similar to: Too many open files

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.

Hi all. I'm in the process of configuring a phone system for my family and friends. I'm wondering if I should try to implement the "Vertical Services" (http://www.nanpa.com/number_resource_info/vsc_assign) in the Asterisk dialplan, or if I should delegate those functions to the various ATA's. For example, the Sipura SPA 2002 can handle*69 internally. On the other

Hi all. I'm looking to change from a standard telephone line to a VoIP phone line at home. I'm looking for recommendations for VoIP providers that I can use with Asterisk. One of the catches is that I often 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

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, 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, 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

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

I have a high volume asterisk 1.40 installation and I ran a SIP SHOW CHANNELS. (see partial output below). My questions are: 1. "wc-l" of the output shows 4000 lines. Does this mean 2000 active calls? (2 channels per call) 2. The latter part of the output shows "unkn" for Form column. Why does it not know the codec? Could it be UDPTL? Or are these calls messed up? 3.

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

Hi all, Is it possible to send a SIP header to a PAP2T or SPAxxxx and cause the device to automatically answer? I can do this with my Polycom phones and would like to do it with my ATA's. Any ideas? -- Take care and have fun, Mike Diehl.

Here is something I wrote years ago. I expect you can adjust it for your needs # cat remove_blank_vmail #!/bin/bash # remove_blank_vmail takes arguments as voicemail boxes and removes messages with audio files shorter then MINSIZE (in bytes) #---------------------------------------------------------------------- # Description: # Author: John Harragin Monroe-Woodbury CSD # Created at: Thu Nov 6

Hi all, Does anyone have any good/bad comments on the A1200P 12-port fxo/fxs card from OpenVox? I'll be using one to with 8-12 fxo interfaces.? The cards will be plugging into a cable-modem / phone adapter.? We weren't able to port the numbers, so we're going to use the existing PSTN connection and replace all of the office phones. With these short distances, will I need to worry

Hello, I m using asterisk version 1.6.2.0 beta. I m trying to test load on it, for which i m using WINSIP installed at two computers and facing two problems. Problem 1: I got 100 users registered to asterisk from each winsip and then initiates 100 calls from one winsip other winsip. But the problem is approx of 60 calls get mature and asterisk give error for the remaining like shown below.

Hello. I've got a Cisco SPA303 that I'm trying to provision via http. I noticed that this device looks very similar to a PAP2T, so I used that as a template for my provisioning file. However, the result is less than stellar. Line 1 registers and works. However, lines 2 and 3 also register as line 1, effectively giving me a 1-line phone with 3 buttons. Also, the line name is the