We're learning all the time and made some significant progress and some
very nice calls scenarios, but specifically with this issue, is there
anything we can do to solve the interop problem with this end-point?
Thanks.
Original Message
Subject: Re: [asterisk-users] FFA
Thanks Kevin,
We managed to get the ReceiveFAX going, while making some minor changes
to the code, like, for example, using the ${UNIQUEID} for the file name.
Regards,
Michael
Original Message
Subject: Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not
Hi
I'm sorry for the my trivial quest.
I Have asterisk 1.4 with TDM 400 with FXO and FXS, and works fine from
several months.
Now I want to connect a device to TDMFXS that want a ring frequecy of
25 hz to activate: i am italian, and usually the ring freq is 20 hz.
The other time (I have used that
On my own version of sox (14.3.0), says -w option is not allowed
ABEJIDE, Ayodele A. (CCNA)
+2348039269311
Date: Mon, 1 Nov 2010 13:39:43 -0700
From: asterisk@sedwards.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Music On Hold Help
Un-top-posting...
On
Say,
If bandwidth e.g. ADSL goes fuzzy, is there a way to force * to unregister the
Peers?
I noticed with qualify=200 for example, even if latency goes above and * shows
Lagged and then UNREACHABLE
The peer's calls are still accepted.
Is there a way to automatically prevent this?
Thanks
Thanks Zeeshan, I have some problems with 1.4.25 so I'll try 1.4.27.
Have a nice day!
On Fri, Oct 29, 2010 at 1:33 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Yes, it works fine in 1.4.22 and 1.4.27 and 1.4.35.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On
Hi guys,
i've seen this too, nagios woke me up because it was an extremely high
volume of tries.
I took a peek into the logs and saw that the attacker's script was
trying extensions from 1 to and then random names. I can see the
log in the messages file that several attempts failed
Joel, after sending my previous posts I did realize your points might
have some validity - and hence I owe you an apology - and that is if
you are a telco or hosted pbx provider then strict fail2ban is not
that good of a solution. While I was talking strictly from a PBX
vendors point of view,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Tuesday, November 02, 2010 10:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FW: Under heavy attack
I'm still on old-fashion copper-wire and have yet to experience the joy of
SIP Trunk-ing and the type of issues discussed in this thread. My thought
to share here is that outgoing calls should be easy for thoroughly
authenticated users and impossible for others...
Probably more
Hi,
Firstly, I'm new to Asterisk and am a system admin rather than a phone
engineer. I've googled and read around but haven't been able to answer my
questions sufficiently to buy hardware and get this thing set up.
Secondly, if I've missed vital information from what is below, please let me
- Original Message -
When I call into my Asterisk box via my VoIP line (using gsm codec)
and then try to make an outgoing DISA call over PSTN I get the
following:
[Nov 1 15:12:54] WARNING[17694]: chan_dahdi.c:8930 dahdi_write: Cannot
handle frames in gsm format
[Nov 1 15:12:54]
Hi,
In Europe many Telcos implement power-save mode
(See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to
'Activation / Deactivation' for more information).
Would you agree to have this feature added to the ones already discuused for
next Asterisk release ?
(See
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jon pounder
Sent: Tuesday, November 02, 2010 10:24 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] FW: Under heavy attack
I'm still
On Tue, Nov 02, 2010 at 04:13:01PM +, Ronny Adsetts wrote:
3. Other ways?
It all rather depends on what your proprietary system has been set up to do.
(If you didn't already have the Samsung box, you wouldn't need to buy one.)
Dedicated telephony hardware tends to be restricted in all
Hi All,
During last three month I have worked on improving functionality of Nortel
phones working with asterisk to replace existing Nortel station by asterisk.
Many improvments done, listed below. I have only i2002 phone and unable to
test if new version of channel correctly works with i2204
Roger Burton West said at 02/11/2010 16:35:
On Tue, Nov 02, 2010 at 04:13:01PM +, Ronny Adsetts wrote:
3. Other ways?
It all rather depends on what your proprietary system has been set up
to do. (If you didn't already have the Samsung box, you wouldn't need
to buy one.) Dedicated
Use Network LCR and PRI Card...
I interfaced two Samsung iDCS 500 r2 into an asterisk box this way. I don't
think Samsung has gotten SIP 100% open yet.
I have since scrapped all the Samsung gear, and have a big pile of parts :D
If you're talking 10 or 20 users on the 100, just scrap it, go
Gordon Henderson said at 02/11/2010 16:39:
On Tue, 2 Nov 2010, Ronny Adsetts wrote:
Firstly, I'm new to Asterisk and am a system admin rather than a
phone engineer. I've googled and read around but haven't been able
to answer my questions sufficiently to buy hardware and get this
thing set
On Tue, Nov 02, 2010 at 05:54:27PM +, Ronny Adsetts wrote:
Would it be possible do you know to use the Samsung handsets with an Asterisk
system? Is it worth even trying to save money here? (I've no idea of the cost
of VoIP handsets for use with Asterisk).
I've never heard of Samsung
Steve;
You are so right - it was the end of the day, I was tired and pissy.
Let me try this again:
Version:
ns211156*CLI core show version
Asterisk 1.4.31 built by root @ some_server.foo.net on a x86_64 running
Linux on 2010-06-10 14:32:34 UTC
Name and version of endpoints involved:
Sip
Asterisk security has always been a big concern. I am sure most of asterisk
pros have taken care of these type of attacks. For non pros I am sharing a
shell script here.
http://www.didforsale.com/blog/?p=253
If you care feel free is use it.
-Jai
On Tue, Nov 2, 2010 at 9:27 AM, Cary Fitch
Hey;
I never thought of that.
It is causing an issue for me. One SIP UA works fine - ring, forward,
etc. While the other does not.
I am a little clueless here - where would I start with this?
Thanks
Glen
On 11/1/2010 19:15, Philipp von Klitzing wrote:
Hi!
[Nov 1 19:55:49]
Some of you have already noticed we've chosen a number of Atlassian
tools to provide services to the Asterisk and Asterisk SCF communities
(Confluence, Crowd, Crucible, Fisheye and Bamboo). Of course, we're not
alone in this since many other open source projects have chosen these
tools as well,
Hello Folks;
Again, excuse my cluelessness.
I have an Asterisk server in the US - and I want to connect it to one in
Europe.
Here is my scenario:
1. call a phone number, my Asterisk box in the US answers
2. perhaps a 'please wait' voice message
3. it dials an extension on the other
On Tue, Nov 02, 2010 at 03:20:48PM -0400, Silver Thorne wrote:
I am not looking for someone to do this for me, I am just not really
sure how to get started. Perhaps some suggested reading, examples,
etc?
The simplest approach would be to skip the answering and just dial
through immediately,
Silver Thorne wrote:
Hello Folks;
Again, excuse my cluelessness.
I have an Asterisk server in the US - and I want to connect it to one
in Europe.
Here is my scenario:
1. call a phone number, my Asterisk box in the US answers
2. perhaps a 'please wait' voice message
3. it dials
On Tue, Nov 2, 2010 at 3:20 PM, Silver Thorne zora...@gmail.com wrote:
Any help at all would be appreciated.
DUNDi
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) |
Blog: http://blog.polybeacon.com | Twitter:
On Tue, Nov 2, 2010 at 11:16 AM, Danny Nicholas da...@debsinc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Tuesday, November 02, 2010 10:06 AM
To: Asterisk Users Mailing List -
Hi,
I've got a client with two ADSL connections for redundancy.
Is it possible to set up asterisk to connect to one SIP provider using both
adsl connections and load balance between the two connections?
Or to use one connection as the main one, and automatically fail over if the
first
Your router would have to do per-destination when it came to load balancing
between the two dsl circuits. That way a single call could only use one dsl
path.
On Nov 2, 2010 7:36 PM, Dan Journo d...@keshercommunications.com wrote:
Hi,
I've got a client with two ADSL connections for
I have sent an e-mail to this list (awaiting moderator approval by the
size) talking about some difficult to make calls with a SIP Provider in
Brazil.
I'm new at this list and have no sure if I have posted my question in
the right place.
If this is not the channel to make this kind of question
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