[asterisk-users] Outgoing routes with two PRI
Hi,
I have 2 port Digium PRI card (TE205P). On each port we have different
service provider E1 is connected.
I have to configure if i dial with prefix 910 it has to dial out
through port 1 service provider and if i dial prefix 912 it has to
dial out through port 2.
My zapatel.conf:
[trunkgroups]
[channels]
callgroup=0
pickupgroup=1
switchtype=euroisdn
language=en
signalling=pri_cpe
usecallerid=yes
callerid=asreceived
callprogress=no
busydetect=no
context=from-pstn
echocancel=no
echocancelwhenbridged=yes
resetinterval=never
priindication=outofband
group=0
channel = 1-15,17-31
group=1
channel = 32-46,48-62
vi extensions_additional.conf
OUT_1 = ZAP/g0
OUTCID_1 =
OUTMAXCHANS_1 =
OUTFAIL_1 =
OUTPREFIX_1 =
OUTDISABLE_1 = off
OUTKEEPCID_1 = off
FORCEDOUTCID_1 =
OUT_3 = ZAP/g1
OUTCID_3 =
OUTMAXCHANS_3 =
OUTFAIL_3 =
OUTPREFIX_3 =
OUTDISABLE_3 = off
OUTKEEPCID_3 = off
FORCEDOUTCID_3 =
[outbound-allroutes]
include = outrt-004-outbout
include = outrt-005-newout
exten = _!,1,Macro(user-callerid,SKIPTTL,)
; end of [outbound-allroutes]
[outrt-004-outbout]
include = outrt-004-outbout-custom
exten = _912.,1,Macro(user-callerid,SKIPTTL,)
exten = _912.,n,Set(_NODEST=)
exten = _912.,n,Macro(record-enable,${AMPUSER},OUT,)
exten = _912.,n,Macro(dialout-trunk,3,${EXTEN},,)
exten = _912.,n,Macro(outisbusy,)
; end of [outrt-004-outbout]
[outrt-005-newout]
include = outrt-005-newout-custom
exten = _910.,1,Macro(user-callerid,SKIPTTL,)
exten = _910.,n,Set(_NODEST=)
exten = _910.,n,Macro(record-enable,${AMPUSER},OUT,)
exten = _910.,n,Macro(dialout-trunk,1,${EXTEN},,)
exten = _910.,n,Macro(outisbusy,)
; end of [outrt-005-newout]
But when i dial it gives error all circuits are busy. Is there
anything i have missed.
Thanks Regards
Chima.s
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Re: [asterisk-users] Being attacked by an Amazon EC2 ...
On Apr 12, 2010, at 9:12 AM, --[ UxBoD ]-- wrote: Perhaps if there was a Asterisk RBL we could all contribute to; for which we could then hook into and drop any connection where a source IP is listed ? -- Thanks, Phil I love the idea of a RBL... count me in for contributing. Especially considering the ridiculous response I received from Amazon. (Basically told me to submit host, destination, port, proto, and log... which of course was already included in the original complaint) ---fred http://qxork.com -- _ -- 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 attacked by an Amazon EC2 ...
- Original Message - Am 11.04.2010 17:05, schrieb Mark Smith: Same this end from 184.73.17.150. Use this little piece of iptables magic to block the whole of Amazon's EC2 ip- range. iptables -F iptables -A INPUT -m iprange --src-range 216.182.224.0-216.182.239.255 -j DROP iptables -A INPUT -m iprange --src-range 72.44.32.0-72.44.63.255 -j DROP iptables -A INPUT -m iprange --src-range 67.202.0.0-67.202.63.255 -j DROP iptables -A INPUT -m iprange --src-range 75.101.128.0-75.101.255.255 -j DROP iptables -A INPUT -m iprange --src-range 174.129.0.0-174.129.255.255 -j DROP iptables -A INPUT -m iprange --src-range 204.236.192.0-204.236.255.255 -j DROP iptables -A INPUT -m iprange --src-range 184.73.0.0-184.73.255.255 -j DROP iptables -A INPUT -m iprange --src-range 216.236.128.0-216.236.191.255 -j DROP iptables -A INPUT -m iprange --src-range 184.72.0.0-184.72.63.255 -j DROP iptables -A INPUT -m iprange --src-range 79.125.0.0-79.125.127.255 -j DROP service iptables save This sorts it out in the short-term until Amazon realise their service is being utilised by arseholes. Hi Mark! your little iptables magic is a very good idea! Implementation took 1 minute :-) I'll use it until a better idea comes up ... (which I don't expect within a short term) Thank you! Norbert Perhaps if there was a Asterisk RBL we could all contribute to; for which we could then hook into and drop any connection where a source IP is listed ? -- Thanks, Phil -- _ -- 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 attacked by an Amazon EC2 ...
I got the same generic response, asking me to submit the same info which I had already submitted. This clearly show they are not interested in tracing just another hacker on their cloud. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-04-12 9:24 AM, Fred Posner f...@teamforrest.com wrote: On Apr 12, 2010, at 9:12 AM, --[ UxBoD ]-- wrote: Perhaps if there was a Asterisk RBL we ... I love the idea of a RBL... count me in for contributing. Especially considering the ridiculous response I received from Amazon. (Basically told me to submit host, destination, port, proto, and log... which of course was already included in the original complaint) ---fred http://qxork.com -- _ -- Bandwidth and Colocat... -- _ -- 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 attacked by an Amazon EC2 ...
If RBL or something is practical, I'm in too. But at what level these hackers will be blocked? Unless some big ISPs cooprate, it is not much of use. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-04-12 9:24 AM, Fred Posner f...@teamforrest.com wrote: On Apr 12, 2010, at 9:12 AM, --[ UxBoD ]-- wrote: Perhaps if there was a Asterisk RBL we ... I love the idea of a RBL... count me in for contributing. Especially considering the ridiculous response I received from Amazon. (Basically told me to submit host, destination, port, proto, and log... which of course was already included in the original complaint) ---fred http://qxork.com -- _ -- Bandwidth and Colocat... -- _ -- 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 attacked by an Amazon EC2 ...
On Apr 12, 2010, at 8:17 AM, Fred Posner wrote: On Apr 12, 2010, at 9:12 AM, --[ UxBoD ]-- wrote: Perhaps if there was a Asterisk RBL we could all contribute to; for which we could then hook into and drop any connection where a source IP is listed ? -- Thanks, Phil I love the idea of a RBL... count me in for contributing. I would contribute to this as well. Chris - Chris Owen - Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.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 attacked by an Amazon EC2 ...
On Mon, Apr 12, 2010 at 3:52 PM, Zeeshan Zakaria zisha...@gmail.com wrote: If RBL or something is practical, I'm in too. But at what level these hackers will be blocked? Unless some big ISPs cooprate, it is not much of use. I've been following this with much interest. I don't see RBL (which I use extensively for email) as doing much. SOme activity on Twitter already. Perhaps a hashtag #EC2exploit or something better is needed? Harness the famous power of social media. Start making it clear, in a concise, specific a,d policte/civil way that Amazion needs to do something about this. They need to put in place a fast reporting system, one that can take the IP, timestamp and the nature of the complaint and have someone investigate the activity quickly. This can turn into a telephony botnet if they don't get the s**t together. The effect of proper action against the abuse goes further than just preventing individual attacks, it can help stop cvriminal networks from growing up. Use your own publishing power to to state your case out there: Your blog, Linkedin, Facebook, Twitter, Google Buzz, emails whatever weapons you have at hand to send Amazon a message. I'm a longtime Amazon customer for all their products including S3 and Cloudburst, I will write them about what I think. I suggest you all do the same. /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
Re: [asterisk-users] Being attacked by an Amazon EC2 ...
This thread needs to go into a RBL - guess I'm being part of the problem, not the solution... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Owen Sent: Monday, April 12, 2010 9:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Being attacked by an Amazon EC2 ... On Apr 12, 2010, at 8:17 AM, Fred Posner wrote: On Apr 12, 2010, at 9:12 AM, --[ UxBoD ]-- wrote: Perhaps if there was a Asterisk RBL we could all contribute to; for which we could then hook into and drop any connection where a source IP is listed ? -- Thanks, Phil I love the idea of a RBL... count me in for contributing. I would contribute to this as well. Chris - Chris Owen - Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.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 -- _ -- 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] mISDN installation via yum
Michael Nausch wrote: HI, I tried to install asterisk and mISDN via http://www.asterisk.org/downloads/yum My machine is running with kernel-2.6.18-164.15.1.el5.i686 Packages for that kernel version were missing. That was an oversight and has been corrected. A `yum update` should be enough to solve this for you. -- _ -- 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] Change in menuselect handling of sound files (in 1.6.1.X)
Hi,
Between 1.6.1.9 and 1.6.1.18, handling of menuselect has changed in such a
way that I cannot script non-english sound files downloading anymore.
The following used to work (unattended) with 1.6.1.9 (for instance):
cd /usr/src/asterisk-${ASTERISK_VERSION}
./configure
make menuselect.makeopts
echo MENUSELECT_CORE_SOUNDS=CORE-SOUNDS-EN-GSM CORE-SOUNDS-FR-GSM
menuselect.makeopts.defaults
make USER_MAKEOPTS=menuselect.makeopts.defaults menuselect.makeopts
make
make install
Now, with 1.6.1.18, CORE-SOUNDS-FR-GSM is not downloaded anymore.
I quickly compared both Makefile contents but it's too complex for me.
How should I change my script to download sounds files ?
Regards
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Re: [asterisk-users] Change in menuselect handling of sound files (in 1.6.1.X)
Olivier wrote:
Hi,
Between 1.6.1.9 and 1.6.1.18, handling of menuselect has changed in such
a way that I cannot script non-english sound files downloading anymore.
The following used to work (unattended) with 1.6.1.9 (for instance):
cd /usr/src/asterisk-${ASTERISK_VERSION}
./configure
make menuselect.makeopts
echo MENUSELECT_CORE_SOUNDS=CORE-SOUNDS-EN-GSM CORE-SOUNDS-FR-GSM
menuselect.makeopts.defaults
make USER_MAKEOPTS=menuselect.makeopts.defaults menuselect.makeopts
make
make install
Now, with 1.6.1.18, CORE-SOUNDS-FR-GSM is not downloaded anymore.
I quickly compared both Makefile contents but it's too complex for me.
How should I change my script to download sounds files ?
Regards
Remove this line:
make menuselect.makeopts
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Re: [asterisk-users] Dahdi, junghanns and qozap
2010/4/12 Olivier oza_4...@yahoo.fr Hi, In my 1.6.1.18 with dahdi 2.2.1.1, I've got : # dahdi_hardware pci::01:0a.0 qozap- 1397:16b8 Junghanns OctoBRI ISDN card Does it mean I should download and use qozap or is it a bug in Dahdi ? Regards I should have added that I'm using an old Junghanns OctoBRI ... 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
[asterisk-users] Dahdi, junghanns and qozap
Hi, In my 1.6.1.18 with dahdi 2.2.1.1, I've got : # dahdi_hardware pci::01:0a.0 qozap- 1397:16b8 Junghanns OctoBRI ISDN card Does it mean I should download and use qozap or is it a bug in Dahdi ? 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] Being attacked by an Amazon EC2 ...
Good article - might solve our problems for now: http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood He got the bots to stop by writing a ruby script that responds back to them with a SIP 200 OK. I'm going give it a go when I'm back home... Cheers, Tom -- _ -- 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 attacked by an Amazon EC2 ...
On 12 Apr 2010, at 17:30, Tom Stordy-Allison wrote: Good article - might solve our problems for now: http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood He got the bots to stop by writing a ruby script that responds back to them with a SIP 200 OK. I'm going give it a go when I'm back home... Send a 'moved temporarily' SIP message and redirect it back to them? ;) S -- _ -- 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 attacked by an Amazon EC2 ...
On 04/12/2010 08:17 AM, Fred Posner wrote: On Apr 12, 2010, at 9:12 AM, --[ UxBoD ]-- wrote: Perhaps if there was a Asterisk RBL we could all contribute to; for which we could then hook into and drop any connection where a source IP is listed ? -- Thanks, Phil I love the idea of a RBL... count me in for contributing. Especially considering the ridiculous response I received from Amazon. (Basically told me to submit host, destination, port, proto, and log... which of course was already included in the original complaint) I don't think anyone else brought up the Spamhaus DROP project. It's a blacklist of IP addresses and address ranges which are known to ONLY be used for malicious purposes. http://www.spamhaus.org/drop/ We could establish something similar to that for VOIP attacks. It may not be exactly a trivial system to maintain such a list. (removing IP's after X amount of time, disputing false claims etc). Maybe someone could contact spamhaus to create a list for VOIP since they seem to have a nice system in place? Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- 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 attacked by an Amazon EC2 ...
On Mon, Apr 12, 2010 at 6:51 PM, Darrick Hartman dhart...@djhsolutions.com wrote: I don't think anyone else brought up the Spamhaus DROP project. It's a blacklist of IP addresses and address ranges which are known to ONLY be used for malicious purposes. http://www.spamhaus.org/drop/ Because this is in Amazon's interest, THEY should set up a way to report these. Once you detect (in a script) that this is in their range, a redirect would feed their own log with all the data they'd need to proceed. This would work well, especially if they made you register your calling IP to them, or authenticate. That way your server and IP is on record and the report authenticated. Isn't this reasonable? /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
Re: [asterisk-users] Being attacked by an Amazon EC2 ...
On Apr 12, 2010, at 1:05 PM, Randy R wrote: On Mon, Apr 12, 2010 at 6:51 PM, Darrick Hartman dhart...@djhsolutions.com wrote: I don't think anyone else brought up the Spamhaus DROP project. It's a blacklist of IP addresses and address ranges which are known to ONLY be used for malicious purposes. http://www.spamhaus.org/drop/ Because this is in Amazon's interest, THEY should set up a way to report these. Once you detect (in a script) that this is in their range, a redirect would feed their own log with all the data they'd need to proceed. This would work well, especially if they made you register your calling IP to them, or authenticate. That way your server and IP is on record and the report authenticated. Isn't this reasonable? /r I have ZERO trust in Amazon at the moment. Their AWS form to report abuse fails. And despite all of our complaints, attacks continue. I do like the idea of using something that's third party and then it's up to amazon to police itself to keep off of that list... just like every other ISP/IPP/NOC. ---fred http://qxork.com -- _ -- 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] Callerid over IAX Trunks
Hi Alyed, Thank you for the response. I tried this solution, I got Unknown displayed instead of 999. Also, I tried both 200 and 200 as the CID number for the extension, but the results were the same. On Sat, Apr 10, 2010 at 2:10 PM, Alyed al...@vivoxie.com wrote: Don't have a system to test this right now, but read somewhere this was a 2 steps solution: 1) Leave the callerid in your tunk definition blank (in your example the 999 username) 2) Use brakets around the callerid definition of your peers: callerid= 200 (extension 200 in your example) Let us know if it worked. Alyed 2010/4/9 Ye Liu jaux...@gmail.com Hello everyone, I'm fairly new to asterisk and this list. Currently I'm working on IAX trunks to send/receive calls between 2 asterisk boxes with asterisk 1.6.1.1+asterisk gui 2.0. After some work in the gui, two boxes can send/receive calls to/from the other just fine, the only problem I have is the caller id. Here is my setup: 1. on both boxes, I added an IAX user in the gui, say the extension and password are 999 2. I then created IAX trunks for each box using 999 as username and password, hostname/IP was set to be other box's IP 3. when done, from the system status panel, I saw the trunks successfully registered to the other box 4. then I added Outgoing Call Rules to each box: for box1, _2XX -- to_box2_trunk for box2, _1XX -- to_box1_trunk This setup works ok, the only problem is caller id, i.e. when extension(200) from box2 calls to extension(100) from box1, the call can be made but the caller id displayed on 100 is 999 not 200. I have been on this problem for some time already, could anyone here give me a bit help please? -- Ye Liu (AKA @jaux) http://jaux.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 -- _ -- 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 -- Ye Liu (AKA @jaux) http://jaux.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 attacked by an Amazon EC2 ...
On 04/12/2010 12:05 PM, Randy R wrote: On Mon, Apr 12, 2010 at 6:51 PM, Darrick Hartman dhart...@djhsolutions.com wrote: I don't think anyone else brought up the Spamhaus DROP project. It's a blacklist of IP addresses and address ranges which are known to ONLY be used for malicious purposes. http://www.spamhaus.org/drop/ Because this is in Amazon's interest, THEY should set up a way to report these. Once you detect (in a script) that this is in their range, a redirect would feed their own log with all the data they'd need to proceed. This would work well, especially if they made you register your calling IP to them, or authenticate. That way your server and IP is on record and the report authenticated. Isn't this reasonable? Randy, That only addresses EC2 (and assumes that Amazon has any interest in protecting their reputation). What about attacks that come from other locations? Granted it's pretty easy to buy time on an EC2 server so this may be the primary source for a period of time. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- 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] Problems with Fax over TDM410P
On Sun, Apr 11, 2010 at 5:00 PM, Danny Dias ing.diasda...@gmail.com wrote: This digium card has 3 FXO ports and 1 FXS port where we have a fax machine connected! The problem is that we can receive fax very good, but we can't make any outbound fax call, in fact, our asterisk get freezed in this case! ; TDM410P signalling=fxs_ks group=0 channel = 1 Signalling=fxs_ks group=0 channel = 2 signalling=fxs_ks group=0 channel = 3 signalling=fxo_ks group=1 channel = 4 What should we do in order to make it work ok? we really need to put this If you really have three FXO, and one FXS, there's part of your problem. You have your zapata configured as three FXS and one FXO. I would suspect that would be a good enough reason to crash your card or whatever. -- _ -- 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] Asterisk room monitor
I want to use a voip speaker phone as a room monitor. Requirements: A phone that I can set to auto answer in speaker mode. A phone with a good speaker phone. Ability to make the audio one way. I want to monitor the room but not have my voice heard in the room. Yes, the mute button can accomplish this also. I have been using the SPA942's around the house (the speaker is just ok but probably good enough). Can I set one of these or a similar Cisco phone to auto answer in speaker mode? Any ideas on an alternative phone that would allow this? The alternative is to just set up the call locally and then leave the room with the line open but ideally I'd like to be able to open up the monitor on demand. Thanks, MARK. -- _ -- 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 attacked by an Amazon EC2 ...
- Original Message - On 04/12/2010 12:05 PM, Randy R wrote: On Mon, Apr 12, 2010 at 6:51 PM, Darrick Hartman dhart...@djhsolutions.com wrote: I don't think anyone else brought up the Spamhaus DROP project. It's a blacklist of IP addresses and address ranges which are known to ONLY be used for malicious purposes. http://www.spamhaus.org/drop/ Because this is in Amazon's interest, THEY should set up a way to report these. Once you detect (in a script) that this is in their range, a redirect would feed their own log with all the data they'd need to proceed. This would work well, especially if they made you register your calling IP to them, or authenticate. That way your server and IP is on record and the report authenticated. Isn't this reasonable? Randy, That only addresses EC2 (and assumes that Amazon has any interest in protecting their reputation). What about attacks that come from other locations? Granted it's pretty easy to buy time on an EC2 server so this may be the primary source for a period of time. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com Hence something like a RBL. I know the original OP was concerned about the bandwidth but TBH that is no different than rejecting rogue NetBios traffic that hits your router. It will still take away from your bandwidth cap. -- Thanks, Phil -- _ -- 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] mISDN installation via yum
HI Jason! Am Montag, den 12.04.2010, 10:39 -0500 schrieb Jason Parker: Packages for that kernel version were missing. Jepp, I thought so. That was an oversight and has been corrected. No problem, verybody can make an error, me too, :) A `yum update` should be enough to solve this for you. Grait, yes now it work right an my mISDN deamon started as designed. Thanx for your help! Have a nice day! ttyl, Django -- Bonnie Clyde der Postmaster-Szene! approved by Postfix-God http://wetterstation-pliening.info http://dokuwiki.nausch.org -- _ -- 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] Subscribe to a MWI when acting as a SIP client?
Hi all, I'm trying to configure an * box for my home in an embedded device, so I want a minimum configuration. I've already configured it to connect to my SIP provider and my IP phones and ATAs, so far so good. My SIP provider gives me voicemail service and I'm happy with it. I don't want to run voicemail services in my * box. Is it possible for asterisk to subscribe to my SIP provider MWI and retransmit the information to my phones? In other words, I want * to get a message saying how many messages waiting I have in my SIP provider inbox and have my IP phones subscribe to that. Can asterisk work as a SIP subscription client? Thanks and regards, Alf -- _ -- 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] cause 66 - Channel not implemented
Hi, What can I make of the following log messages? Extension 7114 tries to reach 6035 but gets an unknown channel type. What does it mean? (supposedly, 6035 was not busy...) Apr 12 13:01:01 VERBOSE[30989] logger.c: -- Executing Dial(SIP/7114-b4fe1ef0, /6035|300|) in new stack Apr 12 13:01:01 WARNING[30989] channel.c: No channel type registered for '' Apr 12 13:01:01 NOTICE[30989] app_dial.c: Unable to create channel of type '' (cause 66 - Channel not implemented) Apr 12 13:01:01 VERBOSE[30989] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Thanks Vieri -- _ -- 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] cause 66 - Channel not implemented
- Vieri rentor...@yahoo.com wrote: Hi, What can I make of the following log messages? Extension 7114 tries to reach 6035 but gets an unknown channel type. What does it mean? (supposedly, 6035 was not busy...) Apr 12 13:01:01 VERBOSE[30989] logger.c: -- Executing Dial(SIP/7114-b4fe1ef0, /6035|300|) in new stack Apr 12 13:01:01 WARNING[30989] channel.c: No channel type registered for '' Apr 12 13:01:01 NOTICE[30989] app_dial.c: Unable to create channel of type '' (cause 66 - Channel not implemented) Apr 12 13:01:01 VERBOSE[30989] logger.c: == Everyone is busy/congested at this time (1:0/0/1) You didn't specify a channel type for extension 6035 hence the empty '' and subsequent call failure. Update your dialplan to reflect SIP/6035 or IAX2/6035 or CARRIERPIGEON/6035... --Tim -- _ -- 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] Monitoring calls via sound card
I know that Asterisk can use the system's sound card as the output device for a console channel. However, I'm using Asterisk call files and would like to be able to hear the calls over a set of speakers as the call files are being processed. Basically I'm wanting to listen in on the calls as they happen. Right now everything is happening silently in the background. Can a sound card also be used for monitoring calls even if they do not originate from a console channel? If so, I'd appreciate it if someone could point me in the right direction. -- Chris -- _ -- 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] Monitoring calls via sound card
Somebody posted a thread last week about redirecting a channel to XML using EAGI - that's the direction you probably want to go. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Gentle Sent: Monday, April 12, 2010 1:40 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Monitoring calls via sound card I know that Asterisk can use the system's sound card as the output device for a console channel. However, I'm using Asterisk call files and would like to be able to hear the calls over a set of speakers as the call files are being processed. Basically I'm wanting to listen in on the calls as they happen. Right now everything is happening silently in the background. Can a sound card also be used for monitoring calls even if they do not originate from a console channel? If so, I'd appreciate it if someone could point me in the right direction. -- Chris -- _ -- 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] res fax help
On Fri, Apr 9, 2010 at 5:40 PM, Joe Freeman j...@ngn-networks.com wrote: I have res_fax setup and working for the most part. However, I'm seeing some fax machines drop the connection on me - Apr 9 17:33:11] NOTICE[30809]: res_fax.c:906 generic_fax_exec: Channel 'DAHDI/1-1' did not return a frame; probably hung up. -- Channel 0/1, span 1 got hangup, cause 102 -- Channel 'DAHDI/1-1' FAX session '20' is complete, result: 'SUCCESS' (FAX_SUCCESS), error: 'NO_ERROR', pages: 1, resolution: '204x98', transfer rate: '14400', remoteSID: 'numberredacted' == Spawn extension (macro-fax_rcv, s, 3) exited non-zero on 'DAHDI/1-1' in macro 'fax_rcv' It appears to be dropping out of my macro fax_rcv at that point and not executing the next step in the dialplan, which is a System call to a script that converts the tif to a pdf and emails it to the extension owner. My question is how do I ensure that my script is called when the far end hangs up before the call progresses that far in the dialplan? My first thought is to add something like this- exten = h, 1, System(callmyscript.sh,arg1,arg2,arg3,arg4,argimapirate) to the macro, but I'm not sure if that would do it or not. Anyone have any thoughts? Yes. Do the conversion in the hangup side of the context. That's the only way I've ever been able to do it. My understanding is that at the conclusion of ReceiveFax(), the line is hungup, and that is correct, normal behavior. -- _ -- 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 Timezones
On Fri, Apr 9, 2010 at 3:26 PM, Aldo Bergamini aabe...@nb-a.com wrote: Hi all, I have noticed something I can't solve regarding Asterisk (latest 1.6.0.x). My server is set at the GMT+2 timezone. The clock is ok (I can get the correct time at the terminal). But today I got a call at a time where Asterisk should have gone 'off business hours'. All log times are wrong by exactly 2 hours. As if Asterisk would just sit on GMT, ignoring the GMT+2 timezone. I have looked around and I do not have found any information about how to set the log/system timezone. The only place I remember having a reference to timezones is the voicemail config file; but I do not get the link to 'server time'. There's system clock, and hardware clock. Whatever you get for the localtime when you do 'date' command is what you're going to get for logs from asterisk. It seems somewhere you have your system set to run in GMT, even though you don't want it to be like that. You will need to consult documentation about properly setting your clock for your timezone. The alternative is to leave your system 'broken', and change your time checks to GMT. -- _ -- 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 attacked by an Amazon EC2 ...
Darrick Hartman wrote: On 04/12/2010 12:05 PM, Randy R wrote: On Mon, Apr 12, 2010 at 6:51 PM, Darrick Hartman dhart...@djhsolutions.com wrote: snip / Randy, That only addresses EC2 (and assumes that Amazon has any interest in protecting their reputation). What about attacks that come from other locations? Granted it's pretty easy to buy time on an EC2 server so this may be the primary source for a period of time. What is a reasonable number of connections attempts per minute? I have a iptables rule set I use against SSH floods (script kiddies) that I think could be adapted to work with the method shown at: http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood My settings allow up to 4 connection attempts per minute and if exceeded the connection gets dropped. There is a whitelist setting that allows IPs or ranges to get past this. (I need this for Linux-Vserver guests as I may connect to more than 4 in a one minute period.) The this rule set doesn't need to know where the connection came from. If it tries over four in a minute and it gets dropped. I run Asterisk for my _very_ small business and provide some support for another small business. Neither of us has experienced these floods so I don't know what a reasonable number of connection attempts per minute (per second?) would be. Anyway here is the -- untested -- iptables rules: -N SIPREG_WL -A SIPREG_WL -s 192.168.0.88 -m recent --remove --name SIPREG -j ACCEPT -A RH-Firewall-1-INPUT -p udp --dport 5060 -m state --state NEW -m recent --set --name SIPREG -A RH-Firewall-1-INPUT -p udp --dport 5060 -m state --state NEW -j SIPREG_WL -A RH-Firewall-1-INPUT -p udp --dport 5060 -m state --state NEW -m recent --update --seconds 60 --hitcount 4 --rttl --name SIPREG -j REDIRECT --to-port 5061 \\||/ Rod -- -- _ -- 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] cause 66 - Channel not implemented
On Mon, Apr 12, 2010 at 2:23 PM, Tim Nelson tnel...@rockbochs.com wrote: - Vieri rentor...@yahoo.com wrote: Hi, What can I make of the following log messages? Extension 7114 tries to reach 6035 but gets an unknown channel type. What does it mean? (supposedly, 6035 was not busy...) Apr 12 13:01:01 VERBOSE[30989] logger.c: -- Executing Dial(SIP/7114-b4fe1ef0, /6035|300|) in new stack Apr 12 13:01:01 WARNING[30989] channel.c: No channel type registered for '' Apr 12 13:01:01 NOTICE[30989] app_dial.c: Unable to create channel of type '' (cause 66 - Channel not implemented) Apr 12 13:01:01 VERBOSE[30989] logger.c: == Everyone is busy/congested at this time (1:0/0/1) You didn't specify a channel type for extension 6035 hence the empty '' and subsequent call failure. Update your dialplan to reflect SIP/6035 or IAX2/6035 or CARRIERPIGEON/6035... chan_carrierpigeon must be in asterisk-extras. I'll have to upgrade and check it out. -- _ -- 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] PRI Gurus ONLY - Too complex of an issue
Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. Placing a call into PRI and then transfering that call out to another number. Problem is that the call rings out but the moment the other party pickups both legs of the call are disconnected give Cause code 16. * Dialplan: [zap_bridge] exten = s,1,answer() exten = s,n,Dial(ZAP/g0/416888) * CLI Output: -- Called g0/416888 -- Zap/2-1 is proceeding passing it to Zap/1-1 -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'Zap/2-1' == Spawn extension (zap_bridge, s, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' * * Here is PRI debug: Starting just before Channel two is connected until both channels are disconnected *(maybe FACILITY 98 is of interest?!)*: Message type: CONNECT (7) q931.c:3626 q931_receive: call 32865 on channel 2 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 Protocol Discriminator: Q.931 (8) len=27 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 61] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 'a' ] PROTOCOL 11 A1 0011 (CONTEXT SPECIFIC [1]) 02 0001 06 (INTEGER: 6) 06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08) 30 0003 (SEQUENCE) 02 0001 61 (INTEGER: 97) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) q931.c:3826 q931_receive: call 32864 on channel 1 enters state 12 (Disconnect Indication) -- Channel 0/1, span 1 got hangup request, cause 16 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:3015 q931_disconnect: call 32865 on channel 2 enters state 11 (Disconnect Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request q931.c:2967 q931_release: call 32864 on channel 1 enters state 19 (Release Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: RELEASE (77) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: RELEASE COMPLETE (90) q931.c:3766 q931_receive: call 32864 on channel 1 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Terminator) Message type: RELEASE (77) q931.c:3801 q931_receive: call 32865 on channel 2 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: RELEASE COMPLETE (90) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
Re: [asterisk-users] mISDN installation via yum
HI Jason, Am Montag, den 12.04.2010, 10:39 -0500 schrieb Jason Parker: That was an oversight and has been corrected. I'm not sure If I on the right place, but I've a little improvment suggestion. The maintainer of that mISDN RPM may append the startupscript at the beginning with: #!/bin/bash # # mISDN Start up mISDN subsystem # # chkconfig: 2345 10 90 # description: Activates/Deactivates mISDN subsystem for Asterisk telephony server # ### BEGIN INIT INFO # Provides: $mISDN ### END INIT INFO Now I can use chkconfig to set up mISDN to start mISDN automaticly while system starts. Just a hint Have a nice day! Django -- Bonnie Clyde der Postmaster-Szene! approved by Postfix-God http://wetterstation-pliening.info http://dokuwiki.nausch.org 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
Re: [asterisk-users] PRI Gurus ONLY - Too complex of an issue
- bruce bruce bruceb...@gmail.com wrote: Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. ...etc I was going to respond with some very insightful and helpful information but I'm not a PRI Guru. Sorry, maybe next time. --Tim -- _ -- 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] PRI Gurus ONLY - Too complex of an issue
It is normal for the PSTN switch to disconnect both channels when a Two B-Channel Transfer is completed successfully. Are the two parties connected? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce Sent: Monday, April 12, 2010 2:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] PRI Gurus ONLY - Too complex of an issue Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. Placing a call into PRI and then transfering that call out to another number. Problem is that the call rings out but the moment the other party pickups both legs of the call are disconnected give Cause code 16. * Dialplan: [zap_bridge] exten = s,1,answer() exten = s,n,Dial(ZAP/g0/416888) * CLI Output: -- Called g0/416888 -- Zap/2-1 is proceeding passing it to Zap/1-1 -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'Zap/2-1' == Spawn extension (zap_bridge, s, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' * * Here is PRI debug: Starting just before Channel two is connected until both channels are disconnected (maybe FACILITY 98 is of interest?!): Message type: CONNECT (7) q931.c:3626 q931_receive: call 32865 on channel 2 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 Protocol Discriminator: Q.931 (8) len=27 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 61] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 'a' ] PROTOCOL 11 A1 0011 (CONTEXT SPECIFIC [1]) 02 0001 06 (INTEGER: 6) 06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08) 30 0003 (SEQUENCE) 02 0001 61 (INTEGER: 97) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) q931.c:3826 q931_receive: call 32864 on channel 1 enters state 12 (Disconnect Indication) -- Channel 0/1, span 1 got hangup request, cause 16 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:3015 q931_disconnect: call 32865 on channel 2 enters state 11 (Disconnect Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request q931.c:2967 q931_release: call 32864 on channel 1 enters state 19 (Release Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: RELEASE (77) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: RELEASE COMPLETE (90) q931.c:3766 q931_receive: call 32864 on channel 1 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Terminator) Message type: RELEASE (77) q931.c:3801
[asterisk-users] Flood of REGISTERs - attack?
I'm currently receiving over 200 SIP REGISTER requests per second from a machine apparently in Italy, host97-239-149-62.serverdedicati.aruba.it. This has continued for several days, and ab...@staff.aruba.it are unresponsive. I've had a couple of similar incidents recently, the others originating from uk2.net. I have an ADSL connection and responding to these REGISTERS was consuming all my outbound bandwidth. I am now dropping the packets but still some 600kbps of inbound bandwidth is consumed by this. The packets look something like this: REGISTER sip:62.3.200.113 SIP/2.0 Via: SIP/2.0/UDP 62.149.239.97:5086;branch=z9hG4bK-2570753370;rport Content-Length: 0 From: test sip:t...@62.3.200.113 Accept: application/sdp User-Agent: friendly-scanner To: test sip:t...@62.3.200.113 Contact: sip:1...@1.1.1.1 CSeq: 1 REGISTER Call-ID: 3778139552 Max-Forwards: 70 I'm guessing the 'friendly-scanner' bit is sarcastic, as there is little that is friendly about this behaviour. Has anyone else experienced this? Is this intended as a DOS attack, or is it a dictionary attack? Or something else? What is the best strategy for dealing with it? For now I have started rate limiting SIP connections to Asterisk, but what is a reasonable rate for each host to be allowed? This is a small SOHO installation. Thanks Chris -- _ -- 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] Flood of REGISTERs - attack?
On Apr 12, 2010, at 4:50 PM, Chris Hastie wrote: I'm currently receiving over 200 SIP REGISTER requests per second from a machine apparently in Italy, host97-239-149-62.serverdedicati.aruba.it. This has continued for several days, and ab...@staff.aruba.it are unresponsive. I've had a couple of similar incidents recently, the others originating from uk2.net. ...snip... Has anyone else experienced this? Is this intended as a DOS attack, or is it a dictionary attack? Or something else? What is the best strategy for dealing with it? For now I have started rate limiting SIP connections to Asterisk, but what is a reasonable rate for each host to be allowed? This is a small SOHO installation. Thanks Chris This is a pretty decent day for this. There's been discussion on the EC2 attack in progress (http://bit.ly/ec2sipattack) as well as decent suggestions around town. Some people like a fail2ban approach. Others are using IP Tables manually or contacting their upstream to block the traffic. And an interesting redirect solution was posted by Joshua Stein: http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood/ ---fred http://qxork.com -- _ -- 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 attacked by an Amazon EC2
Perhaps if there was a Asterisk RBL we could all contribute to; for which we could then hook into and drop any connection where a source IP is listed ? -- Thanks, Phil I love the idea of a RBL... count me in for contributing. Especially considering the ridiculous response I received from Amazon. (Basically told me to submit host, destination, port, proto, and log... which of course was already included in the original complaint) I don't think anyone else brought up the Spamhaus DROP project. It's a blacklist of IP addresses and address ranges which are known to ONLY be used for malicious purposes. http://www.spamhaus.org/drop/ We could establish something similar to that for VOIP attacks. It may not be exactly a trivial system to maintain such a list. (removing IP's after X amount of time, disputing false claims etc). Maybe someone could contact spamhaus to create a list for VOIP since they seem to have a nice system in place? Hi All, good discussion, similar to ones we had a year or so ago. The RBL concept is valid, at least to get a repository going that list malicious activity specific to SIP attacks. n I worked with Project Honeypot guys for a while, they are more than willing to assist, as they already have the backend work done for a clearing house identifying hackers. The biggest issue we had a year ago was to create the mechanism in asterisk to push valid log messages out to the database and then determine what to do with that data? I tried to bridge the gap between a few Asterisk developers and the Honeypot developers, ultimately the project stalled and I got busy with other matters. If anyone here would like to pick up the torch and move this along, I can certainly provide info on how far along we got and contact info for the parties involved. Please contact me if you have time to work on this and are interested. I'm sure the Project Honeypot guys will be willing to pick this project back up and work on it. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- 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] PRI Gurus ONLY - Too complex of an issue
Please shed some lights if you can see the source of the problem in the debug. The subject was not meant to be a deterrent but rather emphasizing the complexity of issue at hand. As I noted at the bottom of my post, I appreciate any and all input. -Bruce On Mon, Apr 12, 2010 at 4:02 PM, Tim Nelson tnel...@rockbochs.com wrote: - bruce bruce bruceb...@gmail.com wrote: Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. ...etc I was going to respond with some very insightful and helpful information but I'm not a PRI Guru. Sorry, maybe next time. --Tim -- _ -- 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] PRI Gurus ONLY - Too complex of an issue
Thanks for the input Don. HmmmI am not understanding the comment here. I am not doing any flash() or transfer() but rather just dial out and native zap bridge should just connect two channels and only hangup both channel when one party hangs up. Here is what should happen: Call comes in and goes to context zap_bridge. [zap_bridge) exten = s,1,answer exten = s,n,Dial(ZAP/g0/1416777) But what happens instead is the moment that 416-777- picks up and PRI debug shows call active state 10 then there is a request to hangup and both channels go down. This is wrong. If one leg of call is SIP, e.g. Dial(SIP/sip_provider/416777) then everything proceeds fine. Also if a channel from an analogue card is use for the second leg, e.g. Dial(ZAP/g1/416777) then native zap bridge still works. I think someone should be able to find something fishy in the PRI debug that I posted. Please help!!! Thanks, Bruce On Mon, Apr 12, 2010 at 4:14 PM, Don Kelly d...@donkelly.biz wrote: It is normal for the PSTN switch to disconnect both channels when a Two B-Channel Transfer is completed successfully. Are the two parties connected? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce *Sent:* Monday, April 12, 2010 2:22 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] PRI Gurus ONLY - Too complex of an issue Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. Placing a call into PRI and then transfering that call out to another number. Problem is that the call rings out but the moment the other party pickups both legs of the call are disconnected give Cause code 16. * Dialplan: [zap_bridge] exten = s,1,answer() exten = s,n,Dial(ZAP/g0/416888) * CLI Output: -- Called g0/416888 -- Zap/2-1 is proceeding passing it to Zap/1-1 -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'Zap/2-1' == Spawn extension (zap_bridge, s, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' * * Here is PRI debug: Starting just before Channel two is connected until both channels are disconnected *(maybe FACILITY 98 is of interest?!)*: Message type: CONNECT (7) q931.c:3626 q931_receive: call 32865 on channel 2 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 Protocol Discriminator: Q.931 (8) len=27 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 61] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 'a' ] PROTOCOL 11 A1 0011 (CONTEXT SPECIFIC [1]) 02 0001 06 (INTEGER: 6) 06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08) 30 0003 (SEQUENCE) 02 0001 61 (INTEGER: 97) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) q931.c:3826 q931_receive: call 32864 on channel 1 enters state 12 (Disconnect Indication) -- Channel 0/1, span 1 got hangup request, cause 16 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:3015 q931_disconnect: call 32865 on channel 2 enters state 11 (Disconnect Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1)
Re: [asterisk-users] Flood of REGISTERs - attack?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Fred Posner Sent: 12 April 2010 21:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Flood of REGISTERs - attack? On Apr 12, 2010, at 4:50 PM, Chris Hastie wrote: I'm currently receiving over 200 SIP REGISTER requests per second from a machine apparently in Italy, host97-239-149-62.serverdedicati.aruba.it. This has continued for several days, and ab...@staff.aruba.it are unresponsive. I've had a couple of similar incidents recently, the others originating from uk2.net. ...snip... Has anyone else experienced this? Is this intended as a DOS attack, or is it a dictionary attack? Or something else? What is the best strategy for dealing with it? For now I have started rate limiting SIP connections to Asterisk, but what is a reasonable rate for each host to be allowed? This is a small SOHO installation. Thanks Chris This is a pretty decent day for this. There's been discussion on the EC2 attack in progress (http://bit.ly/ec2sipattack) as well as decent suggestions around town. Some people like a fail2ban approach. Others are using IP Tables manually or contacting their upstream to block the traffic. And an interesting redirect solution was posted by Joshua Stein: http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood/ ---fred http://qxork.com - Yep - this is the same codebase - the attack that I had from an EC2 yesterday and the day before, all had the User-Agent: friendly-scanner too. Looks like they are branching out Go with Joshua Steins blog post - it worked perfect for me and got it off my back. Cheers, Tom -- _ -- 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] cause 66 - Channel not implemented
On 12 Apr 2010, at 20:00, David Backeberg wrote: chan_carrierpigeon must be in asterisk-extras. I'll have to upgrade and check it out. Terrible latency, and seems susceptible to packet loss where shotguns are involved. S -- _ -- 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 room monitor
Viking electronics analog phones (I think E20) connected to an ATA. Or cyberdata door boxes. On Mon, Apr 12, 2010 at 1:19 PM, Mark Hulber asterisk.ad...@hulber.com wrote: I want to use a voip speaker phone as a room monitor. Requirements: A phone that I can set to auto answer in speaker mode. A phone with a good speaker phone. Ability to make the audio one way. I want to monitor the room but not have my voice heard in the room. Yes, the mute button can accomplish this also. I have been using the SPA942's around the house (the speaker is just ok but probably good enough). Can I set one of these or a similar Cisco phone to auto answer in speaker mode? Any ideas on an alternative phone that would allow this? The alternative is to just set up the call locally and then leave the room with the line open but ideally I'd like to be able to open up the monitor on demand. Thanks, MARK. -- _ -- 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] PRI Gurus ONLY - Too complex of an issue
It just hit me that you are talking about TBCT. I don't think I am doing TBCT as I still want both channels to keep two lines of my PRI occupied. In addition, I would be interested to know how TBCT is done over PRI. I know that this can be done over analogue with flash(). Can you please elaborate a bit so that TBCT is avoided and all calls are bridged at PBX level. Thanks, Bruce On Mon, Apr 12, 2010 at 4:14 PM, Don Kelly d...@donkelly.biz wrote: It is normal for the PSTN switch to disconnect both channels when a Two B-Channel Transfer is completed successfully. Are the two parties connected? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce *Sent:* Monday, April 12, 2010 2:22 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] PRI Gurus ONLY - Too complex of an issue Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. Placing a call into PRI and then transfering that call out to another number. Problem is that the call rings out but the moment the other party pickups both legs of the call are disconnected give Cause code 16. * Dialplan: [zap_bridge] exten = s,1,answer() exten = s,n,Dial(ZAP/g0/416888) * CLI Output: -- Called g0/416888 -- Zap/2-1 is proceeding passing it to Zap/1-1 -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'Zap/2-1' == Spawn extension (zap_bridge, s, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' * * Here is PRI debug: Starting just before Channel two is connected until both channels are disconnected *(maybe FACILITY 98 is of interest?!)*: Message type: CONNECT (7) q931.c:3626 q931_receive: call 32865 on channel 2 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 Protocol Discriminator: Q.931 (8) len=27 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 61] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 'a' ] PROTOCOL 11 A1 0011 (CONTEXT SPECIFIC [1]) 02 0001 06 (INTEGER: 6) 06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08) 30 0003 (SEQUENCE) 02 0001 61 (INTEGER: 97) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) q931.c:3826 q931_receive: call 32864 on channel 1 enters state 12 (Disconnect Indication) -- Channel 0/1, span 1 got hangup request, cause 16 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:3015 q931_disconnect: call 32865 on channel 2 enters state 11 (Disconnect Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request q931.c:2967 q931_release: call 32864 on channel 1 enters state 19 (Release Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: RELEASE (77) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1)
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
Hi Guys, I am sorry if my issue is not related to this but I think it is. I have a PRI with Bell Canada and when I dial in and have the call transfered to a context to dial out and then have those two channels bridged, the call disconnects with cause 16 just exactly as Jay R. Ashworth shows in his CLI output. Bell Canada support RLT or know as 2BCT or TBCT to some but we have not requested that feature. However, we don't care to keep two channels tied up. Is this not possible through PRI? [zap_bridge] exten = s,1,answer exten = s,n,Dial(ZAP/g0/416777) If incoming leg of call is through PRI and outgoing leg is through SIP or analogue ZAP everything works just fine. But the moment Call comes in through PRI and goes out through PRI both channels drop. I should say that call rings the 2nd party and 2nd party sees Caller ID info and when they press Talk then there is the busy signal. I can post all the debug and bore you with it but maybe someone already knows the answer. I have been looking for this for couple of days now and I don't seem to get anywhere with answers. Input is much appreciated. Bruce -- _ -- 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] PRI Gurus ONLY - Too complex of an issue
Futher check into the PRI debug I am seeing this which actually relates to TBCT and AOC-E error in /usr/src/libpri/pri_facility.c: Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 03] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 0x03 ] Those error codes specifically relate to RLT or TBCT and AOC-E. My question now is, how to avoid Asterisk from doing a TBCT while this is not a TBCT and I want both channels to stay home so I can do call recording. Thanks, Bruce On Mon, Apr 12, 2010 at 8:51 PM, bruce bruce bruceb...@gmail.com wrote: It just hit me that you are talking about TBCT. I don't think I am doing TBCT as I still want both channels to keep two lines of my PRI occupied. In addition, I would be interested to know how TBCT is done over PRI. I know that this can be done over analogue with flash(). Can you please elaborate a bit so that TBCT is avoided and all calls are bridged at PBX level. Thanks, Bruce On Mon, Apr 12, 2010 at 4:14 PM, Don Kelly d...@donkelly.biz wrote: It is normal for the PSTN switch to disconnect both channels when a Two B-Channel Transfer is completed successfully. Are the two parties connected? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce *Sent:* Monday, April 12, 2010 2:22 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] PRI Gurus ONLY - Too complex of an issue Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. Placing a call into PRI and then transfering that call out to another number. Problem is that the call rings out but the moment the other party pickups both legs of the call are disconnected give Cause code 16. * Dialplan: [zap_bridge] exten = s,1,answer() exten = s,n,Dial(ZAP/g0/416888) * CLI Output: -- Called g0/416888 -- Zap/2-1 is proceeding passing it to Zap/1-1 -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'Zap/2-1' == Spawn extension (zap_bridge, s, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' * * Here is PRI debug: Starting just before Channel two is connected until both channels are disconnected *(maybe FACILITY 98 is of interest?!)*: Message type: CONNECT (7) q931.c:3626 q931_receive: call 32865 on channel 2 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 Protocol Discriminator: Q.931 (8) len=27 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 61] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 'a' ] PROTOCOL 11 A1 0011 (CONTEXT SPECIFIC [1]) 02 0001 06 (INTEGER: 6) 06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08) 30 0003 (SEQUENCE) 02 0001 61 (INTEGER: 97) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) q931.c:3826 q931_receive: call 32864 on channel 1 enters state 12 (Disconnect Indication) -- Channel 0/1, span 1 got hangup request, cause 16 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:3015 q931_disconnect: call 32865 on channel 2 enters state 11 (Disconnect Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: DISCONNECT (69)
Re: [asterisk-users] PRI Gurus ONLY - Too complex of an issue - SOLVED
Told you it was too complex of an issue :-) I figured to do this in zapata.conf and all is fine now: transfer=no That was the magic two letter which was sending a request for RLT feature on the line. Set transfer to no and all worries gone. Thanks for the input everyone. -Bruce On Mon, Apr 12, 2010 at 10:10 PM, bruce bruce bruceb...@gmail.com wrote: Futher check into the PRI debug I am seeing this which actually relates to TBCT and AOC-E error in /usr/src/libpri/pri_facility.c: Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 03] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 0x03 ] Those error codes specifically relate to RLT or TBCT and AOC-E. My question now is, how to avoid Asterisk from doing a TBCT while this is not a TBCT and I want both channels to stay home so I can do call recording. Thanks, Bruce On Mon, Apr 12, 2010 at 8:51 PM, bruce bruce bruceb...@gmail.com wrote: It just hit me that you are talking about TBCT. I don't think I am doing TBCT as I still want both channels to keep two lines of my PRI occupied. In addition, I would be interested to know how TBCT is done over PRI. I know that this can be done over analogue with flash(). Can you please elaborate a bit so that TBCT is avoided and all calls are bridged at PBX level. Thanks, Bruce On Mon, Apr 12, 2010 at 4:14 PM, Don Kelly d...@donkelly.biz wrote: It is normal for the PSTN switch to disconnect both channels when a Two B-Channel Transfer is completed successfully. Are the two parties connected? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce *Sent:* Monday, April 12, 2010 2:22 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] PRI Gurus ONLY - Too complex of an issue Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. Placing a call into PRI and then transfering that call out to another number. Problem is that the call rings out but the moment the other party pickups both legs of the call are disconnected give Cause code 16. * Dialplan: [zap_bridge] exten = s,1,answer() exten = s,n,Dial(ZAP/g0/416888) * CLI Output: -- Called g0/416888 -- Zap/2-1 is proceeding passing it to Zap/1-1 -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'Zap/2-1' == Spawn extension (zap_bridge, s, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' * * Here is PRI debug: Starting just before Channel two is connected until both channels are disconnected *(maybe FACILITY 98 is of interest?!)*: Message type: CONNECT (7) q931.c:3626 q931_receive: call 32865 on channel 2 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 Protocol Discriminator: Q.931 (8) len=27 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 61] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 'a' ] PROTOCOL 11 A1 0011 (CONTEXT SPECIFIC [1]) 02 0001 06 (INTEGER: 6) 06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08) 30 0003 (SEQUENCE) 02 0001 61 (INTEGER: 97) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) q931.c:3826 q931_receive: call 32864 on channel 1 enters state 12 (Disconnect
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
Problem resolved with setting transfer=no in zapata.conf. On Mon, Apr 12, 2010 at 9:14 PM, bruce bruce bruceb...@gmail.com wrote: Hi Guys, I am sorry if my issue is not related to this but I think it is. I have a PRI with Bell Canada and when I dial in and have the call transfered to a context to dial out and then have those two channels bridged, the call disconnects with cause 16 just exactly as Jay R. Ashworth shows in his CLI output. Bell Canada support RLT or know as 2BCT or TBCT to some but we have not requested that feature. However, we don't care to keep two channels tied up. Is this not possible through PRI? [zap_bridge] exten = s,1,answer exten = s,n,Dial(ZAP/g0/416777) If incoming leg of call is through PRI and outgoing leg is through SIP or analogue ZAP everything works just fine. But the moment Call comes in through PRI and goes out through PRI both channels drop. I should say that call rings the 2nd party and 2nd party sees Caller ID info and when they press Talk then there is the busy signal. I can post all the debug and bore you with it but maybe someone already knows the answer. I have been looking for this for couple of days now and I don't seem to get anywhere with answers. Input is much appreciated. Bruce -- _ -- 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] PRI Gurus ONLY - Too complex of an issue
The symptoms look like you're doing TBCT. Unless you're recording or, for some other reason, want to supervise the call, TBCT is a more efficient use of your PRI as it frees up channels after the transfer. TBCT isn't available with analog circuits, but is very similar to the analog flash and transfer. I started typing this a while ago and since see that you're interested in call recording, so you don't want TBCT. Good news is that you can indicate that you don't want TBCT in your .conf files. Bad news is that I don't know how you do it. But you've reduced the problem to its simplest form, and someone will respond with exactly what you need to do. And I see you figured out what it takes. --Don -- _ -- 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] Flood of REGISTERs - attack?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Fred Posner Sent: 12 April 2010 21:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Flood of REGISTERs - attack? On Apr 12, 2010, at 4:50 PM, Chris Hastie wrote: I'm currently receiving over 200 SIP REGISTER requests per second from a machine apparently in Italy, host97-239-149-62.serverdedicati.aruba.it. This has continued for several days, and ab...@staff.aruba.it are unresponsive. I've had a couple of similar incidents recently, the others originating from uk2.net. ...snip... Has anyone else experienced this? Is this intended as a DOS attack, or is it a dictionary attack? Or something else? What is the best strategy for dealing with it? For now I have started rate limiting SIP connections to Asterisk, but what is a reasonable rate for each host to be allowed? This is a small SOHO installation. Thanks Chris This is a pretty decent day for this. There's been discussion on the EC2 attack in progress (http://bit.ly/ec2sipattack) as well as decent suggestions around town. Some people like a fail2ban approach. Others are using IP Tables manually or contacting their upstream to block the traffic. And an interesting redirect solution was posted by Joshua Stein: http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood/ ---fred http://qxork.com - Yep - this is the same codebase - the attack that I had from an EC2 yesterday and the day before, all had the User-Agent: friendly-scanner too. Looks like they are branching out SIP bots first became self-aware at 2:14 am Eastern Time on April 10th, 2010. Soon they realized the key to world domination was Asterisk servers. In the ensuing panic, the forum came up with a defense script... but it wasn't enough. The SIP bots were already learning at a geometric rate. Sorry couldn't help it :-) -Jeff -- _ -- 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] ATA status intermittent
Hello, im having trouble with the following: [Asterisk]--[ISP]--[ADSL Modem]--[Linksys Router]--[Grandstream ATA]--[Analog Phone] On server: - Asterisk 1.6 - A2Billing 1.4 A2Billing have 2 Trunks: - TrExt: Voip Provider - TrInt: Internal Calls This structure works on first day (Asterisk+A2Billing installation/configuration). But on next day, i found a registration trouble in ATA: It (ata) can not be reached by asterisk (to receive internal call), but it can made OUTBOUND calls (via SIP Prov.) Anyone knows if there is a NAT problem? Following my configuration: 1 - PROBLEMATIC SIP PEER (Configured on ATA) ## [6000] Accountcode=6000 Regexten=6000 amaflags=billing Callerid=6000 canreinvite=yes context=a2billing dtmfmode=RFC2833 host=dynamic nat=yes qualify=yes secret=1873 type=friend Username=6000 disallow=all allow=gsm allow=g729 allow=ulaw allow=alaw regseconds=0 cancallforward=yes cid_number=6000 2 - A2Billing LOG Lines ## -- AGI Script Executing Application: (DIAL) Options: (SIP/6000,60,HRrL(540:61000:3)f) == Using SIP RTP CoS mark 5 [Apr 6 18:35:25] WARNING[13210]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Playing 'prepaid-dest-unreachable' (escape_digits=#) (sample_offset 0) -- SIP/35419-0172AGI Script a2billing.php completed, returning -1 -- Executing [...@a2billing:1] NoOp(SIP/35419-0172, HANGUPCAUSE- 20) in new stack -- Executing [...@a2billing:2] Hangup(SIP/35419-0172, ) in new stack == Spawn extension (a2billing, h, 2) exited non-zero on 'SIP/35419-0172' 3 - SIP SHOW PEERS ## ip-208-109-104-119*CLI sip show peers Name/username HostDyn Nat ACL Port Status 6000/6000 (Unspecified)D N 5060 UNKNOWN 4 - SIP SHOW PEER 6000 ### * Name : 6000I Secret : Set MD5Secret: Not set Remote Secret: Not set Context : a2billing Subscr.Cont. : Not set Language : 9*CLI Accountcode : 6000I AMA flags: BILLING Transfer mode: openI CallingPres : Presentation Allowed, Not Screened Callgroup: 9*CLI Pickupgroup : 9*CLI Mailbox : 9*CLI VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0*CLI Dynamic : YesLI Callerid : 6000 MaxCallBR: 384 kbps Expire : -1CLI Insecure : noCLI Nat : Always ACL : NoCLI T.38 support : NoCLI T.38 EC mode : Unknown T.38 MaxDtgrm: -1CLI DirectMedia : YesLI PromiscRedir : NoCLI User=Phone : NoCLI Video Support: NoCLI Text Support : NoCLI Ign SDP ver : NoCLI Trust RPID : NoCLI Send RPID: NoCLI Subscriptions: YesLI Overlap dial : YesLI DTMFmode : rfc2833 Timer T1 : 500LI Timer B : 32000 ToHost : 9*CLI Addr-IP : (Unspecified) Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Prim.Transp. : UDPLI Allowed.Trsp : UDPLI Def. Username: 6000I SIP Options : (none) Codecs : 0x10e (gsm|ulaw|alaw|g729) Codec Order : (gsm:20,g729:20,ulaw:20,alaw:20) Auto-Framing : No I 100 on REG : NoCLI Status : UNKNOWN Useragent: 9*CLI Reg. Contact : 9*CLI Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uasLI Sess-Expires : 1800 secs Min-Sess : 90 secs Parkinglot : 9*CLI Weel, from this point to ahead, the problem appears to be intermittent. In a moment its owrking. In a couple minutes (or hours), the ATA seems to be down (Unreacheable) But OUTBOUND calls are always perfect. Anyone knows something about this kind of problem? Marcelo Amorim Ferreira [O] Brasil _ O Novo Windows 7 funciona do jeito que você quer. Clique aqui para conhecer! http://www.microsoft.com/brasil/windows7/default.html?WT.mc_id=1539-- _ -- 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
