Thanks Dale!
I will try your method in the one web directory.
Also while I was waiting for a response, I decided to start from scratch in a
different web directory and rewrite my code for a database based system.
I am writing so that the system inserts into table the extension and time of
I have a multi tenant asterisk box where on tenant is receiving calls from the
caller ID as1as and they cannot pickup the call.
The caller ID also does not show up in the call log.
Thoughts?
Thanks,
--Eric
--
_
--
, 2014, at 12:02 PM, Tiago Geada tiago.ge...@gmail.com wrote:
logs ?
full log containing the call?
On 8 January 2014 14:56, Eherr email.eherr9...@gmail.com wrote:
I have a multi tenant asterisk box where on tenant is receiving calls from
the caller ID as1as and they cannot pickup
Not sure if this has been answered but I cannot find a solution.
I am running Asterisk 1.4.26.3
I am seeing the following lines in my log files:
A: [2013-03-05 13:54:27] NOTICE[6928] chan_sip.c: Failed to authenticate user
sip:192.210.138.12;tag=DmVIjOlfYiiL
B: [2013-03-05
I am trying to configure the following scenario but have failed.
Currently, I have an Asterisk box sitting on a Static Public IP address in my
office.
I have a remote office with 3 Polycom IP335s that are registering back to my
local office's publically address Asterisk box.
The
phones but I can't really
tell you all that you need to know.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Wednesday, October 03, 2012 9:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject
seconds ( roughly
5 rings ) and then pull the call back and send it to
voicemail if the line was not answered.
Has anyone seen this feature created and implemented and if so, how?
Thanks,
--eherr
--
_
-- Bandwidth
What is the lowest end machine to run a production asterisk server.
Currently I have an MTE version running on a $2500 server.
I want to get a single Tenant asterisk server for one company but don't want to
waste a $2500 server for one tenant.
Do you guys have any recommendations for
a production asterisk
server
On Fri, May 11, 2012 at 6:15 AM, eherr email.eherr9...@gmail.com wrote:
What is the lowest end machine to run a production asterisk server.
Depends on a lot of variables. I've got some old 1.8GHz 1U servers running
hundreds of calls.
How many calls, how much
Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom
phones.
I want to be able to emulate a key system but I cannot figure it out.
Everything I tried so far is just not working together.
Thanks,
--E
--
Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom
phones.
I want to be able to emulate a key system but I cannot figure it out.
Everything I tried so far is just not working together.
Thanks,
--E
--
Of eherr
Sent: Thursday, May 10, 2012 9:45 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SLA - Shared Line Appearance - Polycom
Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom
phones.
I want to be able to emulate a key system but I cannot
this? -
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/SLA.html
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursday, May 10, 2012 12:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion
and life should be good. You set up the hints
as contacts in the directory.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursday, May 10, 2012 1:31 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject
,hint,SIP/400
Then set up your lines to look for 1000@default, 2000@default, etc.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursday, May 10, 2012 1:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion
- the sla.conf makes the line active the hint makes it accessible by
the phone.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursday, May 10, 2012 1:51 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject
I have fail2ban running on my Asterisk box.
Every so often I receive emails stating that the jails stopped and then started.
Why does this happen?
Why isn't it just continuously running?
Thanks,
-E
--
_
--
:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Sip Registration Hijacking
On 20/01/12 01:36, eherr wrote:
It is also register on an AudioCodes MP-118.
Thanks,
-E
Is the Audiocodes gateway accessible online? Have you set a strong
password for it's web interface
Can you please elaborate on rate limiting. Not how to implement it but rather
how implementation is beneficiary.
Reading up on it, it appears that it just checks the tcp connections and denys
connection if limit is passed.
In my thoughts, this is essentially a live fail2ban monitor in
This is actually an interesting concept however I do think I want to restrict
dialing during a specific time period.
If someone is in the office, I would have to reprogram the route so allow
dialing which adds overhead.
Again, I do like the concept though.
Thanks,
--E
From:
] On Behalf Of Larry Moore
Sent: Saturday, January 21, 2012 1:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sip Registration Hijacking
On 20/01/2012 9:36 AM, eherr wrote:
I have a honey pot box with extensions that are not just numbers ie )
100
I have an asterisk box which has Polycom Soundpoints IP335 and IP650s
registering to it both locally and remote.
I want to be able to incorporate a cordless phone at the remote location; not a
wireless phone.
I want it to also be able to register to the same asterisk box so it can take
...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Monday, January 23, 2012 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cordless SIP phone
On Mon, Jan 23, 2012 at 7:35 AM, eherr email.eherr9...@gmail.com wrote:
I have an asterisk box which has Polycom
] On Behalf Of A J Stiles
Sent: Monday, January 23, 2012 10:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cordless SIP phone
On Monday 23 January 2012, eherr wrote:
I have an asterisk box which has Polycom Soundpoints IP335 and IP650s
registering
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sip Registration Hijacking
On Thu, Jan 19, 2012 at 8:36 PM, eherr email.eherr9...@gmail.com wrote:
I have a honey pot box with extensions that are not just numbers ie )
100-MySipUserName
I have the same problem and I
I have a honey pot box with extensions that are not just numbers ie )
100-MySipUserName
And the passwords are from an openssl generated password ie)
Gq5VNIjDFWIQoUT6
However, this one extension keeps getting hacked and showing up on a different
IP address.
It is also register
What are you using for hardware?
I have experienced SPA2102s supplying a DTMF tone when someone was talking.
This was caused by the talker reaching a certain frequency while talking in
which the SPA popped out a DTMF tone.
I haven't experienced this behavior on polycoms or anything else.
--E
When the side car looses it entries, what does the config file show for the
entries.
This happened to me one time but that was only because for some reason, the
contacts file was deleted by accident and I had to
recreate it. ( I have a backup now too! )
It probably as Dan said, check
Out of curiosity, how many concurrent phone calls for an office that uses
Polycoms could be sustained on a DSL ( 3meg down, 768 up )
line using g711?
Not sure if its 64kbps or 87kbps.
I would say roughly 8 but I don't know if the polycoms add any more payload to
the network for presence
I believe it is set by a character length for formatting the output.
What are you trying to accomplish? Are you just viewing it in the CLI or are
you writing monitoring scripts?
Can you switch names so that they are unique in the beginning?
--E
-Original Message-
From:
I have a Polycom Soundpoint IP335.
There are no inbound routes set to the phones yet.
However, the phones are getting phantom rings.
What is the legitimacy of these calls?
Is there something I need to block to stop it?
I believe its people trying to hack the phones/phone
to your local DHCP server and your Asterisk box.
DHCP Server 192.X.X.X
Asterisk Server 192.X.Y.Y
Phone 192.X.Z.Z
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Friday, November 18, 2011 8:34 AM
To: 'Asterisk Users
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Friday, November 18, 2011 8:50 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom Phantom Ringing
Well this is a remote site.
I am running 1.4.26
I have multiple
I am running into an issue installing asterisk 10.0.0-rc1
I have centos 2.6.18-194.el5 #1 SMP Fri Apr 2 14:58:35 EDT 2010 i686 i686 i386
GNU/Linux installed.
I am at the point of trying to install the dahdi and I am getting the error
message when I do a make all:
You do not appear to
But what is the correct physical setup of a CLEC.
Do you get rack space at a carrier hotel and equipment in there?
Do you get rack space at the local ILEC CO?; which is Verizon here.
What are the types of voice platforms used by CLECs?
Thanks,
--E
-Original Message-
From:
I would agree, unfortunately.
However, I still see it as a glorified webcam chat and not a telecommunication
device like a SIP/soft phone.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson
On the polycom soundpoint ip 650 six line phone:
Say I have 4 lines on hold, is there way to tell who I put on hold.
I cannot see the caller ID of the other lines, only the last line I placed on
hold.
Thanks,
--E
--
When you perform an attended transfer, the extension of the person transferring
is displayed to the co-worker.
Can I override the caller ID to display the caller's callerID during a blind
transfer?
Thanks,
--E
--
_
--
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Wednesday, November 16, 2011 12:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] polycom soundpint ip650 question
On the polycom soundpoint ip 650 six line phone:
Say I
So there is no way to do these with programming.
For instance, setting a variable in the DB and grabbing it to override the
field when transferring?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard
Unfortunately, I only have 1.4.26 installed.
What's the next stable version?
Should I go to 1.6, 1.8, or 10
Thanks,
--E
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Wednesday,
On Wed, Nov 16, 2011 at 1:49 PM, eherr email.eherr9...@gmail.com wrote:
When you perform an attended transfer, the extension of the person
transferring is displayed to the co-worker.
Can I override the caller ID to display the caller's callerID during a blind
transfer?
Thanks,
--E
I know this questions is not really asterisk related, however I know a lot of
people here are in the industry.
I was curious if anyone here could provide insight on how to become a
facilities based CLEC.
I did a lot of google-ing and read info on voip-info.org but it's all the same
I think the question is more along the lines of how does asterisk know
immediately when a sip phone becomes on line and when you
unplug the phone from the network, how does asterisk essentially know
immediately that it status is UNKNOWN
If I am not mistaken.
--E
From:
-registration part is
different. If the phone is gracefully taken off line it
specifically de-registers. If it just can't be reached because it powers off
or the router closes NAT, or whatever, then Asterisk
won't know this until it times out.
On Mon, Nov 14, 2011 at 3:19 PM, eherr email.eherr9
How do people get around when their DID provider is experiencing routing issues
and cannot terminate the DID to your box and all
your clients from that provider can no longer receive inbound calls to their
company?
Thanks,
--E
--
Of eherr
Sent: Friday, November 11, 2011 12:43 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] DID Provider Issues
How do people get around when their DID provider is experiencing routing issues
and cannot terminate the DID to your box and all
your clients
I am having similar issues with Asterisk 1.4.26
It happens at random times; could be once a day or a few hours in between up to
a month or so.
Haven't found a solution to my problem yet either.
--E
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Unfortunately, I didn't compile with DON'T_OPTIMIZE. Would this render my
backtrace.txt completely useless or should I still submit?
Thanks,
--E
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
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