[asterisk-users] Asterisk SIP/IAX peers can't connect after Firewall change?

2010-06-17 Thread Chris Brentano
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

Re: [asterisk-users] Asterisk SIP/IAX peers can't connect after Firewall change?

2010-06-17 Thread Chris Brentano
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

Re: [asterisk-users] Asterisk SIP/IAX peers can't connect after Firewall change?

2010-06-17 Thread Chris Brentano
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

Re: [asterisk-users] (SOLVED) Kernel panic w/ DAHDI 2.x/Digium TE220B

2009-10-22 Thread 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

[asterisk-users] Kernel panic w/ DAHDI 2.x/Digium TE220B

2009-10-20 Thread Chris Brentano
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

[asterisk-users] Whither asterisk-addons?

2009-10-16 Thread Chris Brentano
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,

[asterisk-users] Nehalem Digium Wildcard issues?

2009-10-16 Thread Chris Brentano
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

Re: [asterisk-users] Whither asterisk-addons?

2009-10-16 Thread Chris Brentano
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

Re: [asterisk-users] Chanspy

2009-10-09 Thread Chris Brentano
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

Re: [asterisk-security] Person Trying to Register on my Asterisk multiple times

2009-01-23 Thread Chris Brentano
an ACL or iptables rule, but that can quickly becoming a losing strategy if the attacker has access to different systems or different networks. Good luck! - Chris --- Chris Brentano IT Engineer Jive Software 915 SW Stark St, Suite 400 Portland, Oregon 97205 Email/XMPP: chris.brent

Re: [asterisk-users] Digium training course

2008-09-21 Thread Chris Brentano
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

Re: [asterisk-users] Digium training course

2008-09-21 Thread Chris Brentano
, 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

Re: [asterisk-users] Asterisk and cloud computing (amazon EC2 + S3)

2008-09-11 Thread Chris Brentano
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

Re: [asterisk-users] Problems with D-channel (PRI)

2008-08-25 Thread Chris Brentano
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

Re: [CentOS] Adding new Hard disk to server with RAID-5

2008-08-22 Thread Chris Brentano
#1.) It should just appear as unpartitioned space to the OS. You can then partition it and add that partition to one of your LVs, and then use the LVM and ext filesystem tools to grow your existing LV and then resize the filesystem to fit. Good articles on LVM: -

Re: [asterisk-users] Asterisk broadcast to web

2008-08-11 Thread Chris Brentano
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

Re: [asterisk-users] Max amount of concurrent calls on and iax trunk

2008-08-06 Thread Chris Brentano
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

Re: LDAP auth instructions/how to?

2008-08-05 Thread Chris Brentano
in the security.properties section) And you might also need to add the LDAP specific configuration specified in the LDAP Settings section in this document: http://redback.codehaus.org/configuration.html HTH, Deng On Tue, Aug 5, 2008 at 8:16 AM, Chris Brentano [EMAIL PROTECTED] wrote: Hi all

Re: [CentOS] Whole disk encryption

2008-08-04 Thread Chris Brentano
I think TrueCrypt (www.truecrypt.org) will do this. On 4 Aug, 2008, at 8:51 AM, Plant, Dean wrote: Has there been any updates to support encrypting the whole disk in 5.2? If not, Is anyone doing this and can point me to some good documentation? Thanks Dean

Re: [CentOS] DVI + VGA?

2008-08-01 Thread Chris Brentano
If they're both the same output (i.e. not individual outputs, but just different media options) then I'd say yes. But if they are distinct outputs (say for a multi-monitor setup) then one may be display #1 and the other #2. If it's integrated video on your motherboard there may be options

Re: [CentOS] Using CentOS 5 as server; best way to setup NFSv4?

2008-08-01 Thread Chris Brentano
I would second OpenLDAP, having used it in production at two different employers. It's always been stable and reliable. If you're restarting slapd every 15 minutes I'd take a good hard look at the problem versus just migrating away from it. On that note, we recently migrated to Active

Re: [asterisk-users] func_curl.so Error on load

2008-04-21 Thread Chris Brentano
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

Re: [asterisk-users] func_curl.so Error on load

2008-04-20 Thread Chris Brentano
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

Re: [asterisk-users] Newbie Polycom: Subscription/Presence Problem

2008-04-20 Thread Chris Brentano
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

Re: [asterisk-users] func_curl.so Error on load

2008-04-20 Thread Chris Brentano
: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-users] func_curl.so Error on load

2008-04-19 Thread Chris Brentano
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

Re: [asterisk-users] func_curl.so Error on load

2008-04-19 Thread Chris Brentano
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

Re: [asterisk-users] bandwidth required for Asterisk running on T1

2008-04-11 Thread Chris Brentano
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

Re: [asterisk-users] SIP Conference phones

2007-12-24 Thread Chris Brentano
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

Re: [CentOS] system hibernating?

2007-12-24 Thread Chris Brentano
Hmm, I'm never encountered this myself. Could it be BIOS power management settings? - Chris On 24 Dec, 2007, at 8:58 AM, Jeffrey Ross wrote: I'm in the process of setting up a new system and I have found that the system is hibernating when its sitting idle for a long period of time.