Re: [asterisk-users] Asterisk died without any message, segfault
2010/10/27 Benoit maver...@maverick.eu.org: On 27/10/2010 12:59, Krzysztof Urbaniak wrote: Hi! We've experienced asterisk has gone without any message, it wasn't any segfault, anything in asterisk messages log that says about shutting down. How do you launch asterisk ? did you try without 'safe_asterisk' or anything like it, It was launched by safe_asterisk, asterisk was launched with following parameters -f -vvvg -c just 'asterisk -cvvv' within a 'screen' for example ? Has anybody got similar problem? Have you searched the bugs repository ? Yes, and i don't have found anything about problems like this. Asterisk is version 1.4.29-1 from digium repository. there is a few new releases for 1.4.x, it is mostly bug fixes. I would suggest you try the latest one and if it still dies build it with debugging options I know, but this is a critical machine and we can update it in near future, cause we don't have any service window next days. -- _ -- 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 client floods port 5060 and gets blocked
Hello, Is there any reason why an IP-phone would pounder on port 5060 ? My firewall blocks the public IP because it thinks the remote IP is port scanning on port 5060. I think the phone is just registering but for some reason it does this repeatedly in a very short time. Oct 28 09:01:48 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48073 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:01:49 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48074 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:01:50 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48075 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:01:52 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48076 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:01:56 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48077 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:02:00 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48078 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:02:04 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48079 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:02:08 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48083 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:02:12 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48084 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:02:16 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48085 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:02:20 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48087 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Any input on this ?! Kind regards, Jonas. -- _ -- 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] being bombarded with SIP packets
Over the last two weeks, we have had at least two incidents where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further). The immediate issues were dealt with by having the firewall drop those packets, but I was wondering: 1) if anyone has seen the same problem, and 2) if you've got some iptables rules for limiting inbound SIP by rate? (or some such). thanks Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- 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] 2 HB8 cards in one server - first one is not recognized, the second is
Le 26/10/2010 14:49, Shaun Ruffell a écrit : On 10/26/2010 06:38 AM, Administrator TOOTAI wrote: I installed 2 HB8 cards each of them with a Quad Bri modules in a HP 360 G6 running Debian Squeeze. Here is an output of dmesg wafter server has booted: [...] before asking RMA for the card, I would like to know what you think about this matter. First, Digium technical support would be more than happy I'm sure to help you trouble shoot this. That being said... First thing I would do is update to the current trunk of dahdi-linux. Revision 9397 [1] http://svn.asterisk.org/view/dahdi?view=revisionrevision=9397 was added because of some systems that did not provide reliable polling from the board side, which could result in erroneous your firmware may be corrupted... messages. However, since you have one card that works and one that doesn't I give this a low probability of fixing it. Didn't test this yet but Next, if updating the driver does not help and if the problem follows the card (i.e., you can swap cards and now the second card fails to load), switching cards gives kernel panic :-( on boot I would disable dahdi from starting automatically, power off your system, remove the working card, power on, and try modprobe wctdm24xxp forceload=1 on the chance that the firmware on the board actually is corrupted. Will try card by card, then slot per slot Thanks for your help -- Daniel -- _ -- 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 client floods port 5060 and gets blocked
I assume that you checked and the remote IP is a legitimate IP phone? If not, it could be an attempt to break into your system. If it is a legitimate IP phone, make sure that the SIP configuration is correct - if the SIP authentication fails, you can see this happening. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, October 28, 2010 12:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP client floods port 5060 and gets blocked Hello, Is there any reason why an IP-phone would pounder on port 5060 ? My firewall blocks the public IP because it thinks the remote IP is port scanning on port 5060. I think the phone is just registering but for some reason it does this repeatedly in a very short time. Oct 28 09:01:48 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48073 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:01:49 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48074 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:01:50 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48075 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:01:52 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48076 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:01:56 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48077 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:02:00 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48078 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:02:04 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48079 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:02:08 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48083 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:02:12 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48084 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:02:16 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48085 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:02:20 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48087 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Any input on this ?! Kind regards, Jonas. -- _ -- 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] Asterisk 1.8 IAX Registration
Paul, Thanks, I'll try this patch later tonight. Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: 28 October 2010 03:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 IAX Registration On Tue, Oct 26, 2010 at 8:26 PM, Paul Belanger paul.belan...@polybeacon.com wrote: I'm going to try and look at this during Astricon :) Ok, just uploaded a new patch on https://issues.asterisk.org/view.php?id=18202 Let me know if it worked. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- 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] Astribank Configuration Issues
On Wed, Oct 27, 2010 at 05:37:15PM -0500, Don Kelly wrote: I have recently updated from Centos/*1.2 to Ubuntu Server and FreePBX 2.8.0.2. We have an Astribank with 4 T1 ports and 16 FXS ports. After updating, we had it working for a while with one NT PRI and one TE PRI and, in the process of trying to configure another PRI, I ran into a couple problems. (1) As my configuration changes didn't seem to affect the Astribank, I power-cycled it. I found that it doesn't reload firmware automatically when it's connected. I can force it to load, but am missing something to reload automatically. What's the output of: dahdi_hardware -v lsdahdi Do you have fxload installed? (2) I would appreciate a step-by-step suggestion of how I can make configuration changes that propagate properly to the Astribank. (3) I'd like to know if it's possible to determine what configuration has been loaded into the Astribank without visiting the site and looking at the lights. There's practically no configuration in the Astribank itself. There is firmware that gets loaded to it at startup, as it is not saved on the Astribank itself. But the configuration is on your system. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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 client floods port 5060 and gets blocked
On 10/28/2010 10:44 AM, Kevin Keane wrote: I assume that you checked and the remote IP is a legitimate IP phone? If not, it could be an attempt to break into your system. If it is a legitimate IP phone, make sure that the SIP configuration is correct -- if the SIP authentication fails, you can see this happening. 1. This is a legitimate phone, yes. 2. Registration goes as follow : REGISTER SIP/2.0 401 Unauthorized Re-Register with Digest 200 OK Regards, Jonas. -- _ -- 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] being bombarded with SIP packets
On Thu, 2010-10-28 at 09:41 +0200, Per Jessen wrote: Over the last two weeks, we have had at least two incidents where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further). The immediate issues were dealt with by having the firewall drop those packets, but I was wondering: 1) if anyone has seen the same problem, and 2) if you've got some iptables rules for limiting inbound SIP by rate? (or some such). thanks Per Jessen, Zürich Was it legitimate requests or a brute force attack? If it was a brute force attack have you considered using fail2ban? Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Asterisk died without any message, segfault
Le 28/10/2010 08:41, Krzysztof Urbaniak a écrit : 2010/10/27 Benoitmaver...@maverick.eu.org: On 27/10/2010 12:59, Krzysztof Urbaniak wrote: Hi! We've experienced asterisk has gone without any message, it wasn't any segfault, anything in asterisk messages log that says about shutting down. How do you launch asterisk ? did you try without 'safe_asterisk' or anything like it, It was launched by safe_asterisk, asterisk was launched with following parameters -f -vvvg -c yes well, i have experienced some very weird comportement with safe_asterisk, and colors ... just 'asterisk -cvvv' within a 'screen' for example ? Another question is what physical interface/card/driver are you using ? maybe there is a bug in your current driver I know, but this is a critical machine and we can update it in near future, cause we don't have any service window next days. Is it really better to let this kind of things happens ? can't you just build install the new release and restart on the next unused period / night / .. (after testing on another system) ? -- _ -- 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] being bombarded with SIP packets
Am 28.10.2010 09:41, schrieb Per Jessen: Over the last two weeks, we have had at least two incidents where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further). The immediate issues were dealt with by having the firewall drop those packets, but I was wondering: 1) if anyone has seen the same problem, and 2) if you've got some iptables rules for limiting inbound SIP by rate? (or some such). thanks Per Jessen, Zürich Hello Per, (iptables) rule #1: search the archives You will find nearly as many postings about that problem, as your server SIP packets received ... ;-) Norbert -- _ -- 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] Mobile Phones and Asterisk
Hi,
On 10/28/2010 01:06 AM, GBR Icasiano, Ryan A. wrote:
Hi,
Thanks for your reply. I'm calling a normal phone using the DIAL cmd. Here is
my sample dial command:
exten =s,4,Dial(SIP/xxx${extensi...@media_gateway,10,t)
but when I use:
exten =s,5,NoOp(SIP/xxx${extensi...@media_gateway has state ${DIALSTATUS})
I'm not quite sure what you are trying to do.
So you called the phone for 10 seconds, the phone didn't answer - and
the variable DIALSTATUS told you exactly that.
Is the problem the fact that the line is not ringing out? Is that what
is wrong?
And why do you have some xxx in front of ${extension}? You shouldn't
need them. Just pass ${extension} - which is the number you dialled on
the phone.
Sebastian
I hear a busy tone, after the 10 sec. timeout it returns NOANSWER, as defined
in my DIAL func.
I also tried getting the DEVICE_STATE
exten =s,3,NoOp(SIP/xxx${extensi...@media_gateway has state
${DEVICE_STATE(SIP/xxx${extensi...@media_gateway)})
and same thing happens as stated on the scenario below.
Thanks again!
regards,
RYAN ICASIANO
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
[s...@open-t.co.uk]
Sent: Wednesday, October 27, 2010 5:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Mobile Phones and Asterisk
Hi,
On 10/27/2010 05:55 AM, GBR Icasiano, Ryan A. wrote:
anyone???
regards,
RYAN ICASIANO
Hi,
I changed my sip.conf and added call-limit. At first I thought it works ok,
since i tried calling a cellphone that is currently busy(phone answers 1st
softphone, then another softphone calls the same number, it now returns
INUSE). But then, i tried calling a different number while the first phone
is busy, but it returns INUSE. It seems that the status being returned was
from the peer itself(both phones uses the same peer) and not from the
device(mobile phone) which i believe is more logical.
I also tried using DIALSTATUS(which of course you need to DIAL first), but
then I only hear a busy tone and the dialstatus will return a noanswer. Do I
have to configure it first in order to capture the busy status of a device?
Have you done something similar to this?
I'm using ver. 1.6. Thanks in advance.
I'm not sure I understand your setup. Are you using SIP for trunking, or
for extensions? Are you calling a normal mobile phone, or a SIP client
on a mobile phone?
Sebastian
regards,
RYAN ICASIANO
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of GBR Icasiano, Ryan A.
[raicasi...@globalbridgeresources.com]
Sent: Tuesday, October 26, 2010 10:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Mobile Phones and Asterisk
Hi,
Is the dev_state can also be used to track a mobile phone's status via SIP?
I tried it on several phones(nokia, samsung) but it returns NOANSWER but i
can hear a beep beep beep sound indicating that it is currently busy.
regards,
RYAN ICASIANO
__
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
[s...@open-t.co.uk]
Sent: Tuesday, October 26, 2010 7:50 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Mobile Phones and Asterisk
On 10/26/2010 12:30 PM, ayodele abejide wrote:
Hello Jonathan,
The solution would work only if the ISP has one public address, but in
my solution they have a pool of public address, any other possible solution?
With dynamic dns, you either install a piece of software on your server
(dynamic dns client) or you use the facility provided by your router
(some firewall/router/access point combo's have them). This software
updates automatically the record with dyndns every time your IP address
changes.
Sebastian
ABEJIDE, Ayodele A. (CCNA)
+2348039269311
From: ayodeleabej...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 26 Oct 2010 11:01:09 +
Subject: Re: [asterisk-users] Mobile Phones and Asterisk
thanks i would check it up
ABEJIDE, Ayodele A. (CCNA)
+2348039269311
Date: Tue, 26 Oct 2010 12:52:30 +0200
From: jonathan@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Mobile Phones and Asterisk
Try http://www.dyndns.com/ that should solve your problem with dynamic IPs.
Regards,
Jonathan
On Tue, Oct 26, 2010 at 12:40 PM, ayodele abejide
ayodeleabej...@hotmail.commailto:ayodeleabej...@hotmail.com wrote:
Dear Asterisk-Users,
I have this Asterisk Box I run in my house, I need to terminate and
originate remote calls through the box via internet
Re: [asterisk-users] being bombarded with SIP packets
Norbert Zawodsky wrote: Am 28.10.2010 09:41, schrieb Per Jessen: Over the last two weeks, we have had at least two incidents where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further). The immediate issues were dealt with by having the firewall drop those packets, but I was wondering: 1) if anyone has seen the same problem, and 2) if you've got some iptables rules for limiting inbound SIP by rate? (or some such). thanks Per Jessen, Zürich Hello Per, (iptables) rule #1: search the archives You will find nearly as many postings about that problem, as your server SIP packets received ... ;-) Thanks Norbert - I should take my own medicine, I'm usually the first to suggest searching the archives. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- 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] being bombarded with SIP packets
Ishfaq Malik wrote: On Thu, 2010-10-28 at 09:41 +0200, Per Jessen wrote: Over the last two weeks, we have had at least two incidents where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further). The immediate issues were dealt with by having the firewall drop those packets, but I was wondering: 1) if anyone has seen the same problem, and 2) if you've got some iptables rules for limiting inbound SIP by rate? (or some such). thanks Per Jessen, Zürich Was it legitimate requests or a brute force attack? If it was a brute force attack have you considered using fail2ban? It appears to be brute force, but I haven't bothered to investigate any further. fail2ban is at best a kludge IMHO, and I don't like anything (automatically or otherwise) modifying my firewall. Like Nortbert suggested, I'll check the archives to see what others have done. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- 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] Mobile Phones and Asterisk
Hi,
I can actually place a successful call using that configuration. The telco i'm
currently working requires the prefix.
What I'm trying to do is to capture the status of the mobile phone, if it is
currently engaged in a call or not. I achieved this successfully by emulating
it via a softphone, when I call a softphone and it is currently engaged in a
call, asterisk returns BUSY in DIALSTATUS and will automatically fallback to
the next step in the dialplan.
But this is not the case when applying it to the mobile phone. When the target
phone is currently engaged in a call, and I called the mobile phone, I can hear
a busy tone(which is alright, since the target phone is actually busy), but
it will wait until it timed out as defined in the DIAL cmd, and the var
DIALSTATUS returns NOANSWER, instead of BUSY, as if the mobile phone is
available and it was not answered at all.
It may also have to do on how the tones are being handled, or it can also be
that the mobile phone and the media gateway are the one talking to each other,
and asterisk cannot get the status of the phone itself.
regards,
RYAN ICASIANO
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
[s...@open-t.co.uk]
Sent: Thursday, October 28, 2010 5:27 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Mobile Phones and Asterisk
Hi,
On 10/28/2010 01:06 AM, GBR Icasiano, Ryan A. wrote:
Hi,
Thanks for your reply. I'm calling a normal phone using the DIAL cmd. Here is
my sample dial command:
exten =s,4,Dial(SIP/xxx${extensi...@media_gateway,10,t)
but when I use:
exten =s,5,NoOp(SIP/xxx${extensi...@media_gateway has state ${DIALSTATUS})
I'm not quite sure what you are trying to do.
So you called the phone for 10 seconds, the phone didn't answer - and
the variable DIALSTATUS told you exactly that.
Is the problem the fact that the line is not ringing out? Is that what
is wrong?
And why do you have some xxx in front of ${extension}? You shouldn't
need them. Just pass ${extension} - which is the number you dialled on
the phone.
Sebastian
I hear a busy tone, after the 10 sec. timeout it returns NOANSWER, as defined
in my DIAL func.
I also tried getting the DEVICE_STATE
exten =s,3,NoOp(SIP/xxx${extensi...@media_gateway has state
${DEVICE_STATE(SIP/xxx${extensi...@media_gateway)})
and same thing happens as stated on the scenario below.
Thanks again!
regards,
RYAN ICASIANO
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
[s...@open-t.co.uk]
Sent: Wednesday, October 27, 2010 5:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Mobile Phones and Asterisk
Hi,
On 10/27/2010 05:55 AM, GBR Icasiano, Ryan A. wrote:
anyone???
regards,
RYAN ICASIANO
Hi,
I changed my sip.conf and added call-limit. At first I thought it works ok,
since i tried calling a cellphone that is currently busy(phone answers 1st
softphone, then another softphone calls the same number, it now returns
INUSE). But then, i tried calling a different number while the first phone
is busy, but it returns INUSE. It seems that the status being returned was
from the peer itself(both phones uses the same peer) and not from the
device(mobile phone) which i believe is more logical.
I also tried using DIALSTATUS(which of course you need to DIAL first), but
then I only hear a busy tone and the dialstatus will return a noanswer. Do I
have to configure it first in order to capture the busy status of a device?
Have you done something similar to this?
I'm using ver. 1.6. Thanks in advance.
I'm not sure I understand your setup. Are you using SIP for trunking, or
for extensions? Are you calling a normal mobile phone, or a SIP client
on a mobile phone?
Sebastian
regards,
RYAN ICASIANO
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of GBR Icasiano, Ryan A.
[raicasi...@globalbridgeresources.com]
Sent: Tuesday, October 26, 2010 10:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Mobile Phones and Asterisk
Hi,
Is the dev_state can also be used to track a mobile phone's status via SIP?
I tried it on several phones(nokia, samsung) but it returns NOANSWER but i
can hear a beep beep beep sound indicating that it is currently busy.
regards,
RYAN ICASIANO
__
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
[s...@open-t.co.uk]
Sent: Tuesday, October 26, 2010 7:50 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Mobile Phones and Asterisk
On 10/26/2010 12:30 PM, ayodele abejide wrote:
Hello Jonathan,
The
Re: [asterisk-users] being bombarded with SIP packets
Am 28.10.2010 12:14, schrieb Per Jessen: Ishfaq Malik wrote: On Thu, 2010-10-28 at 09:41 +0200, Per Jessen wrote: Over the last two weeks, we have had at least two incidents where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further). The immediate issues were dealt with by having the firewall drop those packets, but I was wondering: 1) if anyone has seen the same problem, and 2) if you've got some iptables rules for limiting inbound SIP by rate? (or some such). thanks Per Jessen, Zürich Was it legitimate requests or a brute force attack? If it was a brute force attack have you considered using fail2ban? It appears to be brute force, but I haven't bothered to investigate any further. fail2ban is at best a kludge IMHO, and I don't like anything (automatically or otherwise) modifying my firewall. Like Nortbert suggested, I'll check the archives to see what others have done. /Per Jessen, Zürich Per, (didn't want to be unfriendly to you !) As you say, you don't like anything to modify your firewal. My words ! Someone (don't remember who when) on this list showed me a very clever trick (=iptables rule) to drop the packets if too many of them arrive within a given period of time. Works really great ! Do not exatly remember how it was done (and I don't have access to that machine at the moment to have a look). I remeber something like first using iptables module string to inspect the packet if it contains the string REGISTER sip: and then use an iptables hash bucket with a limit of x/second If this limit is exeeded, send the packet to nirvana (= DROP, or if you like LOG DROP, or if you like LOG the 1st DROP all .) Norbert -- _ -- 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 client floods port 5060 and gets blocked
On Thu, 28 Oct 2010, Jonas Kellens wrote: On 10/28/2010 10:44 AM, Kevin Keane wrote: I assume that you checked and the remote IP is a legitimate IP phone? If not, it could be an attempt to break into your system. If it is a legitimate IP phone, make sure that the SIP configuration is correct -- if the SIP authentication fails, you can see this happening. 1. This is a legitimate phone, yes. 2. Registration goes as follow : REGISTER SIP/2.0 401 Unauthorized Re-Register with Digest 200 OK Is it s Snom phone? I've seen Snoms do this... Gordon -- _ -- 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] being bombarded with SIP packets
On Thu, 28 Oct 2010, Norbert Zawodsky wrote: Am 28.10.2010 12:14, schrieb Per Jessen: Ishfaq Malik wrote: On Thu, 2010-10-28 at 09:41 +0200, Per Jessen wrote: Over the last two weeks, we have had at least two incidents where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further). The immediate issues were dealt with by having the firewall drop those packets, but I was wondering: 1) if anyone has seen the same problem, and This is not new - just Read The Fine Archives. Been going on for years. You're not the first, not the last. Google for sipvicious. 2) if you've got some iptables rules for limiting inbound SIP by rate? (or some such). thanks Per Jessen, Zürich Was it legitimate requests or a brute force attack? If it was a brute force attack have you considered using fail2ban? It appears to be brute force, but I haven't bothered to investigate any further. fail2ban is at best a kludge IMHO, and I don't like anything (automatically or otherwise) modifying my firewall. Like Nortbert suggested, I'll check the archives to see what others have done. /Per Jessen, Zürich Per, (didn't want to be unfriendly to you !) As you say, you don't like anything to modify your firewal. My words ! Someone (don't remember who when) on this list showed me a very clever trick (=iptables rule) to drop the packets if too many of them arrive within a given period of time. Works really great ! Possibly me - I did post something - you might want to look at http://unicorn.drogon.net/firewall2 An issue I've found with this is that is that while it works to protect your asterisk box, it does take up a considerable amount of CPU/kernel time to process - so running on embedded hardware isn't a good idea. There are other things you need to do to - but do get the sipvicious source code - it has a crash program in it - however I'm finding that this works less and less now because the criminals who're trying to steal your VoIP minutes have upgraded - however the upgrade is a little nicer when you firewall it out. And do make sure you have alwaysauthreject=yes in the [general] section of sip.conf. Most of the time that will protect you as the criminals will do a single pass to try to identify accounts that are valid, then find none, then move on. Sometimes they don't though and use the 'force' option in sipvicious. Then youy're SOL Gordon-- _ -- 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] being bombarded with SIP packets
Norbert Zawodsky wrote: Per, (didn't want to be unfriendly to you !) Not at all. As you say, you don't like anything to modify your firewal. My words! Someone (don't remember who when) on this list showed me a very clever trick (=iptables rule) to drop the packets if too many of them arrive within a given period of time. Works really great ! Yeah, I have a rule like that for SSH brute force attempts, and I did also find one for the same thing for SIP. Do not exatly remember how it was done (and I don't have access to that machine at the moment to have a look). I remeber something like first using iptables module string to inspect the packet if it contains the string REGISTER sip: and then use an iptables hash bucket with a limit of x/second This is what I found: iptables -N sip-flood iptables -A INPUT -p udp -m udp --dport 5060 -j sip-flood iptables -A INPUT -p tcp -m tcp --dport 5060:5061 --syn -j sip-flood iptables -A sip-flood -m recent --update --seconds 60 --hitcount 20 -j LOG --log-prefix SIP bruteforce attempt: iptables -A sip-flood -m recent --rcheck --seconds 60 --hitcount 20 -j DROP iptables -A sip-flood -m recent --set -j ACCEPT /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- 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 client floods port 5060 and gets blocked
On 10/28/2010 12:52 PM, Gordon Henderson wrote: On Thu, 28 Oct 2010, Jonas Kellens wrote On 10/28/2010 10:44 AM, Kevin Keane wrote: I assume that you checked and the remote IP is a legitimate IP phone? If not, it could be an attempt to break into your system. If it is a legitimate IP phone, make sure that the SIP configuration is correct -- if the SIP authentication fails, you can see this happening. 1. This is a legitimate phone, yes. 2. Registration goes as follow : REGISTER SIP/2.0 401 Unauthorized Re-Register with Digest 200 OK Is it s Snom phone? I've seen Snoms do this... Gordon I have this with Snom 320, Snom 370, Grandstream GXW4008 and YeaLink T28... Jonas. -- _ -- 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] being bombarded with SIP packets
Gordon Henderson wrote: On Thu, 28 Oct 2010, Norbert Zawodsky wrote: Am 28.10.2010 12:14, schrieb Per Jessen: Ishfaq Malik wrote: On Thu, 2010-10-28 at 09:41 +0200, Per Jessen wrote: Over the last two weeks, we have had at least two incidents where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further). The immediate issues were dealt with by having the firewall drop those packets, but I was wondering: 1) if anyone has seen the same problem, and This is not new - just Read The Fine Archives. Been going on for years. You're not the first, not the last. Well, to me it only started 3 days ago. Point taken though, I should have googled first. My main issue was not the brute force attempt in itself, but the increased latency it caused. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- 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] Integrating Asterisk 1.8 with Google Talk and Google Voice
I keep the AGI in Git as a version control system. But, you can view the AGI source here: http://messinet.com/trac/browser/gv/gv.agi And at the very bottom of that page is a link to download it as an individual file here: http://messinet.com/trac/export/b3229dbba3e01c887b3bdf6b0e0d93e897bd8a59/gv/gv.agi This is not the same thing as what is in the Changelog. I am using Asterisk 1.6 with this AGI. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E Thanks Anthony, Interestingly enough outbound dialing started working. Had no clue until someone called and told me my Google Chat status was updated. Is there a way to prevent Google Chat from staying logged in but still be able to dial outbound? People think I'm logged in persistently and send me messages that I miss. Even if I set a status message in asterisk most users are not going to understand... -Stephen -- _ -- 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] Integrating Asterisk 1.8 with Google Talk and Google Voice
On Thu, Oct 28, 2010 at 10:11 AM, Stephen Reese rsre...@gmail.com wrote: Is there a way to prevent Google Chat from staying logged in but still be able to dial outbound? People think I'm logged in persistently and send me messages that I miss. Even if I set a status message in asterisk most users are not going to understand... Had the same issue, but have not had a chance to find a good solution. You could change your status to DND. I tried invisible put seems not to be supported. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- 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 Communicator Friday at 12 Noon EDT
Friday we'll be hearing about SIP Communicator Java VoIP and Instant Messaging client. SIP Communicator is an audio/video Internet phone and instant messenger that supports some of the most popular VoIP and instant messaging protocols such as SIP, Jabber, AIM/ICQ, MSN, Yahoo! Messenger, Bonjour, IRC and a whole lot of other useful features. Open Source / Free Software, and is freely available under the terms of the GNU Lesser General Public License. http://vuc.me Call sip:200...@login.zipdx.com - note the bridge is up about 15 minutes before the scheduled time: http://vuc.me/next IRC #vuc on Freenode.net Hear you there? /r -- _ -- 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 Load Balancing
I have a very simple setup with two SIP routes to my carrier. I need to have
every other phone call placed to that carrier go to a different address.
This is what I need the call flow to look like. I have spent many hours
searching and have not found a working example.
Call1 exten = NXXNX,2,Dial(SIP/${dialedn...@2.4.6.8dialednum...@2.4.6.8
)
Call2 exten = NXXNX,2,Dial(SIP/${dialedn...@1.2.3.4dialednum...@1.2.3.4
)
Call3 exten = NXXNX,2,Dial(SIP/${dialedn...@2.4.6.8dialednum...@2.4.6.8
)
Call4 exten = NXXNX,2,Dial(SIP/${dialedn...@1.2.3.4dialednum...@1.2.3.4
)
Call5 exten = NXXNX,2,Dial(SIP/${dialedn...@2.4.6.8dialednum...@2.4.6.8
)
Call6 exten = NXXNX,2,Dial(SIP/${dialedn...@1.2.3.4dialednum...@1.2.3.4
)
Call7 exten = NXXNX,2,Dial(SIP/${dialedn...@2.4.6.8dialednum...@2.4.6.8
)
Call8 exten = NXXNX,2,Dial(SIP/${dialedn...@1.2.3.4dialednum...@1.2.3.4
)
..
--
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Re: [asterisk-users] SIP Load Balancing
- Original Message -
I have a very simple setup with two SIP routes to my carrier. I need to have
every other phone call placed to that carrier go to a different address.
This is what I need the call flow to look like. I have spent many hours
searching and have not found a working example.
Call1 exten = NXXNX,2,Dial(SIP/${ dialedn...@2.4.6.8 )
Call2 exten = NXXNX,2,Dial(SIP/${ dialedn...@1.2.3.4 )
Call3 exten = NXXNX,2,Dial(SIP/${ dialedn...@2.4.6.8 )
Call4 exten = NXXNX,2,Dial(SIP/${ dialedn...@1.2.3.4 )
Call5 exten = NXXNX,2,Dial(SIP/${ dialedn...@2.4.6.8 )
Call6 exten = NXXNX,2,Dial(SIP/${ dialedn...@1.2.3.4 )
Call7 exten = NXXNX,2,Dial(SIP/${ dialedn...@2.4.6.8 )
Call8 exten = NXXNX,2,Dial(SIP/${ dialedn...@1.2.3.4 )
..
--
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and give those IPs a single name ?
--
Thanks, Phil
--
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Re: [asterisk-users] SIP Load Balancing
On Thu, Oct 28, 2010 at 11:08:12AM -0400, Tim King wrote: I have a very simple setup with two SIP routes to my carrier. I need to have every other phone call placed to that carrier go to a different address. I think what you need to do here is check/set a variable in the astdb. (If the variable is 1, set it to 2 and route via A; otherwise, set it to 1 and route via B.) Translation of this to dialplan logic is left as an exercise for the student. R -- _ -- 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] Adhearsion 1.0 - Now Showing
Thanks to the hard work of many people in the Adhearsion community, I am pleased to be able to announce the immediate availability of Adhearsion version 1.0. Since Jay Phillips first began work on the project in 2006 Adhearsion has changed the way developers think about telephony applications. Now with several years of operating experience and multitudes of applications deployed to production, it is time to acknowledge this important milestone. What does Adhearsion 1.0 mean? • A battle-tested API. Adhearsion 1.0 has defined a well-tested API and it has been proven with over two years of real-world deployment experience. • A “stable” branch. Adhearsion’s exposed API will remain stable throughout this major version number series. No backward-incompatible changes will be made, making it safer for developers to trust future upgrades. • Updated documentation. Thanks to Justin Dupree of Tropo, Adhearsion’s docs received a lot of TLC in the form of content updates and a migration to Github Wiki. Check them out at docs.adhearsion.com • Gem-based Components. The final feature added to Adhearsion prior to 1.0 is the ability to install and use components via RubyGems. Learn more about that in the Gem-based Components wiki page here (http://github.com/adhearsion/adhearsion/wiki/Gem-based-components). To get your hands on Adhearsion 1.0 run, don’t walk, to your nearest command line and issue a sudo gem install adhearsion. As always, don’t be a stranger. We can be found on the Adhearsion mailing list (http://groups.google.com/group/adhearsion), in IRC (irc.freenode.net #adhearsion) and on the Adhearsion website (http://adhearsion.com). PS: Come to the formal release announcement at AstriCon, room 6, at 1:45PM today! /BAK/ -- Ben Klang Adhearsion Project, Lead Maintainer b...@alkaloid.net http://adhearsion.com -- Ben Klang Alkaloid Networks LLC b...@alkaloid.net 404.475.4850 http://projects.alkaloid.net -- _ -- 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] being bombarded with SIP packets
Fail2Ban Regards - Original Message - From: Per Jessen p...@computer.org To: asterisk-users@lists.digium.com Sent: Thursday, October 28, 2010 2:41 AM Subject: [asterisk-users] being bombarded with SIP packets Over the last two weeks, we have had at least two incidents where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further). The immediate issues were dealt with by having the firewall drop those packets, but I was wondering: 1) if anyone has seen the same problem, and 2) if you've got some iptables rules for limiting inbound SIP by rate? (or some such). thanks Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- 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] being bombarded with SIP packets
Two incidents in two weeks is not bad. I get 2-4 a day. There must be many here with even more than that. You should start considering some safety practices like disabling long distance and international calls by default, put a cap on long distance and international calls even for genuine users, and who don't want to have caps, get their consent that they'll not argue with you if their accounts are hacked. Probably do prepaid billing at least for long distance and international calls. Other than that, fail2ban is a must have. Detailed installation instructions you can find at voip-info.org website and also in my blogs at ilovetovoip.com. Regards, Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-28 3:48 AM, Per Jessen p...@computer.org wrote: Over the last two weeks, we have had at least two incidents where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further). The immediate issues were dealt with by having the firewall drop those packets, but I was wondering: 1) if anyone has seen the same problem, and 2) if you've got some iptables rules for limiting inbound SIP by rate? (or some such). thanks Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- 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] SIP Load Balancing
Sorry for the confusion, but the last sentence throws me off. Translation of this to dialplan logic is left as an exercise for the student. Is this example from some sort of book or is this a way of saying I am left to figure the rest out?? I was hoping to find a simple example of how this works. On Thu, Oct 28, 2010 at 11:24 AM, Roger Burton West ro...@firedrake.orgwrote: On Thu, Oct 28, 2010 at 11:08:12AM -0400, Tim King wrote: I have a very simple setup with two SIP routes to my carrier. I need to have every other phone call placed to that carrier go to a different address. I think what you need to do here is check/set a variable in the astdb. (If the variable is 1, set it to 2 and route via A; otherwise, set it to 1 and route via B.) Translation of this to dialplan logic is left as an exercise for the student. R -- _ -- 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] generic_odbc and ltdl are not available to enableODBC support
Hi, are you installed unixodbc-dev? Regards -- _ -- 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 Load Balancing
Thanks For The replies. I have tried piecing the samples together. Just for
testing purposes i have created the following.
[test]
exten =
_X.,1,GotoIf(${SET(DB(avoics/route)=$[!0${DB(avoics/route)}])}?route1:route2)
exten = _X.,n(route1),Set(DB(avoics/route)=1)
exten = _X.,n,SayNumber(1)
exten = _X.,n,Hangup()
exten = _X.,n(route2),Set(DB(avoics/route)=0)
exten = _X.,n,SayNumber(2)
exten = _X.,n,Hangup()
The idea is if I continue dialing any number into this context I should hear
1 2 1 2 1 2
Currently it is skipping to 2 as it should be since my database shows:
/avoics/route : 1
It appears there is something wrong with my set command?
On Thu, Oct 28, 2010 at 2:15 PM, Tilghman Lesher tles...@digium.com wrote:
On Thursday 28 October 2010 13:06:00 Gordon Henderson wrote:
On Thu, 28 Oct 2010, Tim King wrote:
On Thu, Oct 28, 2010 at 11:24 AM, Roger Burton West
ro...@firedrake.orgwrote:
On Thu, Oct 28, 2010 at 11:08:12AM -0400, Tim King wrote:
I have a very simple setup with two SIP routes to my carrier. I need
to
have
every other phone call placed to that carrier go to a different
address.
I think what you need to do here is check/set a variable in the
astdb.
(If the variable is 1, set it to 2 and route via A; otherwise, set it
to 1 and route via B.)
Translation of this to dialplan logic is left as an exercise for the
student.
Sorry for the confusion, but the last sentence throws me off.
Translation of this to dialplan logic is left as an exercise for the
student. Is this example from some sort of book or is this a way of
saying I am left to figure the rest out??
I was hoping to find a simple example of how this works.
It's a way of leafing you to figure the rest out.
It's a bastardised version of a quote from many textbooks - along the
lines of implementation is left as an excercise to the student - ie.
this is the method in general terms, you write nuts bolts of the code.
One reference to it might be:
http://catb.org/jargon/html/E/exercise--left-as-an.html
Roger has told you how to do it - use a variable kept in the astdb and
alternate it
In pseudo code:
if (switch == 1)
Dial (SIP/provider1/number)
switch = 0
else
Dial (SIP/provider2/number
switch = 1
endif
Now your task is write the actual dialplan. Or you can pay me or Roger
to do it for you if you like, but really, it's only a few lines of
dialplan.
GotoIf(${SET(DB(sw/provider)=$[!0${DB(sw/provider)}])}?provider1:provider2)
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] SIP Load Balancing
It seems that the
GotoIf(${SET(DB(avoics/route)=$[!0${DB(avoics/route)}])}?route1:route2) is
always returning false as if the SET command is not returning a value nor is
it changing the value in the DB.
Will this not work because I am running Asterisk 1.4.25.1??
On Thu, Oct 28, 2010 at 3:15 PM, Tim King t...@compnetwork.net wrote:
I updated it as follows and I am still only getting the SayNumber(2)
[tim]
exten =
_X.,1,GotoIf(${SET(DB(avoics/route)=$[!0${DB(avoics/route)}])}?route1:route2)
exten = _X.,n(route1),SayNumber(1)
exten = _X.,n,Hangup()
exten = _X.,n(route2),SayNumber(2)
exten = _X.,n,Hangup()
On Thu, Oct 28, 2010 at 3:05 PM, Tilghman Lesher tles...@digium.comwrote:
On Thursday 28 October 2010 13:32:51 Tim King wrote:
Thanks For The replies. I have tried piecing the samples together. Just
for testing purposes i have created the following.
[test]
exten =
_X.,1,GotoIf(${SET(DB(avoics/route)=$[!0${DB(avoics/route)}])}?route1:ro
ute2) exten = _X.,n(route1),Set(DB(avoics/route)=1)
exten = _X.,n,SayNumber(1)
exten = _X.,n,Hangup()
exten = _X.,n(route2),Set(DB(avoics/route)=0)
exten = _X.,n,SayNumber(2)
exten = _X.,n,Hangup()
The idea is if I continue dialing any number into this context I should
hear 1 2 1 2 1 2
Currently it is skipping to 2 as it should be since my database shows:
/avoics/route : 1
It appears there is something wrong with my set command?
You can drop your separate Set application. The SET() dialplan function
does the alternation for you.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] being bombarded with SIP packets
On 10/28/2010 3:41 AM, Per Jessen wrote: 2) if you've got some iptables rules for limiting inbound SIP by rate? exactly what i was going through; here's how i reacted (throttles both SSH and SIP Register: First, I completely blocked all non-North American Amazon EC2 networks - I won't be registering my sip phone in Nigeria nor from within EC2* any time soon. Then in my iptables startup script: iptables -N THROTTLE iptables -A INPUT -i eth0 -p udp --dport 5060 \ -m string --string REGISTER sip: --algo bm --to 65 -j THROTTLE iptables -A INPUT -i eth0 -p tcp --dport 22 \ -m state --state NEW -j THROTTLE iptables -A THROTTLE -m recent --set --name ABUSE iptables -A THROTTLE -m recent --update --seconds 86400 \ --hitcount 15 --name ABUSE -j LOG $LOGOPTS $PREh15_ iptables -A THROTTLE -m recent --rcheck --seconds 86400 \ --hitcount 15 --name ABUSE -j DROP iptables -A THROTTLE -m recent --update --seconds 3600 \ --hitcount 12 --name ABUSE -j LOG $LOGOPTS $PREh12_ iptables -A THROTTLE -m recent --rcheck --seconds 3600 \ --hitcount 12 --name ABUSE -j DROP iptables -A THROTTLE -m recent --update --seconds 60\ --hitcount 6 --name ABUSE -j LOG $LOGOPTS $PREh6_ iptables -A THROTTLE -m recent --rcheck --seconds 60\ --hitcount 6 --name ABUSE -j DROP iptables -A INPUT -i eth0 -p udp --dport 5060 \ --sport 1024:65535 -j ACCEPT iptables -A INPUT -i eth0 -p tcp --dport 22 \ --sport 1024:65535 -j ACCEPT Note that some SIP clients send more than one register per startup -- e.g.: Siphon on the iPhone registers without credentials first, asterisk sends back unauthorized, then Siphone tries again with the configured username and password. For exactly how i'm using it: mkdir /usr/local/script cd /usr/local/script wget http://jeremy.kister.net/code/iptables/make-non-na.pl wget http://jeremy.kister.net/code/iptables/iptables.init mv iptables.init /etc/init.d/iptables # vi iptables # change the MYLAN to your lan network # change the RDPRANGE to the range defined in /etc/asterisk/rdp.conf ln -s /etc/init.d/iptables /etc/rc2.d/iptables ln -s /etc/init.d/iptables /etc/rc3.d/iptables crontab -e # put in something to run the make-non-na.pl run once per week /usr/local/script/make-non-na.pl /etc/init.d/iptables start * = if you use the Acrobits softphone, you'll need to let EC2 through for push notifications. Currently, I just put 184.72.221.84 in the siprtp section of the iptables script. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Mobile Phones and Asterisk
Hi,
On 10/28/2010 11:20 AM, GBR Icasiano, Ryan A. wrote:
Hi,
I can actually place a successful call using that configuration. The telco
i'm currently working requires the prefix.
What I'm trying to do is to capture the status of the mobile phone, if it is
currently engaged in a call or not.
Maybe others who know better will jump in - but I seriously doubt you
will be able to do this. From my limited knowledge, I believe mobile
phone networks use different signalling then regular terrestrial based
providers. I don't really think that the engaged tone sent back by the
mobile operator will be decoded correctly by Asterisk.
Not to mention that, I don't what happens where you are - but in UK for
example - you don't even get an engaged tone from a mobile phone. You
just get either sent to the user's voice mail, or you are played a
message from the mobile phone operator which essentially tells you that
the user is engaged or unavailable. Operators in many other European
countries do the same. So from the point of what you are trying to
achieve - this is useless in Asterisk.
I would have liked to do the same thing - as I have line divert in
Asterisk to my mobile phone - and I would have liked for Asterisk to
just skip along to my Asterisk voice mail when my mobile is either out
of coverage, or when I'm in a conversation on it. But no such luck. I
believe the mobile operators wouldn't like the idea anyway - as they get
to charge you extra for playing all those messages or sending you to
their voicemail.
I believe in parts of the North American continent things are similar,
but even worse. As the caller gets charged as soon as the mobile phone
starts ringing - apparently simply the act of accessing the mobile
operator's network is chargeable - never mind if you get to speak to
anybody or not.
Then again, maybe things are different where you are - and maybe there
is a way to get Asterisk to recognise the busy tone from your mobile
operator. Maybe somebody here will jump in with a suggestion. It seems
that it has to do with busy signalling in Asterisk. A softphone I
believe will accomplish this out of band - with some commands over SIP.
While PSTN (normal phone lines) and mobiles I believe tend to signal
this with inband tones (part of the sound coming down the line).
You might also want to check your regional settings in Asterisk.
Sebastian
I achieved this successfully by emulating it via a softphone, when I
call a softphone and it is currently engaged in a call, asterisk returns
BUSY in DIALSTATUS and will automatically fallback to the next step in
the dialplan.
But this is not the case when applying it to the mobile phone. When the
target phone is currently engaged in a call, and I called the mobile phone, I
can hear a busy tone(which is alright, since the target phone is actually
busy), but it will wait until it timed out as defined in the DIAL cmd, and
the var DIALSTATUS returns NOANSWER, instead of BUSY, as if the mobile
phone is available and it was not answered at all.
It may also have to do on how the tones are being handled, or it can also be
that the mobile phone and the media gateway are the one talking to each
other, and asterisk cannot get the status of the phone itself.
regards,
RYAN ICASIANO
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
[s...@open-t.co.uk]
Sent: Thursday, October 28, 2010 5:27 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Mobile Phones and Asterisk
Hi,
On 10/28/2010 01:06 AM, GBR Icasiano, Ryan A. wrote:
Hi,
Thanks for your reply. I'm calling a normal phone using the DIAL cmd. Here
is my sample dial command:
exten =s,4,Dial(SIP/xxx${extensi...@media_gateway,10,t)
but when I use:
exten =s,5,NoOp(SIP/xxx${extensi...@media_gateway has state ${DIALSTATUS})
I'm not quite sure what you are trying to do.
So you called the phone for 10 seconds, the phone didn't answer - and
the variable DIALSTATUS told you exactly that.
Is the problem the fact that the line is not ringing out? Is that what
is wrong?
And why do you have some xxx in front of ${extension}? You shouldn't
need them. Just pass ${extension} - which is the number you dialled on
the phone.
Sebastian
I hear a busy tone, after the 10 sec. timeout it returns NOANSWER, as
defined in my DIAL func.
I also tried getting the DEVICE_STATE
exten =s,3,NoOp(SIP/xxx${extensi...@media_gateway has state
${DEVICE_STATE(SIP/xxx${extensi...@media_gateway)})
and same thing happens as stated on the scenario below.
Thanks again!
regards,
RYAN ICASIANO
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
[s...@open-t.co.uk]
Sent: Wednesday, October 27, 2010 5:00 PM
To:
Re: [asterisk-users] google voice + asterisk: calls made to GV# processed but weird
Consider this RESOLVED thanks to the help of [David Vossel](http://www.davidvossel.com/?p=162) (*HIGH FIVE*) and the new wiki entry from [Malcolm Davenport](https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google). The trick was the following in extensions.conf: exten = s,1,Answer() exten = s,n,Wait(2) ;; THIS exten = s,n,SendDTMF(1) ;; AND THIS ARE NEEDED exten = s,n,Background(tnttspWelcome) exten = s,n,Background(CurrentAnnouncement) exten = s,n,Goto(0,1) -- Vinh On Tue, Oct 26, 2010 at 7:07 PM, Vinh Nguyen vinhdi...@gmail.com wrote: Can anyone reproduce this with their google voice number? Wondering whether this issue is just me or not, or whether I am misunderstanding the capabilities of incorporating GV with asterisk. Thanks. Vinh -- _ -- 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] google voice + asterisk: calls made to GV# processed but weird
On Thu, Oct 28, 2010 at 7:30 PM, Vinh Nguyen vinhdi...@gmail.com wrote: Consider this RESOLVED thanks to the help of [David Vossel](http://www.davidvossel.com/?p=162) (*HIGH FIVE*) and the new wiki entry from [Malcolm Davenport](https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google). I managed to finally get a GV number while at Astricon. I hope to play with this more next week. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- 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] MGCP
Hi I have asterisk 1.4 I want to make a MGCP trunk as a client to connect to a provider who is using MGCP protocol, he provided me with user password, I tried a custom trunk: MGCP/$outn...@user:passw...@66.152.163.106:4000 Not seems to help, Any suggestions plz? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MGCP
On Fri, Oct 29, 2010 at 4:21 AM, Baha @ SH i...@saudihome.com wrote: Hi I have asterisk 1.4 I want to make a MGCP trunk as a client to connect to a provider who is using MGCP protocol, he provided me with user password, I tried a custom trunk: MGCP/$outn...@user:passw...@66.152.163.106:4000 Not seems to help, Any suggestions plz? In my research to try to get MEGACO protocol to work (they are very similar) I remember stumbling onto this information. I am not sure where or if it is even correct, it has been so long. MGCP is supported but only the MGCP phones, not the entire protocol. I tried to get full support for MEGCO because many NEC systems at that time used it for VoIP and the only other way to interface with an NEC IPK was via TDM, either POTS or T1, which are and especially were, WAY too expensive. I am out of date with NEC, I haven't touched one in years, but way back then, they were in the top three of PBX market share. You can probably google my name and MEGACO and you may find the info you are looking for. Thanks, Steve Totaro -- _ -- 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] MGCP
On Thu, Oct 28, 2010 at 9:54 PM, Steve Totaro stot...@totarotechnologies.com wrote: On Fri, Oct 29, 2010 at 4:21 AM, Baha @ SH i...@saudihome.com wrote: Hi I have asterisk 1.4 I want to make a MGCP trunk as a client to connect to a provider who is using MGCP protocol, he provided me with user password, I tried a custom trunk: MGCP/$outn...@user:passw...@66.152.163.106:4000 Not seems to help, Any suggestions plz? In my research to try to get MEGACO protocol to work (they are very similar) I remember stumbling onto this information. I am not sure where or if it is even correct, it has been so long. MGCP is supported but only the MGCP phones, not the entire protocol. I tried to get full support for MEGCO because many NEC systems at that time used it for VoIP and the only other way to interface with an NEC IPK was via TDM, either POTS or T1, which are and especially were, WAY too expensive. I am out of date with NEC, I haven't touched one in years, but way back then, they were in the top three of PBX market share. You can probably google my name and MEGACO and you may find the info you are looking for. Thanks, Steve Totaro Straight from the mouth of BKW three years ago. http://www.spinics.net/lists/asterisk/msg76756.html -- _ -- 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] Mobile Phones and Asterisk
Hi,
Thanks for your very informative response. This is really helpful. I wouldn't
be pushing it though since it isn't possible as of now.
Kudos!
RYAN ICASIANO
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
[s...@open-t.co.uk]
Sent: Friday, October 29, 2010 5:50 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Mobile Phones and Asterisk
Hi,
On 10/28/2010 11:20 AM, GBR Icasiano, Ryan A. wrote:
Hi,
I can actually place a successful call using that configuration. The telco
i'm currently working requires the prefix.
What I'm trying to do is to capture the status of the mobile phone, if it is
currently engaged in a call or not.
Maybe others who know better will jump in - but I seriously doubt you
will be able to do this. From my limited knowledge, I believe mobile
phone networks use different signalling then regular terrestrial based
providers. I don't really think that the engaged tone sent back by the
mobile operator will be decoded correctly by Asterisk.
Not to mention that, I don't what happens where you are - but in UK for
example - you don't even get an engaged tone from a mobile phone. You
just get either sent to the user's voice mail, or you are played a
message from the mobile phone operator which essentially tells you that
the user is engaged or unavailable. Operators in many other European
countries do the same. So from the point of what you are trying to
achieve - this is useless in Asterisk.
I would have liked to do the same thing - as I have line divert in
Asterisk to my mobile phone - and I would have liked for Asterisk to
just skip along to my Asterisk voice mail when my mobile is either out
of coverage, or when I'm in a conversation on it. But no such luck. I
believe the mobile operators wouldn't like the idea anyway - as they get
to charge you extra for playing all those messages or sending you to
their voicemail.
I believe in parts of the North American continent things are similar,
but even worse. As the caller gets charged as soon as the mobile phone
starts ringing - apparently simply the act of accessing the mobile
operator's network is chargeable - never mind if you get to speak to
anybody or not.
Then again, maybe things are different where you are - and maybe there
is a way to get Asterisk to recognise the busy tone from your mobile
operator. Maybe somebody here will jump in with a suggestion. It seems
that it has to do with busy signalling in Asterisk. A softphone I
believe will accomplish this out of band - with some commands over SIP.
While PSTN (normal phone lines) and mobiles I believe tend to signal
this with inband tones (part of the sound coming down the line).
You might also want to check your regional settings in Asterisk.
Sebastian
I achieved this successfully by emulating it via a softphone, when I
call a softphone and it is currently engaged in a call, asterisk returns
BUSY in DIALSTATUS and will automatically fallback to the next step in
the dialplan.
But this is not the case when applying it to the mobile phone. When the
target phone is currently engaged in a call, and I called the mobile phone, I
can hear a busy tone(which is alright, since the target phone is actually
busy), but it will wait until it timed out as defined in the DIAL cmd, and
the var DIALSTATUS returns NOANSWER, instead of BUSY, as if the mobile
phone is available and it was not answered at all.
It may also have to do on how the tones are being handled, or it can also be
that the mobile phone and the media gateway are the one talking to each
other, and asterisk cannot get the status of the phone itself.
regards,
RYAN ICASIANO
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
[s...@open-t.co.uk]
Sent: Thursday, October 28, 2010 5:27 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Mobile Phones and Asterisk
Hi,
On 10/28/2010 01:06 AM, GBR Icasiano, Ryan A. wrote:
Hi,
Thanks for your reply. I'm calling a normal phone using the DIAL cmd. Here
is my sample dial command:
exten =s,4,Dial(SIP/xxx${extensi...@media_gateway,10,t)
but when I use:
exten =s,5,NoOp(SIP/xxx${extensi...@media_gateway has state ${DIALSTATUS})
I'm not quite sure what you are trying to do.
So you called the phone for 10 seconds, the phone didn't answer - and
the variable DIALSTATUS told you exactly that.
Is the problem the fact that the line is not ringing out? Is that what
is wrong?
And why do you have some xxx in front of ${extension}? You shouldn't
need them. Just pass ${extension} - which is the number you dialled on
the phone.
Sebastian
I hear a busy tone, after the 10 sec. timeout it returns NOANSWER, as
defined in my DIAL func.
I also tried getting the DEVICE_STATE
exten
Re: [asterisk-users] Mobile Phones and Asterisk
On 10/28/2010 11:18 PM, GBR Icasiano, Ryan A. wrote: Here is what I do today and it works fine: - asterisk/trixbox - Dext/android phone - Bell Canada cell provider - call comes in, to an extension with voicemail - rings a bunch of sip devices (real phones, and the android via linphone if it happens to be near wifi and registered (set to only use wifi not 3g to register) - if not answered call is forwarded back out a pots line and dials the cell number (cell is not subscribed to provider voicemail) - still no answer that pots line is hung up and call drops back into the original extension's vm. (I have not run into a problem with answer detection, only that people don't stay on the line long enough for me to answer on the second set of ringing, but if they are that impatient the call was probably not important anyway) outgoing calls if registered I have a choice once I dial of linphone or dialer to make the call. checking vm is just *98ext from linphone as the dialing app, or dial in and navigate to vm. linphone is a little less polished gui but seems to work the best for me to reliably register when it should. (tried about 5 different sip clients) Hi, Thanks for your very informative response. This is really helpful. I wouldn't be pushing it though since it isn't possible as of now. Kudos! RYAN ICASIANO From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian [s...@open-t.co.uk] Sent: Friday, October 29, 2010 5:50 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Mobile Phones and Asterisk Hi, On 10/28/2010 11:20 AM, GBR Icasiano, Ryan A. wrote: Hi, I can actually place a successful call using that configuration. The telco i'm currently working requires the prefix. What I'm trying to do is to capture the status of the mobile phone, if it is currently engaged in a call or not. Maybe others who know better will jump in - but I seriously doubt you will be able to do this. From my limited knowledge, I believe mobile phone networks use different signalling then regular terrestrial based providers. I don't really think that the engaged tone sent back by the mobile operator will be decoded correctly by Asterisk. Not to mention that, I don't what happens where you are - but in UK for example - you don't even get an engaged tone from a mobile phone. You just get either sent to the user's voice mail, or you are played a message from the mobile phone operator which essentially tells you that the user is engaged or unavailable. Operators in many other European countries do the same. So from the point of what you are trying to achieve - this is useless in Asterisk. I would have liked to do the same thing - as I have line divert in Asterisk to my mobile phone - and I would have liked for Asterisk to just skip along to my Asterisk voice mail when my mobile is either out of coverage, or when I'm in a conversation on it. But no such luck. I believe the mobile operators wouldn't like the idea anyway - as they get to charge you extra for playing all those messages or sending you to their voicemail. I believe in parts of the North American continent things are similar, but even worse. As the caller gets charged as soon as the mobile phone starts ringing - apparently simply the act of accessing the mobile operator's network is chargeable - never mind if you get to speak to anybody or not. Then again, maybe things are different where you are - and maybe there is a way to get Asterisk to recognise the busy tone from your mobile operator. Maybe somebody here will jump in with a suggestion. It seems that it has to do with busy signalling in Asterisk. A softphone I believe will accomplish this out of band - with some commands over SIP. While PSTN (normal phone lines) and mobiles I believe tend to signal this with inband tones (part of the sound coming down the line). You might also want to check your regional settings in Asterisk. Sebastian I achieved this successfully by emulating it via a softphone, when I call a softphone and it is currently engaged in a call, asterisk returns BUSY in DIALSTATUS and will automatically fallback to the next step in the dialplan. But this is not the case when applying it to the mobile phone. When the target phone is currently engaged in a call, and I called the mobile phone, I can hear a busy tone(which is alright, since the target phone is actually busy), but it will wait until it timed out as defined in the DIAL cmd, and the var DIALSTATUS returns NOANSWER, instead of BUSY, as if the mobile phone is available and it was not answered at all. It may also have to do on how the tones are being handled, or it can also be that the mobile phone and the media gateway are the one talking to each other, and asterisk cannot get the status of the phone itself.
Re: [asterisk-users] Re : saturation of bandwidth because of HANGUP
On Mon, Oct 25, 2010 at 11:02 AM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *ALAEDDINE abbech *Sent:* Monday, October 25, 2010 10:52 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Re : saturation of bandwidth because of HANGUP Any news for this problem. Who has this problem --- En date de : *Jeu 21.10.10, ALAEDDINE abbech alasup...@yahoo.fr* a écrit : De: ALAEDDINE abbech alasup...@yahoo.fr Objet: saturation of bandwidth because of HANGUP À: asterisk-users@lists.digium.com Date: Jeudi 21 octobre 2010, 17h55 Hello, I have a problem of saturation of bandwidth because of HANGUP which sends thousands of times per second for a single call. Furthermore, the timestamp is still the same for this HANGUP. Thanks If we had this problem, either we would have posted a reply or would be too busy figuring it out ourselves. I personally don’t believe Asterisk would send out 1000 hangups in 1 second (how would you monitor this? AMI output? /var/log/asterisk/full?) When you wonder if anybody is reading your post, check the archives to see if it actually got there. I’m not going to be arrogant enough to tell you that Asterisk is a U.S. based audience (the posts I get indicate that there are a Large contingent of UK, Indian posters), but do keep in mind that lots of the posters (hopefully) work and post on a primarily 5 day workweek. -- _ -- 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 Furthermore, a little more information in your request would be helpful. I just noticed your thread, but there's basically no information concerning the situation surrounding this strange occurrence. Let me ask you for clarifications, maybe that will help get you going. 1. Are you saying that there's a multitude of SIP BYE requests in the signalling between yourself and your provider? 2. If your answer to #1 is no, then please explain how this problem is saturating your bandwidth, because I'm lost. 3. Is the hangup request coming FROM you or FROM the provider? 4. Does it happen only on certain calls, or when specific conditions are met, or is it EVERY time there is a call running. Those will help me to even begin to help you. Posting a sanitized section of the CLI output and/or from your log(s) during this problem would also be MORE than helpful, it would also get you a better chance at a response. Cheers, Sherwood McGowan That guy...you know, the one who is a VOIP Engineer but has a mohawk and tattoos LOL -- _ -- 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] Re : thousands Hangup per second /saturation of bandwidth
On Mon, Oct 25, 2010 at 11:11 AM, Steve Edwards asterisk@sedwards.com wrote: Un-self-top-posting... --- En date de : Jeu 21.10.10, ALAEDDINE abbech alasup...@yahoo.fr a écrit : De: ALAEDDINE abbech alasup...@yahoo.fr Objet: thousands Hangup per second /saturation of bandwidth À: asterisk-users@lists.digium.com Date: Jeudi 21 octobre 2010, 11h42 Hello, I have a problem of saturation of bandwidth because of HANGUP which sends thousands of times per second for a single call. Furthermore, the timestamp is still the same for this HANGUP. Thanks On Mon, 25 Oct 2010, ALAEDDINE abbech wrote: Any news for this problem. Who has this problem Why you don't answer. 0) This is a volunteer list. Nobody is obligated to answer. 1) Maybe nobody else has experienced this problem. 2) Maybe you failed to provide any information that would allow anybody to offer any suggestions of how to resolve your problem. Let's start with some simple details... a) What OS and version? b) What version of Asterisk? c) What technology is used for the failing call? I'm assuming SIP... d) What endpoint is involved? For example, Cisco 7960 with 8.3 firmware. e) What does your dialplan look like? Please use show dialplan so we can see what Asterisk sees. f) What does the Asterisk console output show after upping debug and verbose levels. g) Can the problem be replicated with a different endpoint? For example, a Zoiper Communicator 1.18.6898 softphone? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 Let me also note for the original posterAbsolutely NOBODY likes doubleposters (or crossposters) who are asking for help. Try reading my response to your OTHER post regarding this. Also, I second everything that Mr Edwards said... I'm still at a loss as to why people a) get pissy because no-one responds to a help request on an all volunteer list, and/or b) don't include even the most base information that would enable said volunteers to begin to help. Ahh sigh. The Mick -- _ -- 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
