Re: [asterisk-users] Asterisk 11 - Security event logging over syslog
Am 24.11.2012 um 01:33 schrieb Matthew Jordan: On 11/22/2012 04:00 PM, Michael Keuter wrote: Hi all, I am just testing with Asterisk 11.01. The SIP security event logging works fine for me for console and file logging, but the security events are not logged over syslog. logger.conf: ... syslog.local0 = notice,warning,error,security Is this on purpose, a fault on my side, or is this a bug? No, that should work. What's the output of 'logger show channels'? -- Matthew Jordan logger show channels Channel Type StatusConfiguration --- --- syslog.local0 Syslog Enabled- NOTICE WARNING ERROR SECURITY /var/log/asterisk/security_log File Enabled- SECURITY Console Enabled- NOTICE WARNING ERROR SECURITY Everything else except security works fine over syslog. Michael http://www.mksolutions.info smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Debugging Information..
I did a little googling, but didn't seem to find anything specific to answer the question. I am trying to debug some calls on an Asterisk system (AsteriskNow) that are dropping, and when the general logs didn't nail anything I turned on SIP Debugging on the trunk to the provider. Basically the complaint is that when some call in, regardless of if the call is answered, or if Vmail answers it, it drops the calls in a matter of seconds. The strange thing is, that the system processes many hundreds of calls daily, but only a couple specific incoming callers are seeing the drops. I would have thought a NAT issue, but why does this only affect a specific group of incoming callers, the rest go about their business just fine. I think thinking bandwidth.com is mucking something up, but again I have no specific proof one way or another, so why the debugging. When one of the problem callers is dropped, in the SIP debugging I see: chan_sip.c: Scheduling destruction of SIP dialog '285991942_79966325@192.168.27.72' in 6400 ms (Method: BYE) Is this the remote end (ie bandwidth.com) dropping the call, or is the local Asterisk server dropping the call? For any that care to look at all the gory details, here is a chunk of the debugging output: [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: --- SIP read from UDP:216.82.224.202:5060 --- ACK sip:4104159233@10.98.4.36:5060 SIP/2.0 Record-Route: sip:216.82.224.202;lr;ftag=gK0e4bc97f Record-Route: sip:67.231.8.93;lr=on;ftag=gK0e4bc97f Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bKebf3.453cc5a5.2 Via: SIP/2.0/UDP 67.231.8.93;branch=z9hG4bKebf3.315d4e14.3 Via: SIP/2.0/UDP 192.168.37.72:5060;branch=z9hG4bK0eB7f5f0b80116aa493 From: sip:2159470824@192.168.37.72;isup-oli=0;tag=gK0e4bc97f To: sip:+14104159233@67.231.8.93;tag=as6974aee7 Call-ID: 353260172_48597606@192.168.37.72 CSeq: 11346 ACK Max-Forwards: 68 Content-Length: 0 - [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: --- (12 headers 0 lines) --- [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: --- SIP read from UDP:216.82.224.202:5060 --- INVITE sip:4104159233@10.98.4.36:5060 SIP/2.0 Record-Route: sip:216.82.224.202;lr;ftag=gK0e4bc97f Record-Route: sip:67.231.8.93;lr=on;ftag=gK0e4bc97f Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bKfbf3.9d9b1065.0 Via: SIP/2.0/UDP 67.231.8.93;branch=z9hG4bKfbf3.c159a6a.0 Via: SIP/2.0/UDP 192.168.37.72:5060;branch=z9hG4bK0eB7f601a4d116aa493 From: sip:2159470824@192.168.37.72;isup-oli=0;tag=gK0e4bc97f To: sip:+14104159233@67.231.8.93;tag=as6974aee7 Call-ID: 353260172_48597606@192.168.37.72 CSeq: 11347 INVITE Max-Forwards: 68 Contact: sip:+12159470824@192.168.37.72:5060 Content-Length: 235 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 22153 5058 IN IP4 192.168.37.72 s=SIP Media Capabilities c=IN IP4 67.231.8.102 t=0 0 m=audio 6576 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 - [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: --- (15 headers 11 lines) --- [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Sending to 216.82.224.202:5060 (NAT) [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Found RTP audio format 0 [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Found RTP audio format 101 [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Found audio description format PCMU for ID 0 [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Found audio description format telephone-event for ID 101 [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Peer audio RTP is at port 67.231.8.102:6576 [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: --- Transmitting (NAT) to 216.82.224.202:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bKfbf3.9d9b1065.0;received=216.82.224.202;rport=5 060 Via: SIP/2.0/UDP 67.231.8.93;branch=z9hG4bKfbf3.c159a6a.0 Via: SIP/2.0/UDP 192.168.37.72:5060;branch=z9hG4bK0eB7f601a4d116aa493 Record-Route: sip:216.82.224.202;lr;ftag=gK0e4bc97f Record-Route: sip:67.231.8.93;lr=on;ftag=gK0e4bc97f From: sip:2159470824@192.168.37.72;isup-oli=0;tag=gK0e4bc97f To: sip:+14104159233@67.231.8.93;tag=as6974aee7 Call-ID: 353260172_48597606@192.168.37.72 CSeq: 11347 INVITE Server: FPBX-2.9.0(1.8.15.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: sip:4104159233@10.98.4.36:5060 Content-Length: 0 [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Audio is at 11444 [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: ---
[asterisk-users] Wierd RTP issue
I have a peculiar RTP issue. I'm experimenting with Jitsi as a softphone on one of my desktop Windows machines. That machine can either be connected to Asterisk via an VPN connection (with a static IP address) or not (via NAT). When it's connected via NAT, all is OK. When it's connected with VPN, the following occurs: The voice path inbound to Jitsi works fine when Jitsi originates the call, no matter what it's calling. The voice path inbound to Jitsi works fine when it's called from another SIP device. The voice path inbound to Jitsi is silent when it's called from something on the other side of a PRI via DAHDI. I've run Wireshark on my desktop and see the RTP packets coming at the same rate and protocol (g711) in all the cases and sip set debug peer xyz doesn't shed any light on the situation (the SDP data looks similar in the working and non-worknig cases). Does anybody have any ideas what to look at next? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] * Waiting for asterisk to shutdown .............
- Original Message - From: Joseph syscon...@gmail.com To: asterisk-users@lists.digium.com Sent: Saturday, November 24, 2012 12:54:12 AM Subject: [asterisk-users] * Waiting for asterisk to shutdown . I'm running asterisk on a small box, Intel-R-_Atom-TM-_CPU_330_@_1.60GHz and when I try to restart the asterisk it fails. /etc/init.d/asterisk restart * Caching service dependencies ... [ ok ] * Killing wrapper script ... [ ok ] * Stopping asterisk PBX gracefully ... * Waiting for asterisk to shutdown . * Failed. When I run /etc/init.d/asterisk status I get: * status: started At this point I have to kill the process ID zap it (/etc/init.d/asterisk zap) and restart it. Why asteriks can not shut down properly? How can I monitor this process and restart it? What version of Asterisk are you running? Michael (elguero) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP password probe
I looking through my logs, I found that people where probing my SIP accounts looking for passwords. Asterisk was helping them out by processing hundreds of requests per minute. I did a bit of Googling and this seems to be a frequent knock against Asterisk's security. It would seem pretty simple to add a configuration setting to sip.conf to delay the response to a bad account or password. There is a half measure to confuse the probe by sending the same error return for either error. It appears that many people have complained that this should be the default setting only changed if your are debugging a problem. There is no reason for a working system to ever have bad passwords so this is clearly an attack in almost every case. A simple delay would solve the problem for most people who use reasonable passwords. I had to install fail2ban which is a PITA but thanks to someone's clear recipe, I was able to get it working. I hope that this can be worked into a release soon. Ron -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Debugging Information..
- Original Message - From: Howard Leadmon how...@leadmon.net To: asterisk-users@lists.digium.com Sent: Saturday, November 24, 2012 3:19:10 PM Subject: [asterisk-users] SIP Debugging Information.. I did a little googling, but didn't seem to find anything specific to answer the question. I am trying to debug some calls on an Asterisk system (AsteriskNow) that are dropping, and when the general logs didn't nail anything I turned on SIP Debugging on the trunk to the provider. Basically the complaint is that when some call in, regardless of if the call is answered, or if Vmail answers it, it drops the calls in a matter of seconds. The strange thing is, that the system processes many hundreds of calls daily, but only a couple specific incoming callers are seeing the drops. I would have thought a NAT issue, but why does this only affect a specific group of incoming callers, the rest go about their business just fine. I think thinking bandwidth.com is mucking something up, but again I have no specific proof one way or another, so why the debugging. When one of the problem callers is dropped, in the SIP debugging I see: chan_sip.c: Scheduling destruction of SIP dialog '285991942_79966325@192.168.27.72' in 6400 ms (Method: BYE) Is this the remote end (ie bandwidth.com) dropping the call, or is the local Asterisk server dropping the call? [snip] --- [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: --- SIP read from UDP:216.82.224.202:5060 --- BYE sip:4104159270@10.98.4.36:5060 SIP/2.0 Record-Route: sip:216.82.224.202;lr;ftag=gK0b66d829 Record-Route: sip:67.231.4.93;lr=on;ftag=gK0b66d829 Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bKe902.53bde7e.0 Via: SIP/2.0/UDP 67.231.4.93;branch=z9hG4bKe902.32697e93.0 Via: SIP/2.0/UDP 192.168.27.72:5060;branch=z9hG4bK0bBac8c2c3cb90659df From: sip:7173381800@192.168.27.72;isup-oli=0;tag=gK0b66d829 To: sip:+14104159270@67.231.4.93;tag=as0850c6db Call-ID: 285991942_79966325@192.168.27.72 CSeq: 297 BYE [snip] If I am reading this right, it looks like a BYE is coming in from the far end, Bandwidth.com. Michael (elguero) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] updates to packages.asterisk.org?
Yep, there has been no updates for at least the last 2.5 months. I would also like to find out what the plans are. -Vladimir On 11/23/2012 5:43 PM, Eric Germann wrote: Will there be an update to the RPM repo on packages.asterisk.org? For example http://packages.asterisk.org/centos/5/asterisk-1.8/x86_64/RPMS/ Latest is showing 1.8.15.1. Thanks EKG -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] * Waiting for asterisk to shutdown .............
Joseph писал 24.11.2012 07:54: I'm running asterisk on a small box, Intel-R-_Atom-TM-_CPU_330_@_1.60GHz and when I try to restart the asterisk it fails. /etc/init.d/asterisk restart * Caching service dependencies ... [ ok ] * Killing wrapper script ... [ ok ] * Stopping asterisk PBX gracefully ... * Waiting for asterisk to shutdown . * Failed. When I run /etc/init.d/asterisk status I get: * status: started At this point I have to kill the process ID zap it (/etc/init.d/asterisk zap) and restart it. Why asteriks can not shut down properly? How can I monitor this process and restart it? I don't know what's in your startup script, for I never used one, but it says about graceful shutdown. It might be executing core stop gracefully or maybe core stop when convenient In the first case, Asterisk will stop receiving new commands and calls, will wait for all current calls to disconnect and then shutdown. In the second case, Asterisk will continue receiving new calls (not sure about commands), and as soon as there are 0 active calls, it will shutdown. As you can see, in the second scenario you'll rarely get to the shutdown on active server. Not sure why do you kill the zap process. Does it hang up? For when you kill it and drop all Zap calls, asterisk may already rest in piece, if my above guess is right. -- With Best Regards Mikhail Lischuk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] * Waiting for asterisk to shutdown .............
On 11/24/12 14:43, Michael L. Young wrote: - Original Message - From: Joseph syscon...@gmail.com To: asterisk-users@lists.digium.com Sent: Saturday, November 24, 2012 12:54:12 AM Subject: [asterisk-users] * Waiting for asterisk to shutdown . I'm running asterisk on a small box, Intel-R-_Atom-TM-_CPU_330_@_1.60GHz and when I try to restart the asterisk it fails. /etc/init.d/asterisk restart * Caching service dependencies ... [ ok ] * Killing wrapper script ... [ ok ] * Stopping asterisk PBX gracefully ... * Waiting for asterisk to shutdown . * Failed. When I run /etc/init.d/asterisk status I get: * status: started At this point I have to kill the process ID zap it (/etc/init.d/asterisk zap) and restart it. Why asteriks can not shut down properly? How can I monitor this process and restart it? What version of Asterisk are you running? Michael (elguero) I'm using asterisk asterisk-1.8.15.1 on Gentoo -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] * Waiting for asterisk to shutdown .............
On 11/25/12 01:30, Mikhail Lischuk wrote:
Joseph пиÑал 24.11.2012 07:54:
I'm running asterisk on a small box, [1]Intel-R-_Atom-TM-_CPU_330_@_1.60GHz
and when I try to restart the asterisk it fails.
/etc/init.d/asterisk restart
* Caching service dependencies ... [ ok ]
* Killing wrapper script ... [ ok ]
* Stopping asterisk PBX gracefully ...
* Waiting for asterisk to shutdown .
* Failed.
When I run /etc/init.d/asterisk status
I get: * status: started
At this point I have to kill the process ID
zap it (/etc/init.d/asterisk zap)
and restart it.
Why asteriks can not shut down properly?
How can I monitor this process and restart it?
I don't know what's in your startup script, for I never used one, but
it says about graceful shutdown.
It might be executing core stop gracefully or maybe core stop when
convenient
In the first case, Asterisk will stop receiving new commands and calls,
will wait for all current calls to disconnect and then shutdown.
In the second case, Asterisk will continue receiving new calls (not
sure about commands), and as soon as there are 0 active calls, it will
shutdown.
As you can see, in the second scenario you'll rarely get to the
shutdown on active server.
Not sure why do you kill the zap process. Does it hang up? For when you
kill it and drop all Zap calls, asterisk may already rest in piece, if
my above guess is right.
--
With Best Regards
[2]Mikhail Lischuk
It does the same thing on my both servers, the one I've mentioned above and my
AMD 8-core server.
The environment is not busy but I've notice it that it does it when I don't
restart it for a few days or a week.
When I start from fresh I can restart it and it shuts down and restart
correctly.
If don't zap it I can not start or restart the asterisk as it keeps telling
me that the process has been started; so I have to zap it first.
Here is Gentoo start-up strips
cat /etc/init.d/asterisk
#!/sbin/runscript
# Copyright 1999-2012 Gentoo Foundation
# Distributed under the terms of the GNU General Public License v2
# $Header:
/var/cvsroot/gentoo-x86/net-misc/asterisk/files/1.8.0/asterisk.initd3,v 1.1
2012/09/03 08:37:12 chainsaw Exp $
extra_started_commands=forcestop reload
depend() {
need net
use nscd dns dahdi mysql postgresql slapd capi
}
is_running() {
if [ -z `pidof asterisk` ]; then
return 1
else
PID=`cat /var/run/asterisk/asterisk.pid`
for x in `pidof asterisk`; do
if [ ${x} = ${PID} ]; then
return 0
fi
done
fi
return 1
}
asterisk_run_loop() {
local result=0 signal=0
echo Initializing asterisk wrapper
OPTS=$*
trap rm /var/run/asterisk/wrapper_loop.pid EXIT
cut -f4 -d' ' /proc/self/stat /var/run/asterisk/wrapper_loop.pid
while :; do
if [ -n ${TTY} ]; then
/usr/bin/stty -F ${TTY} sane
${NICE} /usr/sbin/asterisk ${OPTS} ${TTY} 21 ${TTY}
result=$?
else
${NICE} /usr/sbin/asterisk ${OPTS} 21 /dev/null
result=$?
fi
if [ $result -eq 0 ]; then
echo Asterisk terminated normally
break
else
if [ $result -gt 128 ]; then
signal=`expr $result - 128`
MSG=Asterisk terminated with Signal: $signal
CORE_TARGET=core-`date +%Y%m%d-%H%M%S`
local CORE_DUMPED=0
if [ -f ${ASTERISK_CORE_DIR}/core ]; then
mv ${ASTERISK_CORE_DIR}/core \
${ASTERISK_CORE_DIR}/${CORE_TARGET}
CORE_DUMPED=1
elif [ -f ${ASTERISK_CORE_DIR}/core.${PID} ];
then
mv ${ASTERISK_CORE_DIR}/core.${PID} \
${ASTERISK_CORE_DIR}/${CORE_TARGET}
CORE_DUMPED=1
fi
[ $CORE_DUMPED -eq 1 ] \
MSG=${MSG}\n\rCore dumped:
${ASTERISK_CORE_DIR}/${CORE_TARGET}
else
MSG=Asterisk terminated with return code:
$result
fi
# kill left-over tasks
for X in ${ASTERISK_CLEANUP_ON_CRASH}; do
kill -9 `pidof ${X}`;
Re: [asterisk-users] SIP and RTP on different IP's
On 25/11/2012, at 1:23 PM, Tiago Geada tiago.ge...@gmail.com wrote: linux does sort this out and asterisk listens in both interfaces. however asterisk connects and tells remote end to send rtp back at the same IP where sip is going trough... remote end does try to send it but gets stopped in a firewall.. thus if asterisk did present a different IP to recieve RTP in its SIP header, this would not happen! I think this is outside of asterisk's natural ability You may need a proxy server in between you and the Cisco to achieve this if you can't change the firewall. http://forums.asterisk.org/viewtopic.php?f=1t=84018 Have you tried making the preferred route to these addresses go out eth1, thus getting the required address? Ultimately seems odd the firewall allows access in but not out, guessing you have no control over that? Good luck Cheers Duncan On 23 November 2012 19:39, Duncan Turnbull dun...@e-simple.co.nz wrote: On 24/11/2012, at 2:19 AM, Tiago Geada tiago.ge...@gmail.com wrote: Hello Folks, I am looking for a way that makes asterisk tell remote SIP party that the IP where they will send RTP is not the same as the one I am comunicating via SIP Can this be done anyhow? I can try and explain: We have placed a asterisk box in our partners office. It has eth0 with IP 172.16.1.10 and eth1: 10.34.18.250 linux has its routes set so it can comunicate with several networks in their offices. now there is a cisco call manager that we need to communicate with. Normally via our IP 172.16.1.10, however seems that this cisco uses some sort of 'directmedya=yes' and sets both ends speaking RTP with themselves. There are some extensions in cisco that have a network 10.134.0.0/16 that we can only comunicate via eth1 thus when calling cisco (always via eth0) sometimes we need to say that OUR IP to recieve RTP is not 172.16.1.10, but 10.34.18.250 This is a routing issue, not asterisk I think. You are saying you route to cisco via eth0, it sets up connections to its end points and then drops out of the media flow, but the end points have no route to the eth0 address so they fail Linux usually sorts this out and asterisk replies on the address of the interface it sends out with. So for the most part the response in my experience if its going out eth1 should use the eth1 ip address. If you can get to it via eth0 and thats the preferred route then it will have the eth0 address. If so why can't you change your routing table to use eth1 when you need to go to the cisco then you will have the right address and the far extensions can respond to you correctly Or change the cisco network endpoints so they can successfully access your address on eth0 can this be done? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
