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 y
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 in
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 e
What do You have in "ifconfig" ?
BR,
Michał
W dniu 28 listopada 2009 00:11 użytkownik mickael ropars
napisał:
> It will be the same, I already have 4 E1 interfaces. but no information in
> the MIB
>
> 2009/11/28 michal kalinowski
>>
>> Yes I know about that :) at this moment i have only machine
It will be the same, I already have 4 E1 interfaces. but no information in
the MIB
2009/11/28 michal kalinowski
> 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
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
napisał:
> Michal,
>
> in the IF-MIB you only have 4 interfaces lo,eth0,eth1,sit
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 stat
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
I
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 deprecate
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 as
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 :
> 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 (w
>> > 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 u
> 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 stati
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: Frid
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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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
[mailto:asterisk-users-boun...@lists.
> 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?
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 st
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 Cisc
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 L
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 - Non-Comm
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 wrote:
> From: meetmecall
> Subject: Re: [asterisk
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
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
> H
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 t
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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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 => ::@ipness.net:6060
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
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
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,
>
>
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,
bo
>> 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 whi
On Fri, Nov 27, 2009 at 1:54 PM, Marco Cordeiro
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
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 co
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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asterisk-users mailing list
To UNSUBSCRIBE or update opti
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 /doc to
/usr/share/snmp/mibs/
mkdir /var/agentx
t
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) Oll
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 j
Michal
please wait I found some issues in my con file
2009/11/27 mickael ropars
> 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
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 wrote:
> From: Landy Landy
> Subject: [asterisk-users] Unable to open sound file error
> To: "Asterisk Users Mailing List - Non-Commercial Discussi
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 s
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 l
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
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
"mi
Quoth Jon Morgan
>
>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
>
>span=2,0,0,ccs,hdb3,
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
___
-- Ban
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 context/extension/p
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