Hi all,
I tried searching, so if this has already been discussed please point me in the
right direction.
On separate occasions I've seen cases where Asterisk boxes will be unable to
register with each other via SIP or IAX2 when a Firewall is replaced. I'll
describe two different cases. In
SIP this morning they successfully
re-registered.
Is there some sort of TTL, cache, saved salt value, or other time/session
related tidbit saved that is expiring here?
- Chris
On Jun 17, 2010, at 10:21 AM, Chris Brentano wrote:
Hi all,
I tried searching, so if this has already been
I have a suspicion that it's the saved/cached SIP/IAX2 useragent info:
-- Registered SIP 'paloalto' at 10.XX.X.25 port 5060
Saved useragent Asterisk PBX 1.6.1.6 for peer paloalto
Is there a way to clear this saved info manually?
- Chris
On Jun 17, 2010, at 10:29 AM, Chris Brentano
FYI, in case anyone else encouters this issue. The card that I had
which I could reproduce this with was hardware revision B4. I RMAed
the card with Digium support and got a newer, revision C card, and the
issue is no more.
On 20 Oct, 2009, at 3:25 PM, Chris Brentano wrote:
I've seen
I've seen this consistently on three systems, with three different
cards, and multiple versions of DAHDI. At first I thought the issue
only occurred on newer, Nehalem-based, systems, but I reproduced it on
a Core 2 Duo box as well. I've tested with dahdi-linux 2.2.0.2, dadhi-
linux-complete
I noticed that asterisk.org got a redesign, quite recently it seems,
which is very nice, but the addons package isn't listed for download
any longer, nor are releases posted to
http://downloads.asterisk.org/pub/telephony/
.
That said, looks like it's still available in svn,
Just putting this out there to see if anyone else has seen any issues.
May cross-post to asterisk-dev if it's indeed a bug (and not my own
stupidity).
I've got a Digium TE220 (2xT1 interface w/Echo canceller) that in two
separate Nehalem-based (Xeon E5520 Gainestown) boxes (HP ProLiant
Correction, I did notice it for download at
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-addons-1.6.1.1.tar.gz
- Chris
On 16 Oct, 2009, at 4:06 PM, Chris Brentano wrote:
I noticed that asterisk.org got a redesign, quite recently it seems,
which is very nice, but the addons
Use ExtenSpy for spying on a specific extension.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExtenSpy
On 9 Oct, 2009, at 10:44 AM, Torintino T wrote:
How can i activate ChanSpy to spy on a dedicated extension?
I see the following in /etc/asterisk/extensions_additional.conf
I would also disagree that the written exam is biased towards people
who attended the training. I attended a Bootcamp earlier this year and
thought I was fully prepared to pass the dCAP. Especially since I
already had real-world Asterisk experience. But the written exam
covered material
, at 10:20 AM, Chris Brentano wrote:
I would also disagree that the written exam is biased towards people
who attended the training. I attended a Bootcamp earlier this year and
thought I was fully prepared to pass the dCAP. Especially since I
already had real-world Asterisk experience
I don't really see the advantage to be honest. If I needed Asterisk access from anywhere I'd just run it locally on my laptop, connect to an ITSP via IAX or SIP, and run a softphone app locally. The only exception I can think of is when you'd want people to be able to leave voicemail on your
I hope you have a spare card for things like this! :)
If you do, I'd suggest building another Asterisk box (given you also
have a spare PC for such), copy over the configs, and install the
spare card. After hours, or whenever it may be appropriate, schedule
some downtime and test the new
Off the top of my head... you could probably route the audio of a
softphone (like Zoiper/X-Lite) to something like Nicecast (Mac) or
Icecast.
On 11 Aug, 2008, at 9:21 AM, Andrew Niemantsverdriet wrote:
Hi all,
I have an interesting problem that I am looking for a solution for. I
want to
I have two Asterisk 1.4 boxes connected via IAX over a VPN tunnel on a
10Mbit link. We never did any stress testing as it's a temporary
arrangement, but we've never had any call quality issues or run up
against concurrent call limitations. I'm mostly routing internal
extensions over the
Generally I'd agree. But it could at least more adequately notify the
user, even if they are compiling on a different system than where it
will be running on. It just seems that in most cases people will be
compiling on the system they will be installing on. This is what they
teach at the
PKG_CONFIG_PATH=/usr/lib/pkgconfig and reran
./configure and it was happy again.
- Chris
Tzafrir Cohen wrote:
On Sat, Apr 19, 2008 at 11:11:53AM -0700, Chris Brentano wrote:
Nevermind, I found the problem.
And for the benefit of the readers of the archives: what
I believe this isn't a Polycom thing, but the nature of SIP devices in
general. But, that said, Polycom should start making IAX desk phones. :-)
- Chris
Lee, John (Sydney) wrote:
DND does not do anything for me BLF-wise either (shame). Simply
picking up
the handset won't do, at
:53AM -0700, Chris Brentano wrote:
Nevermind, I found the problem.
And for the benefit of the readers of the archives: what was it?
Fair money on the prospect that he failed to put /usr/local/lib in
/etc/ld.so.conf and run ldconfig.
I'll take your bet
Asterisk 1.4.19, Zaptel 1.4.10 and libpri 1.4.3. When I try to start up
Asterisk (with -cvvv) I get an error regarding func_curl.so
(lines omitted)
...
== Registered custom function STRFTIME
== Registered custom function STRPTIME
== Registered custom function EVAL
== Registered custom
Nevermind, I found the problem.
Chris Brentano wrote:
Asterisk 1.4.19, Zaptel 1.4.10 and libpri 1.4.3. When I try to start
up Asterisk (with -cvvv) I get an error regarding func_curl.so
(lines omitted)
...
== Registered custom function STRFTIME
== Registered custom function STRPTIME
Additionally Mark, a Channelized (also called Integrated) T1 offers 24
channels for voice/data, but after bit robbing (for signalling, etc) you
only get around 56kbps per channel. ISDN PRI over T1 has 23 b-channels
of voice/data and one d-channel for signalling, etc. PRI is preferred
and most
I'll third the Polycom units. We use SoundStation IP 4000s in our
conference rooms and they work great. But yes, they are expensive.
- Chris
On 24 Dec, 2007, at 7:20 AM, Michael Graves wrote:
On Mon, 24 Dec 2007 14:29:46 -, Chris Bagnall wrote:
Greetings list,
Does anyone have
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