oct 7 14:48
Hi,
We can't read the messages in our mailbox always getting
-- SIP/tootaiAUDIO-0001 Playing
'/var/spool/asterisk/voicemail/default/100/Old/msg0002.slin' (language 'fr')
[Oct 11 13:24:50] WARNING[26778]: app_voicemail.c:7802 play_message:
Playback of message
Le 07/10/2011 16:32, Kristijan Vrban a écrit :
remove the c argument
Done but now I have
[Oct 8 19:20:20] WARNING[8771]: res_fax.c:1508 receivefax_t38_init:
channel 'SIP/tootaiAUDIO-00ea' refused to negotiate T.38
[Oct 8 19:20:20] WARNING[8771]: res_fax.c:1529 receivefax_t38_init:
-Commercial Discussion
asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] Asterisk 1.8.7 and ReceiveFAX
On 9/10/2011 1:29 AM, Administrator TOOTAI wrote:
Le 07/10/2011 16:32, Kristijan Vrban a écrit :
remove the c argument
Done but now I have
[Oct 8 19:20:20] WARNING[8771
Hi,
I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken
from deb http://packages.asterisk.org/deb lucid main) including dahdi
from this same repository. No FFA involved.
On incoming calls (only SIP, no telephony card), fax detection is
working but reception failed with
Hi,
my asterisk 1.8.7 is working well with phones (SNOM, Gigaset 620 and
GrandStream) connected from the lan
I now want to connect a snom320 from outside but it failed, having always
[Oct 7 14:48:04] ERROR[3870]: netsock2.c:94 ast_sockaddr_stringify_fmt:
getnameinfo(): ai_family not
Hi list,
we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is
Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk
1.4.36 (Elastix). Both 64bits, no hardware involved, dahdi on both
machines for meetme timing.
Doing core show translation give on the
Le 30/09/2011 14:05, Kevin P. Fleming a écrit :
On 09/30/2011 03:56 AM, Administrator TOOTAI wrote:
Hi list,
we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is
Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk
1.4.36 (Elastix). Both 64bits
Le 30/09/2011 16:59, Eric Wieling a écrit :
I always use the recalc option to show translations, it seems to provide much
more accurate numbers.
Example: core show translation recalc 20
Lenny kernel, new values, still 1000 microseconds between both directions
Translation times
Le 30/09/2011 17:02, Jason Parker a écrit :
On 09/30/2011 09:53 AM, Tony Mountifield wrote:
In article4e85d19f.4090...@digium.com,
Kevin P. Flemingkpflem...@digium.com wrote:
This is why the output was changed to microseconds from milliseconds; in
the older version, the lowest number that
Hi all,
using asterisk 1.4 or 1.6, I face a problem with the read command.
I call my asterisk box which ask me to enter the number I wish to call.
Problem is that if I make a mistake in the number and correct it on the
phone keyboard (smartphone under android, the same with nokias series
E),
Hi,
we have 4 asterisk, versions are 1.4.35 1.4.36 1.6.2.18 and 1.4.42 One
GrandStream GXV3000 is used for the tests. He is registered to asterisk
1.6.2.18 asa well as 1.4.35. Calling echo test is OK on both servers,
get audio and video. Calling echo test from asterisk 1.4.36 bye a SIP
trunk
Le 22/06/2011 01:10, ERIC HERRON a écrit :
I know Asterisk 1.8 can send out texts via SMS()
Can I send Asterisk a text via a DID and it do something?
[...]
You can receive SMSs using smsq (at least in 1.4) But be aware that most
of mobile carriers (eg France) send SMSs to landlines number
Hi,
Nobody on this?
Le 16/05/2011 23:35, Administrator TOOTAI a écrit :
Le 16/05/2011 18:27, Jose P. Espinal a écrit :
Administrator TOOTAI wrote:
Of course it's 1.4.41. And the result is that devices doesn't
register anymore.
Thanks for any hint.
If you are installing from source
Of course it's 1.4.41. And the result is that devices doesn't register
anymore.
Thanks for any hint.
Le 14/05/2011 17:37, Administrator TOOTAI a écrit :
Hi list,
We have devices since more then 4 years which where running well with
Asterisk. But with latest version (1.38 or more) we face
Le 16/05/2011 18:27, Jose P. Espinal a écrit :
Administrator TOOTAI wrote:
Of course it's 1.4.41. And the result is that devices doesn't
register anymore.
Thanks for any hint.
If you are installing from source, check out if some modules did not
load properly due to undefined symbols
Hi list,
We have devices since more then 4 years which where running well with
Asterisk. But with latest version (1.38 or more) we face problem with
those devices when they try to register. We got
[2011-05-14 17:18:06] WARNING[28559]: chan_sip.c:9950 register_verify:
Failed to parse contact
Le 29/04/2011 00:42, Russell Bryant a écrit :
- Original Message -
Sure. Please follow the 2 next stories:
- had a customer running 1.4.26 We upgraded to a new server and
installed 1.4.39, last version at this time. Bang: voicemail doesn't
work as it should, had to fallback to 1.4.26
Le 28/04/2011 16:53, Russell Bryant a écrit :
- Original Message -
PS. Please don't start a discussion about 1.8 quality in this thread,
that's a separate issue. I just want to know what you think about
closing 1.4 support now. If you want to discuss 1.8 quality, start a
new thread.
Le 28/04/2011 21:47, Leif Madsen a écrit :
On 11-04-28 12:04 PM, Administrator TOOTAI wrote:
Ok, so why not stay with asterisk 1.4 security *and* bug/regression fixes for
few weeks/monthes till 1.8 reaches the level that the community accept to switch
to 1.8
What is the guide here? What
Le 28/04/2011 22:43, Leif Madsen a écrit :
On 11-04-28 04:33 PM, Administrator TOOTAI wrote:
Le 28/04/2011 21:47, Leif Madsen a écrit :
On 11-04-28 12:04 PM, Administrator TOOTAI wrote:
Ok, so why not stay with asterisk 1.4 security *and* bug/regression fixes for
few weeks/monthes till 1.8
Le 27/04/2011 21:34, Olle E. Johansson a écrit :
Friends,
We have a discussion on asterisk-dev about the maintenance of the 1.4 branch.
According to the release plans, support for 1.4 was scheduled to close in April
2011 - basically now. After that, only security patches would be committed.
Le 29/03/2011 19:34, Sherwood McGowan a écrit :
On 3/29/2011 12:25 PM, Steve Edwards wrote:
On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan
First thing I'd do is restrict the ip blocks your sip endpoints can
register/call from in sip.conf (or your database's table for sip
endpoints)
On
Hi,
is there a way to know if originate call channel ended the call *before*
connecting to context/extension/priority?
DIALSTATUS is empty, HANGUPCAUSE is always 16, nothing in SIP Headers
nor in AST_CONTROL_FRAME_[HANGUP|ANSWER]
Asterisk is 1.6.2.16
Thanks for any hint
--
Daniel
--
Le 14/02/2011 15:44, salaheddine elharit a écrit :
thanks for your response
i have tested with a regular phone and i get the same result
my question if there is any action to do in dial plan or
extenssion.conf in order to call this number becouse in dial plan i can
bloc a number to be call
Le 15/01/2011 20:38, Cédric Lemarchand a écrit :
Hello,
Hi
[...]
I am sure there are RTP packets losses somewhere, except RTP debug in
the asterisk CLI, how can i determine where the problem come from ?
[...]
You don't tell which protocol (SIP, IAX, H323) nor which asterisk
version.
Le 04/01/2011 20:50, Sebastian a écrit :
Hi,
On 01/04/2011 03:24 PM, Administrator TOOTAI wrote:
Le 04/01/2011 11:50, Gilles a écrit :
[...]
It looks like getting a 3G smartphone with SIP + OpenVPN + unlimited
Internet plan would solve the issue.
I Would avoid OpenVPN (tested an Android
Le 03/01/2011 18:28, Gilles a écrit :
On Mon, 03 Jan 2011 12:27:56 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
As you are a Free Telecom customer, why not using your freephonie
account to forward incoming calls to your mobile?
Thanks for the tip, but experience shows
Le 04/01/2011 11:50, Gilles a écrit :
[...]
It looks like getting a 3G smartphone with SIP + OpenVPN + unlimited
Internet plan would solve the issue.
I Would avoid OpenVPN (tested an Android) as it drains quickly battery
[...]
2. what smartphone supports installing an SIP + OpenVPN
Le 01/01/2011 18:32, Gilles a écrit :
On Wed, 29 Dec 2010 16:55:46 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
I wouldn't be one of your friend: when I'm calling you I call a landline
but finally will be charged for a mobile call (imagine I have free calls
to landlines from my ISP
Le 28/12/2010 20:31, Kevin P. Fleming a écrit :
[...]
If you have a suggestion for a better place for this information to be
made available, please let us know. [...]
For instance in overview of Hx8:
New with the release of the H8 cards is Digium's B400M four-port
EuroISDN S/T module. The
Le 29/12/2010 12:16, Gilles a écrit :
[...]
In case a call comes in and I'm not home, I'd like Asterisk to log the
call, and then send an SIP message to my VOSP so the call is forwarded
to my cellphone and is thus charged to the caller, without Asterisk
having to dial out to my cellphone
Le 27/12/2010 20:09, Kevin P. Fleming a écrit :
On 12/27/2010 12:37 PM, Administrator TOOTAI wrote:
[...]
d...@myphoneserver:/usr/src$ strings
/usr/src/asterisk-1.4.38/channels/chan_dahdi.so | grep '^DAHDI
Telephony'
DAHDI Telephony w/PRI
DAHDI Telephony Driver w/PRI
Asterisk 1.4 has never
Le 28/12/2010 13:10, Kevin P. Fleming a écrit :
On 12/28/2010 05:17 AM, Administrator TOOTAI wrote:
Le 27/12/2010 20:09, Kevin P. Fleming a écrit :
On 12/27/2010 12:37 PM, Administrator TOOTAI wrote:
[...]
d...@myphoneserver:/usr/src$ strings
/usr/src/asterisk-1.4.38/channels/chan_dahdi.so
Le 24/12/2010 16:47, Steve Davies a écrit :
On 24 December 2010 14:40, Administrator TOOTAIad...@tootai.net wrote:
Hi,
We had 2 asterisk 1.4 connected together in iax, all was fine. One of them
was upgraded (server and Asterisk) in 1.6.2.15, the other end is in 1.4.38
When calling to 1.4
Le 27/12/2010 16:20, dave george a écrit :
[...]
[Definition]
#_daemon = asterisk
# Option: failregex
# Notes.: regex to match the password failures messages in the logfile. The
# host must be matched by a group named host. The tag HOST
can
# be used for standard
Hi,
we upgraded an Asterisk 1.4 with mISDN to 1.6 with chan_dahdi. Due to
problems with iax channel posted earlier, we wanted to switch back to
1.4 version.
Server has 2 HBA cards, everything is running fine with 1.6, bri_cpe is
recognized and the 7 euroISDN channels are running well,
Hi,
We had 2 asterisk 1.4 connected together in iax, all was fine. One of
them was upgraded (server and Asterisk) in 1.6.2.15, the other end is in
1.4.38
When calling to 1.4 to 1.6.2 -remember, it's iax- all is good. But
calling from 1.6.2 to 1.4 give a bad audio to calling party (words are
Hi,
I had 2 Asterisk servers connected together in iax with auth=rsa and
proper keys for user and peer in each direction. It worked well till I
upgraded one of them to Asterisk 1.6.13 Since I get No authority found
I thought that problem came from keys as the server with 1.6.13 was
changed
Le 17/12/2010 07:45, Gilles a écrit :
On Thu, 16 Dec 2010 17:05:35 -0500, Jamie A. Stapleton
jstaple...@computer-business.com wrote:
Just add something like this to your dialplan:
exten=1234,1,Dial(SIP/u...@domain.com)
Then, when you dial 1234 on your XLite, it will connect you to
Le 17/12/2010 12:48, Gilles a écrit :
On Fri, 17 Dec 2010 10:16:00 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Then create a prefix for SIP calls
exten=_9.,1,Dial(SIP/${EXTEN:1})
and you dial 9u...@domain.com from XLite
Remember that calling sip URL is not as easy with a phone
Le 17/12/2010 16:52, Gilles a écrit :
On Fri, 17 Dec 2010 14:36:58 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Domain part disappear.
exten=_9*.,1,Dial(SIP/${EXTEN:1...@ekiga.net)
In Xlite call 9*031600
Thanks for the tip but I wanted to be able to call _any_ SIP number
Le 15/12/2010 15:21, Gilles a écrit :
[...]
;IMPORTANT: outgoing must be BEFORE incoming
[vosp_outgoing]
type=peer
host=myvosp.com
username=myaccount
secret=mypasswd
fromuser=myaccount
fromdomain=myvosp.com
nat=yes
canreinvite=no
[vosp_incoming]
type=peer
host=myvosp.com
nat=yes
canreinvite=no
Le 13/12/2010 11:43, Gilles a écrit :
[...]
In case someone from France follows this thread, I'm interested in any
feedback about professional-grade ADSL that supports VoIP, as a
serious alternative to ISDN for telephony
We are selling our own xDSL but a France Telecom Pro can do the job.
Le 05/12/2010 20:28, Olivier a écrit :
[...]
Which Dahdi version ?
I had to use latest trunk to have mine working.
Thanks for your reply
SrvPhone2*CLI dahdi show version
DAHDI Version: 2.4.0 Echo Canceller:
FYI I got it: cable was defect.
--
Daniel
--
Hi,
I have 2 servers: one is running 2 B410P cards with 8 euroisdn lines
(mISDN) connected on it, everything runs fine.
I prepare a new server - HP 360 G8- with 2 HA8 cards each of them 1
module of 4 lines. Already had with this machine an RMA on both cards as
they was faulty and crashed the
Le 26/10/2010 14:49, Shaun Ruffell a écrit :
[...]
First, Digium technical support would be more than happy I'm sure to
help you trouble shoot this. That being said...
First thing I would do is update to the current trunk of dahdi-linux.
Revision 9397 [1]
Le 26/10/2010 14:49, Shaun Ruffell a écrit :
On 10/26/2010 06:38 AM, Administrator TOOTAI wrote:
I installed 2 HB8 cards each of them with a Quad Bri modules in a HP 360
G6 running Debian Squeeze. Here is an output of dmesg wafter server has
booted:
[...]
before asking RMA
Le 05/10/2010 05:13, VoIP Question a écrit :
Hello,
Hi
I would like to verify if a specific SIP header exists, and if yes,
extract the partial content from another header.
1. Is there a way to verify if a specific header exists?
2. How do I extract data that is between the first : and
Le 06/09/2010 15:10, Olivier a écrit :
Hi,
Hello
1. Do you have any experience with receiving incoming SMS on an analog
or ISDN landline ?
How can then you differentiate an SMS call from a voice call ?
From http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms it seems the
way to tell an
Le 06/09/2010 17:39, Olivier a écrit :
2010/9/6 Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net
Le 06/09/2010 15:10, Olivier a écrit :
Hi,
Hello
1. Do you have any experience with receiving incoming SMS on an
analog
or ISDN landline
Le 06/09/2010 19:31, Randy R a écrit :
[...]
Some of this may have changed, but when I has asterks and a fixed-line
SMS service from France Télécom, that's the way it worked.
End of 2009 SMS sended to landlines where easy to treat, we even setup
an SMS2Mail gw. Those days, we only treat
Le 18/08/2010 16:03, Tino a écrit :
Hello Johann,
Thanks for your advice in this matter. But i am not sure how to pass
the numbers to be sent sms in the dialplan.
agi(script,param1,param2,...,paramX) from your dialplan where script
lies in /var/lib/asterisk/agi-bin
On Wed, Aug 18, 2010 at
Hello
Le 27/07/2010 20:57, Cassius Smith a écrit :
Here's a strange thing.
I'm deploying Asterisk 1.6.2.9 with a pile of Cisco 79xx phones. For
conference rooms we're using Polycom IP6000's. We bought two of them
brand new.
[...]
Any ideas? I'm stumped.
If tour register server is
Le 25/07/2010 02:11, Norbert Zawodsky a écrit :
Hello again!
Hi
after it being relatively quiet her for the last weeks, my Astrerisk
server was the target of 3 of that nasty REGISTER attacks during the
last days.
[...]
Do like most of us are acting: use fail2ban.
--
Daniel
--
Hi list,
I face a problem with voice SMSs. In some countries, if you send an SMS
to a landline number, the mobile operator will record the message and
then call this number. When picking up the phone you hear You get an
SMS from phone number, press 1 to listen the message, 2 to repeat the
Le 15/07/2010 10:38, Gordon Henderson a écrit :
On Thu, 15 Jul 2010, Administrator TOOTAI wrote:
Hi list,
I face a problem with voice SMSs. In some countries, if you send an SMS
to a landline number, the mobile operator will record the message and
then call this number. When picking up
Le 23/06/2010 21:28, Gordon Henderson a écrit :
[...]
I'd like to have a look, but can't - I think there may be issues with your
registrar for your domain - from where I am, there are no glue records for
the nameservers, therefore I can't look it up... Looks like it was last
edited just over
Le 21/05/2010 16:19, Motiejus Jakštys a écrit :
Hi, List,
I am looking for a cheapest (and therefore most funny) way to attach
GSM card to my asterisk home box.
Have a look at chan_mobile (bluetooth connection)
--
Daniel
--
Gordon Henderson a écrit :
Just a heads-up ... my home asterisk server is being flooded by someone
from IP 184.73.17.150 which is an Amazon EC2 instance by the looks of it -
they're trying to send SIP subscribes to one account - and they're
flooding the requests in - it's averaging some
Hi,
what is the state at this time for 64bits applications and compatibility
with 1.6.2
Mainly speaking about FFA, SFA, G729.
Thanks for any information
--
Daniel
--
_
-- Bandwidth and Colocation Provided by
Juan C. Villa a écrit :
[...]
The total lag from Germany to USA (2 way) is around ~110ms (Just tested
it today). Who this cause any issues with my VoIP applications? Right
now I have two VoIP boxes installed in Switzerland which are connected
to my server in California (avg response time =
Juan C. Villa a écrit :
Hey Guys,
HI Juan
I am considering leasing a new server in Germany to run my Asterisk
infrastructure and I was wondering how response time would affect the
performance of the system. Right now I have a response time of around
60-70ms with my server in
Hi
Daniel Bareiro a écrit :
[...]
Hours ago the IP changed and the domain was updated satisfactorily, but
in spite of this I was obtaining the registering failures that I
mentioned above. After to restart Asterisk (1.4.24.1), I no longer had
this problem of registering. But there would be
sean darcy a écrit :
[...]
Context names cannot be duplicated, unless you suffix them with (+) to
allow them to be added together. It does not matter whether it is the
'global' context or any other context.
Well
Dialplan reloaded.
== Parsing '/etc/asterisk/extensions.conf': ==
Hi,
we had an attack on a server and we don't understand how it was
possible, Asterisk 1.4.28/Debian Lenny 5.1 Attacker came from PALTEL,
network 188.161.128.0/18
Hacked account had following setup:
[111]
type=friend
username=111
context=from-111
host=11.22.33.44
dtmfmode=auto
qualify=yes
wins mallow a écrit :
On Wed, 2010-01-27 at 11:47 +0100, Administrator TOOTAI wrote:
[...]
Check your sip.conf
allowguest=no
Guest are allowed and going to a different context. Logs are showing
that calls are going out to the from-111 context, so its this account
which
Olle E. Johansson a écrit :
27 jan 2010 kl. 11.47 skrev Administrator TOOTAI:
Hi,
we had an attack on a server and we don't understand how it was
possible, Asterisk 1.4.28/Debian Lenny 5.1 Attacker came from PALTEL,
network 188.161.128.0/18
Hacked account had following setup:
[111
Hi Kevin
Kevin P. Fleming a écrit :
[...]
This conversation brings to mind two possible ways we could improve
Asterisk to help users from falling into this trap:
1) When a sip.conf entry is defined as 'type=friend' *and* has a
specific host IP address (not dynamic), we could just ignore the
Hi,
is someone able to provide inbound DID for South America, at least
Bolivia, Colombia, Panama and Venezuela.
Please contact me of list, thanks
Regards
--
Daniel
--
_
-- Bandwidth and Colocation Provided by
Myles Wakeham a écrit :
[...] Are there tools or
add-ons available for this that will email me when a SIP registration
goes offline?
Any suggestions for this would be greatly appreciated.
Hi Myles,
first, best wishes to the list for this new 2010 year.
To answer your question, you
Administrator TOOTAI a écrit :
Hi,
I installed AsteriskNow and upgraded FreePBX to 2.6.0. In a sip
extension definition, when I set language, it is not reported in the
extensions_custom.conf file (eg language=xx).
Am I missing something or is it not the right way to set language?
Hello
Hi,
I installed AsteriskNow and upgraded FreePBX to 2.6.0. In a sip
extension definition, when I set language, it is not reported in the
extensions_custom.conf file (eg language=xx).
Am I missing something or is it not the right way to set language?
BTW, is this a valid place for AsteriskNow
Hello,
I had an 1.4.21-2 Asterisk running on Debian/Etch with app_nv_faxdetect
running on it without any problem.
I upgraded the server to Debian/Lenny and Asterisk 1.4.27 and
app_nv_fax_detect is not working anymore: on an incoming call,
application is launched and never exit :-(
I
Lee Howard a écrit :
In your sip.conf file allowguest defaults to yes. This means that
anyone that can reach the SIP ports on that system has access to make
unauthenticated calls, by default. The administrator actually has to go
in and turn it off to prevent unauthenticated SIP calls (in
ABBAS SHAKEEL a écrit :
Hello
Hi
I am thinking to develop a softphone that is integrated into web.(in form of
APPLET or some thing else)
Ie a user with with just a PC with Net Browser(fire fox etc) Installed can
make call..
Is there some thing developed before like this that is open
Hi,
after having tested SFA in august, I didn't use it for some times and
now I receive the subject error when calling through Skype channel.
Has anyone an idea on what can be the problem?
Thanks
--
Daniel
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Alex Samad a écrit :
Hi
how do i set the call-limit on a dahi line - its connected to the pstn
network - shared fax line. How do i tell asterisk not to send more than
1 call there !
exten = _XXX.,20(Start),Set(GROUP()=PSTN)
exten = _XXX.,n,GotoIf($[${GROUP_COUNT(PSTN)}=0]?lineOpen)
Hello,
with Asterisk 1.6.1.6 I try to hangup a call if called extension is not
existing. For this purpose I would use the internal i extension but
seems not to work.
[MyContext]
exten = s,1,NoOp(Call is treated as it should)
exten = s,n,NoOp(next step)
exten = s,n,NoOp(aso ...)
exten =
Miguel Molina a écrit :
[...]
The 'i' extension only works in applications like Background(),
WaitExten() and everything that uses DTMF to route extensions within a
context.
Well, from reading voip.org it's not really clear than ...
[...] Because the call is not
accepted there's no need
Hello everybody,
I try to install -Ubuntu 8.04 server- a B410P and a TDM2400P together
with Asterisk 1.4.26-2, dahdi-linux-2.2.0.2 and dahdi-tools-2.2.0.
Problem I face is the following one:
CLI module load chan_dahdi.so
== Registered application 'DAHDISendKeypadFacility'
== Registered
Rob a écrit :
Yes ... as a matter of fact here is the sip.conf ... obviously private info
removed
[...]
Did you try to call Gizmo numbers to see if you have success with them?
** Hear your Gizmo5 number repeated back to you.
*0 Test your router's SIP compatibility.
411 The
randulo a écrit :
Hi,
Hello
I've tried two SIP clients so far and both have unusable outgoing
audio quality.
[...]
Anyone have any recommendations?
I made few test with various client, Sip and IAX, on iPhone first
generation:
. frings: good quality but to much delay. Also I don't
Karl Fife a écrit :
[...] there are times when I want to send the call to another context in its
original un-reformatted state. Naturally the ${EXTEN} variable has been
changed. It occurred to me to use CALLERID(DNID) as such:
exten =
Kayton Sapale a écrit :
Hi all,
HI alone :-)
Thanks to the previous replies that helped me with this before, but I
got side-tracked in the middle of trying to figure this out, so
apologies for posting the same issue. I use a Nokia e71, with an
asterisk server and am having an issue
Rob a écrit :
Hi all,
Hi
I'm using GIZMO with my asterisk (1.4.13) box ... I've had CALL IN for a
while and it works fine I just added CALL OUT ... I have no problem
with call setup ... the called party hears me ... but I can't hear them
again if the call comes INTO the server
Kayton Sapale a écrit :
Thanks Daniel. It looks like I didn't paste everything into the
email, but not sure if this will make a difference:
No need to send agian the same datas, I cutted non relevant part in my
answer.
From your other mail I'm sure that your problem is dialplan related.
Doug Lytle a écrit :
Lutgring, Sam wrote:
I have an IAX trunk configured between 2 Asterisk servers. Everything
is working great except if the caller presses # during the call. If
they press # the local PBX comes on and says transferring and tries to
transfer to a blank extension.
Marco Sambo a écrit :
Hi all,
I've a problem: I update my asterisk to version 1.4.25, and the attended
transfer doesn't work.
[...]
Marco,
attented transfer are broken in 1.4.25, please upgrade to 1.4.26 (see
changelog).
--
Daniel
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