Hello,
1. is it possible that Asterisk does not translate between codecs H263
and H264 ?
2 If I set videosupport=yes in sip.conf [general], can I turn off the
video support on a peer ?
Kind regards,
Jonas.
--
_
-- Bandwi
On 12/08/2010 02:48 PM, Alex Saavedra wrote:
Jonas,
I've been using H.264 and H.263+ with a few Grandstream GVX3140. When
using H.264 the image quality was better, and required bandwidth
appeared lower compared with H.263+. In fact H.264 is expected to
consume less bandwidth for as much as 50
Hello list,
what is the difference between these 2 codecs ?
What codec to choose if bandwith is an issue ? (like in most cases I guess)
Kind regards,
Jonas.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital
On 12/07/2010 12:18 PM, Andrew Thomas wrote:
> For the Yealink - you can use a 'remote' XML file. The XML is stored on
> a web server and is retrieved by the phone every time you press the
> phones 'key'. This has the advantage of not needing the directory to be
> pushed to the handset - and the
Hello list,
I'm trying to set up a video call from my Ekiga client to a Grandstream
GXV3140 IP-phone. The call succeeds but there is no video.
I have in sip.conf :
videosupport=yes
disallow=all
allow=alaw
allow=g726
allow=g729
allow=gsm
allow=h261
allow=h263
allow=h263p
allow=h264
The Grand
On 12/02/2010 04:31 PM, Ishfaq Malik wrote:
> On Thu, 2010-12-02 at 16:01 +0100, Jonas Kellens wrote:
>
>> On 12/02/2010 03:47 PM, Ishfaq Malik wrote:
>>
>>> On Thu, 2010-12-02 at 15:19 +0100, Jonas Kellens wrote:
>>>
>>>
>>
On 12/02/2010 04:33 PM, Ishfaq Malik wrote:
> On Thu, 2010-12-02 at 16:01 +0100, Jonas Kellens wrote:
>
>> On 12/02/2010 03:47 PM, Ishfaq Malik wrote:
>>
>>> On Thu, 2010-12-02 at 15:19 +0100, Jonas Kellens wrote:
>>>
>>>
>>
On 12/02/2010 03:56 PM, Gordon Henderson wrote:
> On Thu, 2 Dec 2010, Jonas Kellens wrote:
>
>
>> Hello,
>>
>> I have Snom, Cisco, Grandstream& YeaLink phones.
>>
>> Is there a way to push a centralized phone book to these phones ??
>>
On 12/02/2010 03:47 PM, Ishfaq Malik wrote:
> On Thu, 2010-12-02 at 15:19 +0100, Jonas Kellens wrote:
>
>> Hello,
>>
>> I have Snom, Cisco, Grandstream& YeaLink phones.
>>
>> Is there a way to push a centralized phone book to these phon
Hello,
I have Snom, Cisco, Grandstream & YeaLink phones.
Is there a way to push a centralized phone book to these phones ??
Kind regards,
Jonas.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
Hello list.
This is not a life-threatening question, but still quite important for
debugging.
I have the following crontab :
15 0,8,12,17 * * * /usr/sbin/asterisk -rx 'logger rotate'
Because I have debug level 9, logfiles get quite large.
I notice that the rotation of the logfiles goes to pl
place.
William Stillwell
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Wednesday, November 24, 2010 11:44 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] kernel:
Hello list,
I'm experiencing a lot of server freezes lately. The server just... freezes.
I notice in the log files (/var/log/asterisk/messages &
/var/log/messages) that logging stops at the time the server hangs.
Logging continues when the server has been restarted (which is the only
solution
On 11/24/2010 10:28 AM, --[ UxBoD ]-- wrote:
Hello,
I notice that the following proces is running :
astcanary /var/run/asterisk/alt.asterisk.canary.tweet.tweet.tweet 1527
What is this ??
Kind rega
Hello,
I notice that the following proces is running :
astcanary /var/run/asterisk/alt.asterisk.canary.tweet.tweet.tweet 1527
What is this ??
Kind regards,
Jonas.
--
_
-- Bandwidth and Colocation Provided by http://www.api-
On 11/09/2010 03:20 PM, Gareth Blades wrote:
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
:
exten => _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
exten => _367,n,Set(LIMIT_WARNING_FILE=/path_to_your_audiofiles/file)
# do not add any extension!
exten => _367,n,Dial(SIP/test6,,L(11000,5000,5000))
Am 11.11.2010 10:31, schrieb Jonas Kellens:
Hello,
Limiting the call duration with the
Hello,
Limiting the call duration with the L-option of the Dial()-command is
working fine, however the announcement is not played.
Dialplan :
exten => _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
exten => _367,n,Dial(SIP/test6,,L(11000,5000,5000))
The call lasts for 11 seconds, but 5 minutes befo
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 t
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 2010, the phone does not
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 2010, the phone does not register to server 2... No mather how long
I wait,
On 11/06/2010 09:18 PM, Sherwood McGowan wrote:
> On Sat, Nov 6, 2010 at 2:45 PM, Jonas Kellens
> wrote:
>
>> On 11/06/2010 07:18 PM, Tilghman Lesher wrote:
>>
>>> On Saturday 06 November 2010 11:22:06 Jonas Kellens wrote:
>>>
>>>
On 11/06/2010 07:18 PM, Tilghman Lesher wrote:
> On Saturday 06 November 2010 11:22:06 Jonas Kellens wrote:
>
>> Hello,
>>
>> I just experienced a spontaneous reboot of Asterisk. This is my log file
>> /var/log/messages :
>>
>> Nov 6 16:37:37 vps2
Hello,
I just experienced a spontaneous reboot of Asterisk. This is my log file
/var/log/messages :
Nov 6 16:37:37 vps2301 kernel: miniserv.pl invoked oom-killer:
gfp_mask=0x201d2, order=0, oomkilladj=0
Nov 6 16:37:37 vps2301 kernel:
Nov 6 16:37:37 vps2301 kernel: Call Trace:
Nov 6 16:37
On 11/01/2010 10:58 AM, Gareth Blades wrote:
> Those SRV records are wrong. You have to specify both servers with
> different priorities against the same hostname. You have sip and sip2
> defines with different SRV records so whichever one you configure on the
> phone thats the only record it will
Hello,
this is a test to add a channel to multiple GROUPs.
this is my dialplan :
exten => s,n,NoOp(groepcount = ${GROUP_COUNT(40)})
exten => s,n,Set(GROUP(40)=40)
exten => s,n,NoOp(This channel is member of : ${GROUP_LIST()})
exten => s,n,NoOp(groepcount = ${GROUP_COUNT(40)})
exten => s,n,NoOp
On 11/01/2010 10:58 AM, Gareth Blades wrote:
Those SRV records are wrong. You have to specify both servers with
different priorities against the same hostname. You have sip and sip2
defines with different SRV records so whichever one you configure on the
phone thats the only record it will seen.
Hello,
I have this in my dialplan :
exten => s,n,Set(vgLabel=vg(${number}+1))
exten => s,n,GoTo(${vgLabel})
But in stead of vgLabel becoming the SUM of 2 numbers, it is just a string :
[Nov 3 16:17:27] -- Executing [...@macro-f:43]
Set("SIP/test-0002", "vgLabel=vg(1+1)") in new stack
RFC states that the client should periodically check and
switch back to the primary server is it becomes reachable.
Jonas Kellens wrote:
Hello,
I know YeaLink for example supports this...
Can you tell me for sure that when the production Asterisk server
becomes reachable again, the
Hello,
any more input on this subject ?!
Kind regards,
Jonas.
Original Message
Subject:Re: [asterisk-users] SIP client floods port 5060 and gets
blocked
Date: Thu, 28 Oct 2010 13:42:12 +0200
From: Jonas Kellens
To: Asterisk Users Mailing List - Non-Commercial
On 10/28/2010 12:52 PM, Gordon Henderson wrote:
> On Thu, 28 Oct 2010, Jonas Kellens wrote
>> On 10/28/2010 10:44 AM, Kevin Keane wrote:
>>
>>> I assume that you checked and the remote IP is a legitimate IP phone? If
>>> not, it could be an attempt to break i
On 10/28/2010 10:44 AM, Kevin Keane wrote:
I assume that you checked and the remote IP is a legitimate IP phone?
If not, it could be an attempt to break into your system.
If it is a legitimate IP phone, make sure that the SIP configuration
is correct -- if the SIP authentication fails, you c
Hello,
Is there any reason why an IP-phone would pounder on port 5060 ? My
firewall blocks the public IP because it thinks the remote IP is port
scanning on port 5060.
I think the phone is just registering but for some reason it does this
repeatedly in a very short time.
Oct 28 09:01:48 a
On 10/27/2010 01:55 PM, Andrew Latham wrote:
> Jonas
>
> A quick look at the snom wiki will tell you that I am right...
>
At what page are you looking then ??
I only see : http://wiki.snom.com/Settings/http_scheme
Jonas.
>> On 10/26/2010 06:30 PM, Andrew Latham wrote:
>>
>>> snom p
On 10/27/2010 10:06 AM, Steve Totaro wrote:
On Tue, Oct 26, 2010 at 11:31 AM, Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
Hello,
has anyone experience with auto provisioning IP-phones on
different locations through a central public provisioning server ?
You us
On 10/26/2010 06:30 PM, Andrew Latham wrote:
> snom phones can do http digest authentication...
>
I think this "digest authentication" is for accessing the phone's web
interface, not for contacting a provisioning server
Jonas.
--
__
On 10/26/2010 05:52 PM, bakko wrote:
> Hello,
>
> many SIP phones offer you the possibility to provisioning them over a FTP
> connection (with username and password).
>
> Regards
>
> - Bakko
>
In this case I will want to use Snom phones. TFTP is available, but no
FTP (with indeed then a usern
On 10/26/2010 05:41 PM, Andrew Latham wrote:
> You can provision over a WAN and access-lists or iptables can limit
> the networks allowed. Define what level of security you need first.
> For further security you can use an inbound proxy and check the http
> headers for agent identification. This
On 10/26/2010 05:40 PM, Matt Desbiens wrote:
> I havent had much auto provisioning experience, however, what about
> just using IPTables to create an access list essentially for known IPs
> to connect via HTTP/HTTPS and block all other addresses. This would
> only work if the phones are coming
Hello,
has anyone experience with auto provisioning IP-phones on different
locations through a central public provisioning server ? You use http or
https ?
Is there a danger that one uses a different MAC-address in the
provisioning link to obtain SIP username / password settings ?
Kind re
Hello,
I'm trying to use SipSak to check if my Asterisk server is
available/running with the following :
sipsak -vv -s sip:usern...@sip.domain.tld -c sip:usern...@sip.domain.tld
--password guessthis --hostname XX.XX.XX.63
The SIP OPTION is received by Asterisk as follow :
OPTIONS sip:usern
o a higher priority then the
backup then the RFC states that the client should periodically check and
switch back to the primary server is it becomes reachable.
Jonas Kellens wrote:
Hello,
I know YeaLink for example supports this...
Can you tell me for sure that when the production Aster
Hello,
I know YeaLink for example supports this...
Can you 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
Hello list.
When using SIP DNS SRV to define a production Asterisk server with high
priority and a backup Asterisk server with a lower priority on this
DNS-server, will this work as follow :
- production server is reachable, so registration of the IP-phone goes
to this server
- production se
On 10/13/2010 12:09 AM, Paul Belanger wrote:
> On Tue, Oct 12, 2010 at 8:08 AM, Jonas Kellens
> wrote:
>
>> [Oct 12 14:03:32] DEBUG[9064] chan_sip.c: Trying to put 'SIP/2.0 401' onto
>> UDP socket destined for public_ip:2049
>>
>> Is som
Hello,
what does this message mean ?
[Oct 12 14:03:32] DEBUG[9064] chan_sip.c: Trying to put 'SIP/2.0 401'
onto UDP socket destined for public_ip:2049
I find this in my debug log file when "core set debug 25".
Is something failing, or is this just informative ?
Kind regards,
Jonas.
--
_
Hello,
there is a really great difference in the Via-header of the
REGISTER-message between the Zoiper and the Snom.
Also the Zoiper has a Contact-header, and the Snom REGISTER has not...
Snom :
REGISTER sip:sip.domain.tld SIP/2.0
_*Via: SIP/2.0/UDP 192.168.114.200:2049;branch=z9hG4bK-p4ayhth
On 10/07/2010 06:50 PM, Daniel Tryba wrote:
> On Thu, Oct 07, 2010 at 02:57:27PM +0200, Jonas Kellens wrote:
>
>> nat=yes is set as a global parameter and also in the realtime MySQL
>> sip_buddies database I have for every peer nat=yes.
>>
>> I then find it very s
On 10/07/2010 04:18 PM, Philipp von Klitzing wrote:
> Hi!
>
>
>> I'm having difficulty with registering a SIP account in a Snom 320 IP-
>> phone.
>>
> Do a SIP trace on your SNOM phone, and you will most probably see that
> the 401 reply of Asterisk does not arrive on the phone. Then chec
On 10/07/2010 02:36 PM, Daniel Tryba wrote:
> On Thu, Oct 07, 2010 at 02:24:59PM +0200, Daniel Tryba wrote:
>
>>> It's the same account, the same password, but other agent.
>>>
>>> Can anyone help me with this please ?! I see no difference but there
>>> must be !!
>>>
>> The difference
On 10/07/2010 02:36 PM, Daniel Tryba wrote:
> On Thu, Oct 07, 2010 at 02:24:59PM +0200, Daniel Tryba wrote:
>
>>> It's the same account, the same password, but other agent.
>>>
>>> Can anyone help me with this please ?! I see no difference but there
>>> must be !!
>>>
>> The difference
On 10/07/2010 02:36 PM, Daniel Tryba wrote:
> On Thu, Oct 07, 2010 at 02:24:59PM +0200, Daniel Tryba wrote:
>
>>> It's the same account, the same password, but other agent.
>>>
>>> Can anyone help me with this please ?! I see no difference but there
>>> must be !!
>>>
>> The difference
, and then when I hook
them up at the site there is trouble with nat. I'm also behind nat here...
I do not find an rport-parameter in the snom's webinterface...
Jonas.
On 10/07/2010 02:24 PM, Daniel Tryba wrote:
On Thu, Oct 07, 2010 at 01:54:58PM +0200, Jonas Kellens wrote:
It&
Hello,
I'm having difficulty with registering a SIP account in a Snom 320
IP-phone. This is what sip debug tells me :
[Oct 7 13:28:42] VERBOSE[20314] chan_sip.c: [Oct 7 13:28:42]
<--- SIP read from UDP:public_ip:58697 --->
REGISTER sip:sip.domain.tld SIP/2.0
Via: SIP/2.0/UDP 192.168.114.200
Hello list,
this works :
exten => _*XXX*,n,SIPAddHeader("Call-Info:\; answer-after=0")
exten => _*XXX*,n,Dial(SIP/${SIPACCOUNT})
The phone auto-answers the call...
this does not work :
exten => _*XXX*,n,SIPAddHeader("Call-Info:\; answer-after=0")
exten => _*XXX*,n,Page(SIP/${SIPACCOUNT})
Th
On 09/30/2010 12:16 PM, Jonas Kellens wrote:
Hello list,
I get the following error :
pbx_extension_helper: No application Page for extension
Apparently I have no timing source installed.
But I thought that Dahdi did not need to be installed for timing ?!
And that there is some internal
Hello list,
I get the following error :
pbx_extension_helper: No application Page for extension
Apparently I have no timing source installed.
But I thought that Dahdi did not need to be installed for timing ?! And
that there is some internal timing in Asterisk 1.6.2.10 ?
Kind regards,
Jona
Hello list,
how can I go from *100* to 100 ?
I know I can do something like ${EXTEN:1} but that way I only get rid of
just one *.
Kind regards,
Jonas.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
On 09/22/2010 02:45 PM, Philipp von Klitzing wrote:
> .slin is not .wav
>
Other files that are also in wav format play without any problem :
[Sep 22 15:02:35] -- Playing
'vm-youhave.slin' (language 'nl')
[r...@asterisk16 asterisk-1.6.2.10]# ls -l /var/lib/asterisk/sounds/nl/
total 388
On 09/22/2010 01:38 PM, Watkins, Bradley wrote:
> This is indicative that you have set the channel's language to something
> that expects there to be a singular and plural version of the 'new' (as
> in 'one new message' versus 'five new messages') sound.
>
> According to the code, that includes Dut
Hello list,
it seems that a sound file is not present on my system, although I have
made a standard install...
[Sep 22 12:28:51] WARNING[22117]: file.c:650 ast_openstream_full: File
vm-INBOXs does not exist in any format
[Sep 22 12:28:51] WARNING[22117]: file.c:953 ast_streamfile: Unable to
On 09/21/2010 04:22 PM, Jon Farmer wrote:
> On 16 September 2010 22:23, Barry Miller wrote:
>
>
>> For an interim fix, setting res_agi=1.4 in the [compat] section of
>> asterisk.conf should work. See UPGRADE-1.6.txt .
>>
> I have tried this but it still complains about the pipe not bein
Hello list,
I read this in sip.conf :
notifyringing = no ; Control whether subscriptions already
INUSE get sent RINGING when another call is sent (default: yes)
What does this mean ?!
Does this mean that when I mark this as "yes", a phone that already has
taken a call will be se
On 09/17/2010 06:00 PM, Mark Deneen wrote:
On Fri, Sep 17, 2010 at 11:51 AM, Jonas Kellens
wrote:
On 09/17/2010 05:29 PM, Mark Deneen wrote:
On Fri, Sep 17, 2010 at 4:21 AM, Jonas Kellens
wrote:
warning: exec file is newer than core file.
Jonas,
I
On 09/17/2010 05:29 PM, Mark Deneen wrote:
> On Fri, Sep 17, 2010 at 4:21 AM, Jonas Kellens
> wrote:
>
>> warning: exec file is newer than core file.
>>
> Jonas,
>
> I encourage you to read the output. Did you run gdb with a core file
> dumped from the
On 09/16/2010 07:58 PM, Paul Belanger wrote:
> Please do not send me direct email, post them to the list for others
> to help. Your backtrace is optimized (). You
> need to reinstall asterisk with DONT_OPTIMIZE enabled, described in
> doc/backtrace.txt.
>
Hello,
I have compiled with DONT_OP
On 09/16/2010 05:45 PM, Paul Belanger wrote:
> On Thu, Sep 16, 2010 at 8:19 AM, Jonas Kellens
> wrote:
>
>> I get so little output :
>>
>>
> You are still doing it incorrectly. As said, doc/backtrace.txt has all
> the required information.
>
b
On 09/16/2010 12:41 PM, Philipp von Klitzing wrote:
> Hi!
>
>> Does this shine new light to the problem ?!
>>
> No. Once more: Go and read doc/backtrace.txt.
>
> And check if you have any meaningful information in /var/log/messages for
> the timestamp when asterisk crashed.
>
> Philipp
>
010 06:49 PM, jon pounder wrote:
On 09/15/2010 12:42 PM, Leif Madsen wrote:
On 10-09-15 05:25 AM, Jonas Kellens wrote:
I think I've found it :
Asterisk always reboots on this part :
[Sep 15 11:16:32] -- Goto (azura,pbx,1)
[Sep 15 11:16:32] -- Executing [...@azura:1]
NoOp("
Hello list,
are there special things that needs to be done when converting BLF from
asterisk 1.4 tot 1.6.2 ?!
I have replaced call-limit with call-counter, but it seems that the
lights on the phone no longer give the status of the extension they monitor.
On Snom phones, when the lights shou
On 09/15/2010 09:41 PM, Dan Journo wrote:
Hi,
I think ive found a bug but need someone to double check.
Whenever I issue a "reload" in Asterisk, any realtime extensions stop
receiving calls.
I have to reboot the sip phones in order to get them to re-register.
Can anyone see if they have a
On 09/15/2010 03:47 PM, Zeeshan Zakaria wrote:
>
> Hi,
>
> I went over your dialplan and though it looks fine at first glance,
> but because I have no experience with Asterisk 1.6, so I would like to
> ask if commas in mysql query are ok without escape character? In my
> asterisk 1.4 I would typ
On 09/15/2010 02:45 PM, Steve Howes wrote:
On 15 Sep 2010, at 13:22, Jonas Kellens wrote:
I have indeed found the core file in /tmp (that is where 'locate' does
not look huh...)
'updatedb'?
S
Off course I did that, Steve, before I did a locate on 'c
On 09/15/2010 02:03 PM, Philipp von Klitzing wrote:
> Hi!
>
>
>> I know I post a lot concerning this issue, but this is because this
>> problem occurs on a production system and I feel very hot breathing down
>> my neck.
>>
> Why not reduce the pressure and revert to 1.4.30 for the produc
On 09/15/2010 12:59 PM, Gareth Blades wrote:
> I cant help you with fixing the actual cause but have you considered
> moving the mysql and as much of the associated logic to an AGI running
> something like a perl or php script. From previous posts that generally
> seems to me the more reliable way
Hello Philipp,
I know I post a lot concerning this issue, but this is because this
problem occurs on a production system and I feel very hot breathing down
my neck.
I have tested during several weeks my implementation on a test system
which is similar to the production system. The only diffe
)
exten => s,n,Set(vakantieresult=continue)
exten => s,n,MacroExit
exten => s,n(opvakantie),NoOp(op vakantie !)
exten => s,n,GoToIf($["${NA}"="hangup"]?hangup:route)
Do you guys see why Asterisk has problems with this part of the dialplan ?!
Jonas.
On 09/15/20
calls if the
amount of free memory falls below this watermark
So I don't think that asterisk restarts/reloads because of the above
settings ?!
Jonas.
On 09/15/2010 10:36 AM, Jonas Kellens wrote:
The reboot occured a 10:11:11, this my debug log :
...
[Sep 15 10:11:11] DEBUG[12353]
ANINFO' in tree 'description'
[Sep 15 10:11:37] DEBUG[12434] xmldoc.c: Cannot find variable 'IAXPEER'
in tree 'description'
[Sep 15 10:11:37] DEBUG[12434] xmldoc.c: Cannot find variable 'IAXVAR'
in tree 'description'
[Sep 15 10:11:37] DEBUG[1
risk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, September 14, 2010 2:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11
On 09/14/2010 09:12 PM, Carlos Chavez wrote:
On Tue, 2010-09-14
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, September 14, 2010 2:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11
On 09/14/2010
On 09/14/2010 09:21 PM, Danny Nicholas wrote:
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Tuesday, September 14, 2010 2:05 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re
On 09/14/2010 09:12 PM, Carlos Chavez wrote:
> On Tue, 2010-09-14 at 20:27 +0200, Jonas Kellens wrote:
>
>> And again !! Without me doing anything !!
>>
>> PBX Core settings
>> -
>>Version: 1.6.2.11
>>Bui
Core verbose is 9 !!
What info can I give to you ?! What info can I get from my debug log ?!
I don't see anything remarkable... What am I looking for ?!
Jonas.
On 09/14/2010 08:50 PM, Paul Belanger wrote:
On Tue, Sep 14, 2010 at 2:27 PM, Jonas Kellens wrote:
And again !! Witho
Can you then please tell me what kind of debugging I need to enable ?!
On 09/14/2010 08:41 PM, Steve Howes wrote:
On 14 Sep 2010, at 19:27, Jonas Kellens wrote:
And again !! Without me doing anything !!
Yea, you didn't even enable any kind of debugging or anything. Am
: 0
Maximum load average:0.00
Minimum free memory: 0 MB
* Startup time:20:24:51
Last reload time:20:24:51*
Jonas.
On 09/14/2010 03:08 PM, Jonas Kellens wrote:
On 09/14/2010 02:30 PM, Jonas Kellens wrote:
Hello list,
has anyone
lease tell me if this is a setting or if this is a really really major
bug !
Jonas.
On 09/14/2010 03:08 PM, Jonas Kellens wrote:
On 09/14/2010 02:30 PM, Jonas Kellens wrote:
Hello list,
has anyone else also noticed spontaneous reboots ?!
I noticed this today and also yesterday. Can't rea
On 09/14/2010 02:30 PM, Jonas Kellens wrote:
Hello list,
has anyone else also noticed spontaneous reboots ?!
I noticed this today and also yesterday. Can't really see if there is
a fixed time between the reboots.
Normally al of my SIP peers are registered. When I put up the CLI
today
Hello list,
has anyone else also noticed spontaneous reboots ?!
I noticed this today and also yesterday. Can't really see if there is a
fixed time between the reboots.
Normally al of my SIP peers are registered. When I put up the CLI today
I saw that a lot of SIP accounts where UNREACHABLE a
kups every seconds
; default is 300 (5 minutes)
But I still have these messages...
Jonas.
On 09/13/2010 08:38 AM, Jonas Kellens wrote:
Hello list,
my CLI is spammed with :
[Sep 13 08:31:38] > doing dnsmgr_lookup for 'ssw6.itsp.tld'
[Sep 13 08:31:38] > doi
arg2")
If anyone wants to update the very old wiki, be my guest.
Kind regards,
Jonas.
On 09/13/2010 08:25 PM, Roger Burton West wrote:
On Mon, Sep 13, 2010 at 08:15:34PM +0200, Jonas Kellens wrote:
[Sep 13 20:14:59] -- Launched AGI Script
/var/lib/asterisk/agi-bin/cleanpick
On 09/13/2010 06:01 PM, Carlos Chavez wrote:
On Mon, 2010-09-13 at 17:48 +0200, Jonas Kellens wrote:
Hello,
can anyone please tell me how I can give arguments to my AGI script ?!
I think asterisk sees the name of the AGI + the channel as one
filename, and of course this file then does not
Hello,
can anyone please tell me how I can give arguments to my AGI script ?!
I think asterisk sees the name of the AGI + the channel as one filename,
and of course this file then does not exist.
Jonas.
On 09/13/2010 10:26 AM, Jonas Kellens wrote:
Hello list,
what is the correct syntax
Hello list,
what is the correct syntax ?
exten => s,n,Queue(${queuename}${timeout},cleanpickup.agi^${CHANNEL})
[Sep 13 10:23:58] WARNING[23551]: res_agi.c:886 launch_script: Failed to
execute
'/var/lib/asterisk/agi-bin/cleanpickup.agi^SIP/329909007906-017a':
File does not exist.
Hello list,
my CLI is spammed with :
[Sep 13 08:31:38] > doing dnsmgr_lookup for 'ssw6.itsp.tld'
[Sep 13 08:31:38] > doing dnsmgr_lookup for 'ssw6.itsp.tld'
[Sep 13 08:31:47] > doing dnsmgr_lookup for 'ssw4.itsp.tld'
[Sep 13 08:31:48] > doing dnsmgr_lookup for 'ssw4.itsp.tld'
[Sep 13 08:31:49] >
Hello,
everything goes well on asterisk 1.4.30, but with asterisk 1.6.2.11 I
get the following :
[Sep 12 18:59:29] WARNING[2066]: chan_sip.c:12738 check_auth: username
mismatch, have <32990900>, digest has <3291119600>
[Sep 12 18:59:29] NOTICE[2066]: chan_sip.c:20082 handle_request_invite
Hello list,
I'm using asterisk 1.6.2.11 with realtime SIP (mysql DB).
I notice that when the SIP peer registers, the fields 'fullcontact',
'ipaddr', 'port', 'regserver', 'regseconds', 'lastms' are filled with
values.
But after a while, these fields become empty.
Asterisk CLI shows :
asteri
On 09/09/2010 05:37 PM, Paul Belanger wrote:
> On Thu, Sep 9, 2010 at 10:04 AM, Jonas Kellens
> wrote:
>
>> asterisk*CLI> core show application Dial
>>
>>
> did you have libxml-doc installed when you build asterisk?
>
> *CLI> module load app_
On 09/09/2010 04:12 PM, Danny Nicholas wrote:
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Thursday, September 09, 2010 8:56 AM
Hello list,
how come on my Asterisk 1.6.2.11, I have no help available ?!
asterisk*CLI> core show application Dial
-= Info about application 'Dial' =-
[Synopsis]
Not available
[Description]
Not available
[Syntax]
Not available
[Arguments]
Not available
[See Also]
Not available
Kind re
401 - 500 of 866 matches
Mail list logo
