Il 11/07/2014 16:45, Rusty Newton ha scritto:
On Wed, Jul 9, 2014 at 3:55 PM, Massimo Nuvoli mass...@archivio.it wrote:
I found a very strange proble whit two asterisk servers in the same network.
Scenario
Asterisk A with extensions 5XX
Asterisk B with extensions 2XX
There is NO link between
I found a very strange proble whit two asterisk servers in the same network.
Scenario
Asterisk A with extensions 5XX
Asterisk B with extensions 2XX
There is NO link between the two asterisks.
Call from 501 to 503, 503 ringing
Call from 201 to 203, 203 ringing
The 202 extension comand a
I need in a strange applicatio a way to detect the tone (busy, ring
etc. etc.) of analog line (zap channel), while channel UP.
I found the application NV line detect, but is very old, and may be
not mantained.
I can patch asterisk to actually support this application but i think
someone other
://www.ilovetovoip.com
On 2010-07-07 4:16 AM, Massimo Nuvoli mass...@archivio.it
mailto:mass...@archivio.it wrote:
I found a strange thing on Asterisk 1.4.32, the same defect on 1.4.26.?
I spend 4 hours to try to solve... but found only a workaround.
As is easy to reproduce the problem i
bruce bruce ha scritto:
Hi Guys,
I am looking to buy a 25 Watt output CyberData VoIP amplifier and to use
2 Bogen sp308a speakers with it for a 40, 000 squar feet area and 21
feet height. Is that enough? Is there calculator online I can use to
determine the number of speakers needed? I
I found a strange thing on Asterisk 1.4.32, the same defect on 1.4.26.?
I spend 4 hours to try to solve... but found only a workaround.
As is easy to reproduce the problem i need to know if this is a bug or
if there is some idiot configuration that i miss.
Maybe also the bug is know...
This is the problem:
Call coming into a queue in ringall strategy, if a member (SIP) of the
queue is busy when entering the queue, and this member comes free
after a little time, the member never rings..
How to solve this?
I changed all parameters of the queue with no results...
Wath i need:
Pascal Bruno ha scritto:
Hi,
I am experiencing a weird issue with my MV-372.
Mobile1 Mobile2 are both registered to my asterisk server, I am able
to use them for outgoing call with no problem, but when I call the sims
in my gateway, they are routed to the right
Pascal Bruno ha scritto:
This way the gateway does not have to register, and I can keep the
settings that passes the right caller id. Another way would be to have
asterisk read another field for the caller id, because the number of the
caller is somewhere on the sip invite.
ouch :-) sorry
John Novack ha scritto:
Not sure how you would do that, as the X100 card is an FXO card,
won't provide either battery or dial tone to the cordless. What you
will want for that is an FXS card or ATA. The X100 card will
connect to a central office line, and with the later software echo
Julian Lyndon-Smith ha scritto:
If I have the following in the dialplan
exten = foo,n,Dial(SIP/1234Zap/G1c/55443322)
and SIP/5432 calls this extension,
is it possible to show different callerid numbers to each of the target
numbers ?
The reason I ask is that if the call is from an
broadband Voice ha scritto:
Irqbalance was causing the the processor handling the interrupts of
the zap cards to change very often.
This would impose a delay during the change and cause the zttest
numbers to drop/be inconsistent.
Irqbalance is a good idea BUT some kernel
Steven ha scritto:
zttest:
--- Results after 44 passes ---
Best: 100.00 -- Worst: 99.987793 -- Average: 99.988350
Dell 2950
Wildcard TE410P (2nd Gen) (rev 02) (with echo cancel daughter board)
1 PRI configured to Telco
1 PRI configured to old Panasonic DBS 576 being used just as a
broadband Voice ha scritto:
Does anyone have any compatibilty issues with Dell *PowerEdge^TM 2950 III
2-Socket, Quad-Core 2U*? I plan on using this with the Digium T1 cards.
Thanks.
Consider 2 socket dual core CPU with more mhz, Asterisk is more IO
than computation. The quad core CPU is
Benjamin Jacob ha scritto:
Hello ppl,
Searched all over, but couldn't find anything conclusive.
Does an off-the-shelf version of Asterisk run without any issues on a
64-bit machine?
Does anyone have any 'conclusive' figures?
Apologies if this is a repeat question. Would appreciate if I
gincantalupo ha scritto:
Hi,
somtimes my Gigaset 450IP loses its registration.
Is there anybody who knows why and how to solve it?
TIA
Giorgio Incantalupo
I try some trick and i found:
maxexpiry=120
defaultexpiry=120
in sip.conf
I put this in a production env and all things are ok
I found no info about this strange error:
logger.c: No more room in scheduler
logger.c: Asked to delete sched id -1???
Only in verbose mode. Someone know how to solve this?
Asterisk 1.2.13 with sangoma A104EC
Hints?
Thnks.
begin:vcard
fn:Massimo Nuvoli
n:Nuvoli;Massimo
org:Progetto Archivio
Sean Bright ha scritto:
Does this occur in the latest 1.2.17 release?
I dont know, this is a production system with 2 pri linked to telco
and 2 pri linked to a pbx, i planned a large update but the release
in use is the 1.2.13.
And, also, i checked the changelog of the 1.2.17, and i found no
Thomas Stein ha scritto:
Hello.
Just installed the Beronet BN4S0 card. But i can't connect to my ISDN Line.
misdnportinfo gives (what does :Layer 4 protocol 0x0401 is detected, but
not allowed for TE lib mean?):
This is normal, the channel is used by asterisk and so is not
available
Morten Isaksen ha scritto:
On 3/1/07, *Kevin P. Fleming* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Tomislav Parèina wrote:
Is it possible to use Digium (or Sagnoma, or Beronet) cards with
Asterisk on Vmware?
The card manufacturer is irrelevant, as is the type of
I am at the end of a long way... i try to work with a number of isdn
boards (BRI not PRI) and i found only a lot of problems.
First, the bristuff that is near working, but not so perfect ISDN
designed interface. This is not bad but in a production environment
this solution is not usable.
Second
Thermal Wetland ha scritto:
Does anyone know if you can have multiple TE110P cards in one chassis?
One server with two TE110P, shared interrupts, APIC routed irq, all
things near ok. Sometime only one of these run HDLC error and some
strange error, i think a 0,05% probability of error.
:-)
Tomislav Parčina ha scritto:
For what we do with Asterisk(lots of meetme and Zap - IAX2) It
does spread the load across both cores. In our initial
comparisons for equal call traffic, the P4-D had half or the
average loadavg for a 6 hour time period of the P4 of the same
speed.
MATT---
Hi
jurgen ha scritto:
Hi,
The problem happens when I record a call using MixMonitor. Even though
it's recording natively in g729, a single call uses 2 decoders and one
encoder! The only explanation I can think of for that is that
MixMonitor is transcoding the g729 streams to something else,
Ira ha scritto:
At 11:23 PM 8/2/2006, you wrote:
I'm just wondering how many users are willing to have the caller wait an
additional 2 rings (in addition to the 3 audible rings for the Asterisk
receiving phone). This seems like something that should have some sort of
workaround. No?
You
Sebastian Reitenbach ha scritto:
I found the same indentical problem, the trouble was the switchtipe, i
am using national and i switched to unknown.
is unknown allowed for switchtype?
when I take a look here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf
then
Sebastian Reitenbach ha scritto:
but when I issue a reload chan_zap in the asterisk console, then I can see
the
following in the log output:
Jul 21 14:20:16 WARNING[1345] chan_zap.c: Ignoring signalling
Jul 21 14:20:16 WARNING[1345] chan_zap.c: Ignoring switchtype
Jul 21 14:20:16
Sebastian Reitenbach ha scritto:
so about 80% of the incoming calls work well, but especially with one sender
we have a problem, there is always the last digit missing. This is a 1-800
service in the US, forwarding the call to our asterisk. As a workaround I
configured it to call to
Lincoln Zuljewic Silva ha scritto:
Hello all. I have a Digium TE110P board and when I do a zap show
status on CLI I get:
DescriptionAlarms
IRQbpviolCRC4
Digium Wildcard TE110P T1/E1 Card 0OK00
Douglas Garstang ha scritto:
I have dialled into a Queue, and an agent has answered the call with
AgentcallbackLogin().
The agent hits #1, to transfer the call. Asterisk responds with 'Transfer',
followed by dial tone.
As soon as I enter a digit, Asterisk responds with 'I am sorry. That is
Dirk Enrique Seiffert ha scritto:
Hello,
I couldn't find any examples on the auto attendent reading responses from
a database, though it looks like a common task to me. Can anybody provide
some hints, links, directions or experiences for this kind of
configuration?
You should look at the AGI
Idris AVCI ha scritto:
I use Asus Barebone server's that can handle 120 Zap -- SIP calls. You
can find additional info on http://www.snc.com.tr/ractory_rx1ba-n.asp
which is a Asus distributor in Turkey. My server's details are;
2 X Xeon 3.0 HT Supported
2 GB RAM
2 X 250 Sata (RAID 1)
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Angelito Manansala ha scritto:
Hello List,
Can anyone here has a working configuration of any digium e1 card that is
connected to cisco 3800.
The problem is the router configuration... you need these setups to
try some configuration on the Linux
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Angelito Manansala ha scritto:
I noticed that when i reload chan_zap.so command there is a warning like
this:
== Parsing '/etc/asterisk/zapata.conf': Found
Jun 29 21:50:58 WARNING[739]: chan_zap.c:10886 setup_zap: Ignoring
switchtype
Jun 29
Jean-Michel Hiver ha scritto:
Hi,
I'd like to use the convenience of apt packaging, but debian sarge's
default asterisk is something like 1.0.7.
Are there any apt repositories which provide newer versions of the
software?
Also you can use the unstable branch of debian, all things are
This is the problem:
two Queues
Agent logged in as agentcallback and member of the two queues.
When a call come in the queue, asterisk call the extension provided
by the agentcallbacklogin.
The need is in the extension to have a variable with the queue id.
something like:
exten =
Kyle Sexton ha scritto:
Could you just set the variable in the part of the dialplan where they
enter the queue and then reference it here?
:-) very simple, tested but not working, and logically i think it is
right.
In asterisk a variable (dialplan SET) is bound to the incoming
channel, but,
Kevin P. Fleming ha scritto:
Kyle Sexton wrote:
Could you just set the variable in the part of the dialplan where they
enter
the queue and then reference it here?
Yes, that is the way to do this. Set a variable in the dialplan before
putting the _caller_ into the queue, and prefix the
Massimo Nuvoli ha scritto:
Kevin P. Fleming ha scritto:
Kyle Sexton wrote:
Could you just set the variable in the part of the dialplan where they
enter
the queue and then reference it here?
Yes, that is the way to do this. Set a variable in the dialplan before
putting the _caller_
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