similar to: ATA-186 de-register

Are there any other codecs that can be used with the 7960 and the ATA-186? I have been using the default gsm codec and wanted to see if I could make use of something a little less bandwidth intensive. Kim Callis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030723/34e950e0/attachment.htm

Hi! Sure, just look for: Wonder Shaper. It's a HTB based shaper configuration wich have some very good features, I use a variation of that here at my College. http://lartc.org/wondershaper/ It is the page (a simple google search). Also make sure to uncomment the line tos=lowdelay in every config file of asterisk that have it. Hope it is usefull, sincerely, Ildefonso Camargo

[ -netters, please Cc me or security@ with replies. ] I'm running into trouble integrating dynamic racoon-based IPSec into a network with ipfw and natd. I need to be able to allow VPN access from any address from authenticated clients. I've got the dynamic VPN working, with racoon negotiating SAs and installing SPs, but the problem is that I can't tell whether an incoming packet on

I have a handful of 7912's connected to my asterisk 1.2.14 server. (6 to be exact). I get the X on the display sometimes for loosing registration. I have the config file for the 7912's SipRegInterval: 60 and asterisk is the default. ; maxexpirey=3600 ;defaultexpirey=120 I've not changed them. How can I keep these phones online and stop loosing registration? Thanks, Jerry

Hello I am trying to redirect all http traffic of unauthorized wifi users on a wireless hotspot to a login page. The problem I have is that I can not disable the regular address translation (I want the source address to stay the same). 10.0.0.7 is the wifi client 195.250.155.29 is the web wifi user tries to access from his browser 195.113.17.94 is my login page 10.0.0.1 is the wifi

I noticed that there is a soft button for transfer when you initiate a call. I pressed it, and it actually put the call on hold, although I was able to call another extension. Is that soft button functional? And if so, how do you make use of it? And if not, how does one transfer a call? Kim C. Callis -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi. We just opened a gaming center and have chosen to run a FreeBsd box for our firewall. IPFW is configured at it's very basic running natd through rl0 and allowing any to any connections from the lan to the outer world. Natd controls access to the lan. We have a 6.0 mb/s ADSL net connection for all the gaming clients to use, however if a gamer starts downloading a file, that file

hi i have a problem with my icmp, i have a router that performs nat. i cannot ping to internet hosts from more than one stations situated behind NAT at once. if i want to ping from another station i have to stop the ping that was initiated from the first host, and after a few seconds i can ping from another station.i've checked firewll and i have no ipfw rules that could stop icmp traffic.

I've been using ipfw for a while to create a router with NAT and packet filtering, but have never combined it with stateful filtering, instead using things like "established" to accept incoming TCP packets which are part of a conversation initiated from the "inside". I'd like to move to using keep-state/check-state to get tighter filtering and also to allow outgoing

I have a FreeBSD box acting as a firewall and NAT gateway I would like to set it up to transparently pass IPSec packets -- I have an IPSec VPN client running on another machine, connecting to a remote network. Is there a way to do this? I can't find any hints in the man pages.

Alright, I am basically cheap, and I have a cellular plan which allows for free incoming calls (Nextel). I was wondering if there was any way to do sort of a dialback trick in the extensions.conf. I call into the system from my cell phone (maybe via DISA), I dial an internal extension, and dial a phone number. Then * sends to my cellphone the number dialed thus giving me a in call on the cell. Or

As I was working on my extensions.conf file, I started to segment calling privileges. For the everyday workers, I don't free reign to LD access unless it is business related. So I was wondering if there was a way to implement some type of accounting code to be entered before accessing LD, which of course would be noted in the CDR (however it is implemented, either comma delimited or MySQL).

I was trying to do a little searching to see if there has even been a comparison between Asterisk and VOCAL or any of the other OSS packages? "Practical Voice Over IP using VOCAL" published by O'Reilly and Associates, attempts to make a strong case about how scalable VOCAL. Of course, considering that the book is written by the makers of VOCAL, it tends to have a one sided slant.

Hi, I have a system with a 4.11 Kernel. Unless I'm doing something very wrong, there seems to be something odd with ipfw. Take the following rules: ipfw add 00280 allow tcp from any to any 22 out via bge0 setup keep- state ipfw add 00299 deny log all from any to any out via bge0 ipfw add 0430 allow log tcp from any to me 22 in via bge0 setup limit src-addr 2 ipfw add 00499 deny log

When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just keep getting this message every 30 seconds or so : May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its endpoint '*') does not exist Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets to

Yesterday, I started to experiment with Cisco to Cisco SIP calls using the g729 codec. According to the documentation, both the ATA-186 and 7960 are able to make use of the g729. >From an earlier e-mail, I made a change to the configuration of the ATA, changing the values: LBRCodec:3 RxCodec: 3 TxCodec: 3 The first thing I noticed was that when I did a sip show channels, the format had

Considering that I had a failure with compiling the latest version of asterisk because of cdr_mysql, I am going to assume that I need to have a copy of the mysql headers on the system in order to compile cdr_mysql.so. Does that sound correct? Kim C. Callis -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, after reading about it on the list I decided to set up a Free World Dialup account. For those of you who don't know, that is a sip proxy where you and your friends can singn up free and then you can just connect to it with any sip client and call anybody that is registered for free. Pretty much like iaxtel (I belive that was the name of it) for the iax protocol. It even supports clients

Hi, in the kernel I have these lines: [...] device miibus # MII bus support device rl device ed options IPFIREWALL #firewall options IPFIREWALL_VERBOSE #enable logging to syslogd(8) options IPFIREWALL_VERBOSE_LIMIT=0 #limit verbosity options IPDIVERT #divert sockets options DUMMYNET

Thanks, Mark! Here's a summary of what one needs to do in order to get NAT working with Asterisk. Please note that I have a Cisco ATA-186, and your experience may be slightly different based on the equipment you're using. You'll need to have a CVS updated version of Asterisk as 2003-03-06 ~2:00 PM EST. NOTE: This currently works for outbound calling only, not inbound. In other