search for: mdiehl

...if I should delegate those functions to the various ATA's. For example, the Sipura SPA 2002 can handle*69 internally. On the other hand, doing it in the Asterisk dialplan gives me centralized control. I'm torn. Anyone want to weigh in on this? -- Mike gpg key: http://diehlnet.com/~mdiehl/mdiehl.asc 83AD D927 758D 4BFC A800 0277 4B26 75A4 F0D1 C7EB

...en telecommute and sometimes I do some side business; these practices violate many provider's acceptable use policies. So, I need a provider who doesn't care how I use the phone, and one that works well with Asterisk. Any comments welcome. Thanx, -- Mike gpg key: http://diehlnet.com/~mdiehl/mdiehl.asc 83AD D927 758D 4BFC A800 0277 4B26 75A4 F0D1 C7EB

Hi All, Since 21 june skype is available to be used on Linux, with a static binary, which includes QT, of 8 meg its big. http://www.skype.com/help_linux_faq.html I presume, with some hacking, there could be a possibility to use the Skype program as a Channel. (Eq. Skype is started, and with a visual scripting thing a connection is made and Asterisk connects via OSS (or the alsa emulation

Hi all, Lately, I've seen an increase in the number of attacks against my system from the so-called "Friendly Scanner." When one of these script kiddies targets my server, all I see for symptoms is a few of my trunks become lagged due to server load and a stream of messages on the console that resemble this: [Aug 2 20:27:50] == Using SIP VIDEO CoS mark 6 [Aug 2 20:27:50] ==

Hi all, While trying to track down a T.38 issue, I came across a series of log entries like this: ============================================================================ [Jan 26 10:23:31] WARNING[32508]: udptl.c:948 ast_udptl_new_with_bindaddr: Unable to allocate socket: Too many open files [Jan 26 10:23:31] ERROR[32508]: acl.c:488 ast_ouraddrfor: Cannot create socket

I am wondering if anyone has sample XML config for the Sipura 2100 ATA. We have been autoprovisioning our 2002s with success and the 2100's take the same XML that we have come up with, but I am not sure of the syntax for specific things that I need these boxes to do, such as turning T.38 on. If anyone is willing to share their xml for autoprovisioning Sipura 2100's, it would be much

Hi all, I seem to be having a strange problem with a sip trunk. On a fairly frequent basis, I'll make a call, ore receive a call, and there will be NO sound. The strange part is that both endpoints are public IP addresses so NAT isn't in play and a sniffer trace reveals that the packets simply aren't being sent. It only seems to happen on a particular trunk. The same phone

Hi all, I've got a problem where many times, there is silence at the first 1-2 seconds of a call. Then it clears up and it's crystal clear. I've not put a sniffer on it, yet, but I suspect that the media channel is still being set up. The server shouldn't be too overloaded. Can anyone give me some advise on how to solve/mitigate this problem? Mike.

Hi all, I'm trying to tighten things up a bit and I seem be be running into something that doesn't make sense to me. I've got 2 contexts, one for customers, and one for guests, that I include into [customers] and [default], in extensions.conf, as below: ============================================================= [default] include = dial_GUEST [customers] include = parkedcalls

Hi all, Lately, I've been seeing more and more instances where I get a flood of warning messages like this: [Apr 26 14:09:50] WARNING[21054] udptl.c: No UDPTL ports remaining The next thing I know, my server is dropping calls and starting to misbehave. I use fax via T.38, so I can't just turn udptl off. I could expand the port range, but I suspect that will just mask the situation.

I recently upgraded Asterisk from 1.8.x to 13.x and am now finding that music on hold isn't working like it used to. It seems that even though the correct MoH class is being set, the system still plays the "default" music. All of my call handling is done with an AGI script. When a call is made, the AGI script sets the MoH class before dialing. The log indicates that the correct

...;$@"; do if [ -d ${BASEDIR}${ext} ];then for msgdir in $(ls -d ${BASEDIR}${ext}/*); do ProcessDir ${msgdir} done else echo "${BASEDIR}${ext} is not a valid directory" fi echo "Processed extension $ext" done On Mon, Oct 9, 2023 at 3:06 PM Mike Diehl <mdiehl at diehlnet.com> wrote: > Hi all, > > I need to be able to delete a voicemail message using a program. > > Is is sufficient to simply delete the .wav and .txt files in the spool > directory? > Or do I need to also renumber the remaining files? > > For example, let say...

Hi all, I need to be able to delete a voicemail message using a program. Is is sufficient to simply delete the .wav and .txt files in the spool directory? Or do I need to also renumber the remaining files? For example, let say a given mailbox has 20 messages in it and I want to delete message number 5. Can I just delete the 2 files and expect that asterisk will renumber them? Or do I

...ewall). Also, Digium regularly warns users that fail2ban is NOT a > security system: http://forums.asterisk.org/viewtopic.php?p=159984 > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of mdiehl > Sent: Tuesday, August 15, 2017 3:38 PM > To: asterisk-users at lists.digium.com > Subject: [asterisk-users] Detecting DoS attacks via SIP > > Hi all, > > Lately, I've seen an increase in the number of attacks against my system > from the so-called "Friendly Scanne...

I'm still playing with * at the moment. Just trying to figure out how much it can do. Anyway, it looks like * supports voice modems. So here come the obvious questions... How well does * support vmodems? How do I set * to use my vmodem for incoming calls? Has anyone gotten it to work? Thanx, Mike Diehl.

Hi all, I'm looking into being able to send/receive SMS messages with my asterisk box in the US. I've seen the SMS command as well as the Kannel program. I'd prefer to do it from Asterisk. I've tried something like: exten => 999,n,sms(15551234567,s,"This is a test") in my dialplan, but when this runs, it dials the phone number and then nothing. What am I

Hi all, I'm trying to get moh working on * version 1.4.4. I've setup a test extension that answers the call and runs the musiconhold command with the appropriate class name. All I get on the phone is silence. The console tells me that moh started and immediately stopped, but it complains that there is "No class: moh0" *CLI> [Feb 22 12:17:36] WARNING[31142]:

Hi all, I've cross-posted this to the -users and -biz groups. Hope that's OK. I have a customer who REALLY needs to be able to send/receive faxes reliably. I could probably get hylafax configured, but I'm not sure how reliable it is. If it is considered reliable, would someone let me know? Otherwise, is there a product/service they can buy that will allow them to fax to/from

Hi all. I could have sworn this working at one time... But it doesn't look like any of the functions provided by features.so is working for me. (one-touch monitoring, attended/blind transfer, etc) I've (re)loaded features.so, as well as bridge_builtin_features.so. The config file looks sane. What else should I try? TIA, -- Take care and have fun, Mike Diehl.

Hi all, I have 2 servers running 1.6.2.9 and I'm about to build a third server. This suggests the possibility of doing a rolling upgrade of all of my servers. This brings up the question of what version to install and upgrade to. I don't have many upgrade opportunities, so I'd like to get as much bang for my buck. Since I've applied some custom patches to my 1.6, I'd