[asterisk-users] Timeout for conferences
Hi, The dialin conference via asterisk is over, one person is still in the conference room and accidentally does not hang up properly. Her meter at the phone company keeps running... I'd like to implement something to the effect of checking whether there is only one participant in the conference, and when this is the case, to cancel the call after a predefined time (perhaps 5 or 10 mins. to allow for some waiting for latecomers). Has someone already written some code or a quick idea for this scenario? Regards, --AvH ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoIP-GSM gateway problem
I bought a MV-372 for 2 SIM cards as the one channel model seems to work well (see http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk). The setup is such: - Inet -- VoIP provider --- POTS | | (iax2, NAT) | asterisk (on abox with iptables fw) | (SIP, LAN) |-- SNOM190 phones | -- SIP-GSM-module --- SIM cards -- mobile phone networks Sound, however, is too bad for the SIP to GSM module to be usable. Call initiation from the LAN to the GSM network works but the audio stream stops and continues for about 1 to 3 seconds each, irregularly alternating between various durations of both states. Latency is around 300ms for the module which registers as a SIP extension. The machine is a PII with a 400MHz Celeron. Transmission is via the alaw codec, as recommended. The RTP Packet Length setting for the GSM module is 20ms. Do any of you have suspicions why the module does not work as expected? (The vendor has not yet answered yet but it's weekend in Taiwan as well). Perhaps the GSM-module firmware is not up to par, and/or the SNOM doesn't cooperate well in the bridged connection. --AvH --- from sip.conf: [general] port=5060 externip=23.45.67.89 bindaddr=123.456.789.220 localnet=123.456.789.0 defaultexpirey=120 maxexpirey=3600 context=internal disallow=all allow=alaw language=de canreinvite=no [GSM] type=friend host=dynamic defaultip=123.456.789.222 secret=xxx qualify=yes username=xx fromuser=xx context=gateway call-limit=2 dtmfmode=inband allow=alaw insecure=very [SNOM190] [3] type=friend host=dynamic defaultip=123.456.789.221 secret=xx qualify=yes I've tried nat=yes and no canreinvite=yes as well qualify on and off in both clients ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mysterious SIP packets to Cogent
In my SOHO setting even when nobody is using the phone my firewall drops outgoing packets from the asterisk box to a Cogent server, din't find naything through Google about it: (out: eth0 xxx.xxx.xxx.xxx.:2129 - 66.250.40.33:24441 UDP len:193 ttl:64). Anyone know what this traffic is supposed to be good for? Greetings --AvH ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk from Debian Packages
Tzafrir Cohen wrote: On Mon, Dec 11, 2006 at 12:11:34AM +0100, Andreas von Heydwolff wrote: I'm using 1.2.13~dfsg-2 from Debian unstable in a small SOHO environment, it's doing its job. However, the startup scripts seem to hose something and it's running but not working with /etc/init.d/asterisk start, but running it from commandline solved the problem. Asterisk has been up for a couple weeks again. Hadn't the time to look into that yet, perhaps a problem with old config files from previous versions. Please report bugs (reportbug asterisk) . Others may have the same problem as you. Have you modified /etc/init.d/asterisk ? What do you have in /etc/default/asterisk? Hi again. Sorry, was just too busy in th meantime. It's all working just as it should, must have been a temporary glitch. 1.2.13~dfsg-2 is doing fine on a sarge/etch mix with debian kernel 2.6.18-8. Had to install the self compiled zaptel modules with # dpkg -i --force-overwrite though as some config file is shared with the kernel's. --AvH ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] snom 190 (etc.?) dialscript for * debugging and kaddressbook
Thought I might just as well share these scripts, they may work with other phones too: *1)* Dialing from the KDE 3.5.5 address book works with a script that gets triggered from the kaddressbook (Settings - Script Hooks - Phone) with my command snom_dial_number %N The script snom_dial_number itself goes like this: - #!/bin/sh ENCODEDNUMBER=$(echo $@ | sed 's/\+/00/g' | sed 's/\///g' | sed \ 's/-//g' | sed 's/\#/\%23/g' |sed 's/ //g' | sed 's/0043/0/g') konqueror -geometry 700x30+350-810 \ http://172.16.0.2/command.htm?DIAL=$ENCODEDNUMBERDIAL #EOF Substitute 172.16.0.2 with your phone's IP number and 0043 with your country code. The format of numbers in my address book is +CC AC NUMBER which works also for exporting via gnokii to my Nokia mobile. The script handles the empty spaces and eventual hyphens. (BTW, for SMS sending via bluetooth I added to the script hooks cat %F | gnokii --sendsms %N) *2)* When working on the dialplan on the office asterisk server via ssh from home I needed to test outgoing calls - but nobody was physically there. What to do? Being logged in on a shell on my remote asterisk machine I used the following script to trigger outgoing calls from an office snom 190 phone to my phone beside me on the desk. A timeout of 3 secs for POTS or 15 secs for my mobile guaranteed that no voicebox would take over but I heard a short ring when calls got through, to add a real life ringtone to remote visual feedback from asterisk -rv. httpsnom-dialtest - #!/bin/bash # Created 070107 by AvH # $1 is the extension to dial if [ $1 = ] then echo enter number please ; exit fi # command for snom 190 phone, taken from # http://80.237.155.31/kb/index.php?View=entryCategoryID=21EntryID=40 SOURCE=command.htm?number=$1 # origin EXT=2 # IP number of phone echo Dialing from $EXT # the actual command, -w is a timeout echo -e GET $SOURCE HTTP/1.0\n\n | nc -w 1 $EXT 80 /dev/null #EOF I guess the second script can be put into use for KDE as well. Any ideas for improvements? Cheers, --AvH ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
