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
Hi all,
I am currently not able to configure SNMP for asterisk, but I am not able to
acess to the asterisk MIB (the asterisk MIB is in /usr/share/snmp/mibs/)
Does somebody has an example of smnpd.conf file wich is working ?
regards
Mickael
___
--
Quoth Jon Morgan jon.mor...@motors.co.uk
We have a 2 port Digium TE220P card, one span is configured to connect to our
ISDN30 provider (British Telecom), the other span connects to our internal
PBX. Here's the zaptel.conf snip:
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
Hello Mickael
Here You have the snmpd.conf file
cat /etc/snmp/snmpd.conf
rocommunity your_community
master agentx
agentXperms 0660 0550 nobody asterisk
SNMPD_FLAGS=${SNMPD_FLAGS} -x /var/agentx/master
mibs +ASTERISK-MIB
and also you need create file /etc/snmp/snmp.conf with following entry
mibs
Hi Michal,
thanks a lot for you quick answer I appreciate.
I run your commands and I have the following answer
[localhost snmp]# snmpwalk -c local -v 1 localhost asterisk
no answer
[localhost snmp]# snmpwalk -c local -v 2c localhost asterisk
ASTERISK-MIB::asterisk = No Such Object available on
What operating system do You have ? What asterisk version You compile ?
After install net-snmp do You recompile asterisk with res_snmp module ?
I'm used instruction from here
http://voxilla.com/2009/02/03/configuring-asterisk-snmp-support-1131
and everything work correctly.
BR,
Michał
W dniu 27
I am running on Trixbox so my OS is Cent0S 5.4 and the Asterisk version is
1.4.22-4
on asterisk side Snmp module is running:
module load res_snmp.so
== Parsing '/etc/asterisk/res_snmp.conf': Found
Loading [Sub]Agent Module
Loaded res_snmp.so = (SNMP [Sub]Agent for Asterisk)
see below my
List.
How can I resolve this problem?
I've searched on the web but, can't really find a solution.
Please help.
--- On Wed, 11/25/09, Landy Landy landysacco...@yahoo.com wrote:
From: Landy Landy landysacco...@yahoo.com
Subject: [asterisk-users] Unable to open sound file error
To: Asterisk
Michal
please wait I found some issues in my con file
2009/11/27 mickael ropars mrop...@gmail.com
I am running on Trixbox so my OS is Cent0S 5.4 and the Asterisk version is
1.4.22-4
on asterisk side Snmp module is running:
module load res_snmp.so
== Parsing
Hi,
I am new to the list, so I hope my questions aren't too stupid.
I am using Asterisk 1.4.21.2 and already set it up to use realtime, so a CDR
for an incoming SIP call is written in my mysql database. This works fine.
The problem is that I don't want to have my phone ringing all the time. I
Hi,
I would like to have my register directives from sip.conf in my mysql database:
register = user[:secret[:authuse...@host[:port][/extension]
I already have the sip users and the other config in the DB but how to get the
register in there, too?
In an old mail (Mon Oct 3 00:49:15 MST 2005)
I use CentOS, and it works fairly well. But I had to piece together info from
several places. I've tried it several different wants and this way worked, as
long as asterisk is run as root.
Copy asterisk-mib.txt and digium-mib.txt from asterisk_source/doc to
/usr/share/snmp/mibs/
Hello All,
Do you guys suggest any 1800 DID Provider in the US ?
I'm having a hard time to find one.
Thanks,
Marco
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To UNSUBSCRIBE or update
thanks all for your help, I really appreciate.
now it's working
My problem was due to
Nov 27 12:56:28 trixbox1 snmpd[5743]: /etc/snmp/snmpd.conf: line 61: Error:
example config COMMUNITY not properly configured
Nov 27 12:56:28 trixbox1 snmpd[5743]: /etc/snmp/snmpd.conf: line 62: Error:
example
On Fri, Nov 27, 2009 at 1:54 PM, Marco Cordeiro
marco.corde...@globalstar.com.br wrote:
Do you guys suggest any 1800 DID Provider in the US ?
We like OnSip.com / Junction Networks stable and various service
levels from none of hosted pbx. You should post this to the -biz list.
/r
I am trying to come up with a way to read a digit *before* the call is
answered. My Asterisk version is 1.6.2.0-rc6
SIP early media works fine (I can receive and transmit audio before the
call is answered), but as soon as I start the read application, Asterisk
answers the call which is not
Hello, I would appreciate if someone can give some help on what I want:
When someone call my box (from outside), to a certain ZAP port, it will put
him on hold, and immediately the box calls to outside SIP trunk to a
preconfigured certain number, then when the other party picks up the phone,
original message-
From: mickael ropars mrop...@gmail.com To: Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 27 Nov
2009 11:18:30 +0100
-
Hi Michal,
thanks
Can anybody recommend good quality replacement for Linksys SPA-3102 ATA?
I have to original Sipura 3K for over 4-years that are still working fine but
the Linksys 3102 I purchase are very poor quality (not to mention the echo on
PSTN line).
One unit quit working 2-weeks after arrival (needed to
Hello All;
Anyone can advise for the good phone (Polycom, Linksys, ... etc) that is a
stable and support the codecs: g723, g729, and speex?
Actually I would like to have the speex codec because it have the ability to
compress to very high compression so we can work with the low bandwidth (for
It is not that easy to give the answer. There are lots of itsp typical
ways of registration and you haven't provide the info needed to help
you out.
You need a register line in the general part of sip.conf. It should
look something like (mine looks like this
register =
Try IPComms.
j
On Fri, 27 Nov 2009, Marco Cordeiro wrote:
Hello All,
Do you guys suggest any 1800 DID Provider in the US ?
I'm having a hard time to find one.
Thanks,
Marco
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I finally saw why it was doing it: In Mobile - Settings - SIP From field
there is 4 options:
Tel/User (Standard)
User/User (Standard)
Tel/Tel/ (Not Reg)
User/Tel (Not Reg)
when I choose any of the first two, I dont have this problem but when I use
the last two I have this problem. At the same
Everuthing is working fine, but I have another question to SNMP users:
There is no hardware info in the MIB.
How can you do to send alarm (when one interface is down for exemple), is
there no way to check its status?
NB: I am using a Digium card
regards
Mickael
2009/11/27 mickael ropars
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
Erik.
I already solved this problem and posted it.
I was reloading all the setting but, it wasn't changing the provider's ip info.
After doing a restart now everything worked.
Thanks any ways for your help.
--- On Fri, 11/27/09, meetmecall i...@meetmecall.nl wrote:
From: meetmecall
We have swapped out the phone multiple times for the user.
Only one user.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cb
Sent: Wednesday, November 25, 2009 11:52 PM
To: Asterisk Users Mailing List -
It’s a single user and we have swapped everything.
The phone is an Aastra 6731i and its PoE.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Michael Wyres
*Sent:* Wednesday, November 25, 2009 6:27 PM
*To:* Asterisk Users Mailing
Hi Blaz -
Do you maybe know for a fairly good quality IAX2/SIP hard phones in up to 40
USD?
I don't think there are any IAX hardphone in production anymore. You
might be able to find a used Atcom 320, but probably not for anywhere
close to $40.
It looks like voipsupply.com has some old Cisco
Hi Mike -
I've got a Polycom 501 that's been working with Asterisk for some time.
However, I don't seem to be able to put a call on hold and get it back. It
goes on hold just fine. But when I press the resume button, nothing
happends.
Anyone seen this befor? Any ideas on where to start
We have swapped out the phone multiple times for the user.
Only one user.
Bad PoE port on the switch?
How about local interference that the user cannot control? Does the
same phone experience static when moved elsewhere?
Do you have a power brick for the phone so you can try it as non-PoE?
We swapped PoE switches, phones, cable and switch ports multiple times.
What do you mean by local interference? Cell phone? The person swears
nothing is near the phone.
Its very strange.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
hi there,
How can we track that the calls within queue has been hang up or disposed
within extension.conf ?
I am trying to run agi script once the call within queue has been finished.
Please advice.
amir
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Could the static be in the user's hearing aid?
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovey Forman
Sent:
We swapped PoE switches, phones, cable and switch ports multiple times.
What do you mean by local interference? Cell phone? The person swears
nothing is near the phone.
There are lots of things that can cause interference. Radios,
elevators, bad electrical wiring, you name it. Is the static
So, does anyone know of a way to detect whether a call from a SIP phone
is the first step of an attended transfer or an original call?
It could probably work if you put a SIP proxy in between (ref. Kamilio).
Another way might be to set up a special transfer extension that all
users use to
Your Digium card is for linux standard interface like eth0 (ethernet),
check IF-MIB.txt and OID from there.
BR,
Michał
2009/11/27 mickael ropars mrop...@gmail.com:
Everuthing is working fine, but I have another question to SNMP users:
There is no hardware info in the MIB.
How can you do to
Hello Micha ( all) ,
On Fri, 27 Nov 2009, michal kalinowski wrote:
Your Digium card is for linux standard interface like eth0 (ethernet),
check IF-MIB.txt and OID from there.
BR,
Micha?
When doing a snmpwalk of the IF-MIB having a (*) installed there is no
mention of an interface
Good evening all, hope everyone in the US had a nice Thanksgiving!
On one of our internal servers, I decided to make the leap from 1.4.2x
to 1.6.2.0-rc6 so I could start learning about the changes and new
features that have been implemented. I upgraded all the configs, removed
all the
Check this command snmpwalk -c your_community -v 1 localhost interfaces
in my system it's looks like that:
IF-MIB::ifNumber.0 = INTEGER: 4
IF-MIB::ifIndex.1 = INTEGER: 1
IF-MIB::ifIndex.2 = INTEGER: 2
IF-MIB::ifIndex.3 = INTEGER: 3
IF-MIB::ifIndex.4 = INTEGER: 4
IF-MIB::ifDescr.1 = STRING: lo
Michal,
in the IF-MIB you only have 4 interfaces lo,eth0,eth1,sit0
l0 which is the loopback interface
eth0, eth1 : ethernet interface
sit0 : use for PTP tunneling (use for IPv6)
so no information on the digium interface.
my IF MIB has also those interfaces
I found one the solution to get
Yes I know about that :) at this moment i have only machine with
lo,eth0,eth1,sit0.
On monday I will check that command on the server with e1 card.
BR,
Michał
W dniu 27 listopada 2009 23:51 użytkownik mickael ropars
mrop...@gmail.com napisał:
Michal,
in the IF-MIB you only have 4 interfaces
It will be the same, I already have 4 E1 interfaces. but no information in
the MIB
2009/11/28 michal kalinowski michal.kalinow...@interia.pl
Yes I know about that :) at this moment i have only machine with
lo,eth0,eth1,sit0.
On monday I will check that command on the server with e1 card.
What do You have in ifconfig ?
BR,
Michał
W dniu 28 listopada 2009 00:11 użytkownik mickael ropars
mrop...@gmail.com napisał:
It will be the same, I already have 4 E1 interfaces. but no information in
the MIB
2009/11/28 michal kalinowski michal.kalinow...@interia.pl
Yes I know about that
Thanks to a tip from someone who replied to me off list, I tried using
the 'den.teliax.net' proxy and that solved my issue. I'll have to follow
up with Teliax to see what the difference is.
Go figure. And thanks to Darrick for the info!
Jeff
On 11/27/2009 05:27 PM, Jeff Iddings wrote:
Good
Hello Mickael ,
On Fri, 27 Nov 2009, mickael ropars wrote:
Michal,
in the IF-MIB you only have 4 interfaces lo,eth0,eth1,sit0
l0 which is the loopback interface
eth0, eth1 : ethernet interface
sit0 : use for PTP tunneling (use for IPv6)
so no information on the digium interface.
In 2007, I released a Polycom Provisioning Tool. I retired the package
earlier this year, and have had so many requests for it, I have revived the
concept, new, improved, and still FREE.
It now lives here:
http://www.phoneprovisioning.com/
Provision any Polycom phone from the web, and
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