Hello,
I have strange situation with asterisk 1.8.18.0 , randomly i got this
message in cli:
WARNING[15925] asterisk.c: No more connections allowed
All connections freeze and all extensions doesn't work anymore. Is any bug
or is any setting that can solve this problem?
Thank you.
Jonson.
Hi all,
I have problems dialling out because my new telco (the previous gave no
problems) tells me my PBX adds a leading 0 and that's why I cannot dial
out (but I can receive calls).
I make a small extensions.conf as a test:
exten = 666,1,Dial(DAHDI/g1/339xx)
but cannot dial out
2012/11/20 gincantalupo gincantal...@fgasoftware.com
Hi all,
I have problems dialling out because my new telco (the previous gave no
problems) tells me my PBX adds a leading 0 and that's why I cannot dial out
(but I can receive calls).
I make a small extensions.conf as a test:
exten =
Hi,
I set up tls and srtp with lyn and asterisk 1.8 before.
I think your ssl connection is not setup .
so, it may due to key and certificate problem.
If the key and cert is ok with CA, the ssl connection will up auto..
I work this before and I can connect to lync server with TLS and srtp
Hi,
just going through the code i found that this CONFIG_VOICEBUS_ECREFERENCE
undef in the voicebus then how it will defined to run the echo cancell on
the respective drievers wctdm24xxp ??
explain how this CONFIG_VOICEBUS_ECREFERENCE enabled and where it is
enabled while run time.
Hi Leandro,
thanks for your answer.
I already have tried those parameters but without any positive result.
The telco technician has tried the line with its machine and it
worked...remote telco technicians say they get a leading zero...
I'm thinking there is something strange in the middle
That is a real mistery! I like a lots these cases when all seems not
working despite all being correctly configured, but you know first or later
you'll find the answer.
From your website, it seems you are selling/renting PBX based on asterisk,
so you can be sure nobody has messed with the
Andre Gronwald wrote:
it is just because i think that something is not wrong (which is
correct, because i address a currently not existing peer). and if there
is a way to handle it better, then i would like to know it (virtual
queues is just oversized, but maybe there is a simple usage of
Jonson Player wrote:
Hello,
Hola,
I have strange situation with asterisk 1.8.18.0 , randomly i got this
message in cli:
WARNING[15925] asterisk.c: No more connections allowed
This message is output when the number of Asterisk consoles (asterisk -r
instances) has reached the limit. This
Face wrote:
Well, thanks for responding. I went back to 10.10.0 and things seem to
be working fine now!
This is certainly good to know but I'd like to know why upgrading to 11
did not seem to work for you. This is the first case since it's been out
where it doesn't appear to have been
On 11/20/2012 03:32 AM, Jonson Player wrote:
Hello,
I have strange situation with asterisk 1.8.18.0 , randomly i got this
message in cli:
WARNING[15925] asterisk.c: No more connections allowed
All connections freeze and all extensions doesn't work anymore. Is any
bug or is any setting
Hi Leandro,
I'm sure nobody has added something... tried prilocaldialplan and
pridialplan but nothing changed.
Question: if pridialplan or prilocaldialplan would work, should I see
the 0 inside PRI frame with intense debug or it is hidden?
Yes...the technician did it...there is only one
On Tue, 2012-11-20 at 15:03 +0100, gincantalupo wrote:
I'm sure nobody has added something... tried prilocaldialplan and
pridialplan but nothing changed.
Question: if pridialplan or prilocaldialplan would work, should I see
the 0 inside PRI frame with intense debug or it is hidden?
Somebody
Not only, you have to restart dahdi/zaptel as well.
Leandro
2012/11/20 Frederic Van Espen frederic...@gmail.com
On Tue, 2012-11-20 at 15:03 +0100, gincantalupo wrote:
I'm sure nobody has added something... tried prilocaldialplan and
pridialplan but nothing changed.
Question: if
You've exceeded the allowed maximum number of simultaneous remote console
connections (128). While it may be a bit aggressive to have it kill all
current connected consoles, its also a bit excessive to have
128 connected remote consoles.
While the behaviour may not be entirely desirable, this
In my past experience the best recourse for dealing with a DAHDI trunked
asterisk system is this sequence
Service asterisk stop
Service dahdi restart
Service asterisk start
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
On Tue, Nov 20, 2012 at 05:18:12PM +0530, upendra wrote:
Hi,
just going through the code i found that this CONFIG_VOICEBUS_ECREFERENCE
undef in the voicebus then how it will defined to run the echo cancell on
the respective drievers wctdm24xxp ??
explain how this
Hello
This is a question regarding whether there's any way within hangup
extensions to determine whether the caller or callee leg (or both) of a
bridged call has hung up. The test case I have is running under
Asterisk 1.8.17.0, but the behaviour is observed in 1.8.18.0 (and also
1.6.2.18).
On 11/20/2012 8:03 AM, gincantalupo wrote:
Hi Leandro,
I'm sure nobody has added something... tried prilocaldialplan and
pridialplan but nothing changed.
Question: if pridialplan or prilocaldialplan would work, should I see
the 0 inside PRI frame with intense debug or it is hidden?
Yes...the
This is a question regarding whether there's any way within hangup
extensions to determine whether the caller or callee leg (or both) of
a
bridged call has hung up. The test case I have is running under
Asterisk 1.8.17.0, but the behaviour is observed in 1.8.18.0 (and
also
1.6.2.18).
Hello Josua,
Thank you for this answers:
First of all, yes i run in crontab at 15 min an analyzing script which
collect show sip channels with asterisk -rx . This could be my problem...
I think that this commands could remain stalled and doesn't logout after
execution of command.
A friend of
Hello Danny,
Could you tell me how can i put time out at execution of remote commands
with asterisk -rx show sip channels.
I think that is my problem... after i execute asterisk -rx commands
something remain stalled and somehow i think that could block my asterisk...
I mean all new connections
Hello Matthew,
Could I rise with some option the number of simultaneous console? I don't
have simultaneous console but is good to know in case i didn't get any
other workaround to fix this problem.
Thank you.
On Tue, Nov 20, 2012 at 3:56 PM, Matthew Jordan mjor...@digium.com wrote:
On
No, you don't need your script for this.
Prevent attacks by using fail2ban to block brute force attacks using iptables,
securing your server at the OS level, and NEVER EVER EVER let leave the web GUI
for FreePBX open to the internet. I'm sure others have more suggestions.
Over the years 100%
My suggestion would be to issue a core restart when convenient around
midnight, assuming your installation doesn't do 24/7 calls.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonson Player
Sent: Tuesday, November 20, 2012 3:54 PM
Hey Longst,
I'd recommend having a look into the LUA support Asterisk offers for its
dialplans or AGI. These are the only realistic ways to add functions, unless
you want to write your own C module and compile it in. Adhearsion is an option
as well, if you are proficient with ruby.
Cheers,
I wish to ask if there is way to keep IAX trunk connection up. I have a
small server on Xen VPS but notice that my IAX trunk drops after some time.
I understand there is cron job to function as sip watchdog.
My asterisk is 11.0.1
Thanks for suggestions.
CK
--
Switching to SIP is likely your best solution. IAX is buggy. Always has
been, and I'll bet always will be.
On Tue, Nov 20, 2012 at 7:34 PM, asterisk asterisk aster...@ck-lee.comwrote:
I wish to ask if there is way to keep IAX trunk connection up. I have a
small server on Xen VPS but notice
Hi ,
it implies that under this CONFIG_VOICEBUS_ECREFERENCE code inside this
condition will never execute.
if we need it should be defined .
Regards
Upendra
On Tue, Nov 20, 2012 at 8:55 PM, Shaun Ruffell sruff...@digium.com wrote:
On Tue, Nov 20, 2012 at 05:18:12PM +0530, upendra wrote:
I upgrading to 11 because I want to use the MessageSend command from the
AMI, ver 10 dose not have MessageSend In the list of
commands. Unfortunately I remove ver 11 and I dont think I can provide the
information you asked.
On Tue, Nov 20, 2012 at 4:46 PM, Joshua Colp jc...@digium.com wrote:
Hello All,
I was wondering if somebody could elaborate the change in
translation of codecs specifically the amount of time increased in Asterisk
11. For example
*Asterisk 11*
***alaw **speex *
*gsm **15000 **15000 *
*ulaw9150 15000*
* *
*Asterisk 1.6.x*
*
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