[asterisk-users] Read Telnet Packet
Dear All, I want to read telnet packet continuously whenever a new call is originated and store into a variable after that pass into window server. I have written a Perl script to read telnet packet but problem is that whenever I executed Perl script then got a telnet packet( mean Only when i execute Perl script) here I want to put scheduler,event or other technique whenever a new call will come Perl script automatically run. Regards Akhilesh -- _ -- 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] Read Telnet Packet
On Fri, Oct 11, 2013 at 11:34:48AM +0530, akhilesh chand wrote: Dear All, I want to read telnet packet continuously whenever a new call is originated and store into a variable after that pass into window server. I have written a Perl script to read telnet packet but problem is that whenever I executed Perl script then got a telnet packet( mean Only when i execute Perl script) here I want to put scheduler,event or other technique whenever a new call will come Perl script automatically run. You can use a packet sniffer such as tcpdump or wireshark instead. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.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] Read Telnet Packet
On Friday 11 October 2013, akhilesh chand wrote: Dear All, I want to read telnet packet continuously whenever a new call is originated and store into a variable after that pass into window server. I have written a Perl script to read telnet packet but problem is that whenever I executed Perl script then got a telnet packet( mean Only when i execute Perl script) here I want to put scheduler,event or other technique whenever a new call will come Perl script automatically run. This is really a Perl question, not an Asterisk question, and you might have more joy asking on a Perl list or forum. Anyway, there are two ways to accomplish what you want. The easy way is to have inetd start your Perl script whenever a packet is sent to a particular port; when the script starts, its STDIN and STDOUT will already be connected to the port. The other way is to have your Perl script run as a daemon; then fork off a clone of itself to deal with requests as they come in. After a fork, the child process will inherit a copy of the socket object; and this will persist even after you undef it in the parent process. If you want to invoke a Perl script from within the dialplan, you can either use a full-on AGI script (which will even allow you to pass variables back and forth and do other funky stuff) or use System() for a quick and dirty call to a script that doesn't need to send anything back. -- AJS Answers come *after* questions. -- _ -- 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] GSM to SIP Adapter
Greetings,I'm looking for a really cheap GSM-SIP gateway, Single channel (one SIM card). any suggestions? Tarek Sawah -- _ -- 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] GSM to SIP Adapter
On Friday 11 October 2013, Tarek Sawah wrote: Greetings,I'm looking for a really cheap GSM-SIP gateway, Single channel (one SIM card). any suggestions? Tarek Sawah We've been using OpenVox G400P cards (PCI; there is also a G400E, which is PCI express for newer motherboards). Sends and receives text messages, and makes and answers phone calls. Accepts up to four RF modules, each of which accepts one SIM card. If you only need text message functionality (not voice calls), then almost any old mobile phone with a USB or RS232 cable can be used as a GSM modem -- and you probably have one lying in a drawer. -- AJS Answers come *after* questions. -- _ -- 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] GSM to SIP Adapter
Thank you for the reply, actually we are looking for something like the followinghttp://www.ebay.com/itm/GSM1SIP-GSM-over-IP-GoIP-SIP-Quad-Bands-voip-gateway-Quad-band-1XGSM-GoIP-VoIP-/181075100268how ever our requirement are a bit wire like SMS in addition to Call capability. Tarek Sawah From: asterisk_l...@earthshod.co.uk To: asterisk-users@lists.digium.com Date: Fri, 11 Oct 2013 15:33:36 +0100 Subject: Re: [asterisk-users] GSM to SIP Adapter On Friday 11 October 2013, Tarek Sawah wrote: Greetings,I'm looking for a really cheap GSM-SIP gateway, Single channel (one SIM card). any suggestions? Tarek Sawah We've been using OpenVox G400P cards (PCI; there is also a G400E, which is PCI express for newer motherboards). Sends and receives text messages, and makes and answers phone calls. Accepts up to four RF modules, each of which accepts one SIM card. If you only need text message functionality (not voice calls), then almost any old mobile phone with a USB or RS232 cable can be used as a GSM modem -- and you probably have one lying in a drawer. -- AJS Answers come *after* questions. -- _ -- 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
[asterisk-users] chan_sip.c:9602 copy_header: No field 'CSeq' present to copy
Just put a new phone in place with the latest firmware from Cisco. This is the first SPA501G we have with this firmware. In the Asterisk CLI we are now seeing the error message below about once every second. When we unplug the phone, the messages quit. NOTICE[15539]: chan_sip.c:9602 copy_header: No field 'CSeq' present to copy Thanks in advance for any assistance on this. -- _ -- 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 11.6 nat problem
Jeremy Kister wrote: using asterisk 11.6.0-rc1 i just converted my nat=yes to nat=auto_force_rport,auto_comedia I have my asterisk box on the same subnet as a cisco 1760 (vgw1). a few times per day, Asterisk thinks vgw1 is dead (by qualify/options). A 'sip reload' always fixes the problem. i left 'sip set debug peer vgw1' on the console. but i dont see what's causing the issue.. http://kister.net/tmp/ast-sip.conf http://kister.net/tmp/ast-console.txt can anyone spot the issue? Jeremy, It looks like at some point Asterisk decides that vgw1's SIP port is no longer 5060. This may have to do with the NAT settings for that device: Before 'sip reload' | After 'sip reload' --|- * Name : vgw1 | * Name : vgw1 ... | ... Force rport : Auto (Yes) | Force rport : Auto (No) Symmetric RTP: Auto (Yes) | Symmetric RTP: Auto (No) ... | ... Addr-IP : 10.9.1.9:59934 | Addr-IP : 10.9.1.9:5060 ... | ... Status : UNREACHABLE| Status : OK (19 ms) Since the device is on the same subnet as your Asterisk server, you could try setting 'nat=no' for the vgw1 peer. That may not be a good long-term solution because of its security implications¹, but it could help determine if the NAT settings are the cause of the problem and serve as a stopgap until you figure out why the port is changing. Alternatively, you could try setting 'port=5060' for the vgw1 peer, but that's the default so it may still get changed. ¹ From sip.conf.sample: IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from the nat setting in a peer definition, then the peer username will be discoverable by outside parties as Asterisk will respond to different ports for defined and undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the other, then valid peers with settings differing from those in the general section will be discoverable. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9
Hi,
Seems a great workaround from Gareth Blades. Thanks I will try it.
Any way to make asterisk log a line in /var/log/messages ?
On 10 October 2013 19:44, Michelle Dupuis mdup...@ocg.ca wrote:
Gareth:
Did you check if your message (or security) log recorded anything during
these attempts? If so, can you post the content of the logs during this
attack?
M
--
*From:* asterisk-users-boun...@lists.digium.com [
asterisk-users-boun...@lists.digium.com] On Behalf Of Asghar Mohammad [
asghar...@gmail.com]
*Sent:* Tuesday, October 01, 2013 11:53 AM
*To:* Asterisk Users List
*Subject:* Re: [asterisk-users] Failed to authenticate user
1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9
Hi,
Bad boys trying to guess a valid username.
in sip.conf uncomment alwaysauthreject=yes and Asterisk always reject 1st
invite.
On Tue, Oct 1, 2013 at 5:26 PM, Gareth Blades
mailinglist+aster...@dns99.co.uk wrote:
On 01/10/13 15:44, gincantalupo wrote:
On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo
gincantal...@fgasoftware.com wrote:
Hi,
I get a lot of these messages on my Asterisk CLI:
Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS
;tag=03f82bb9
as if my PBX machine is trying to authenticate to itself. It seems
someone is attacking my asterisk PBX.
Is there a way to fix this problem?
in sip.conf I have guest connections permitted and have them going to the
default context which contains :-
[default]
; all unauthenticated connection attempts from the internet come in here.
exten = _[+*#0-9].,1,NoOp(Unauthenticated call attempt -
${SIP_HEADER(Contact)})
exten = _[+*#0-9].,n,Congestion
Then in fail2ban I have it match the following :-
failregex = Registration from .* failed for \'HOST\' - Wrong password
Unauthenticated call attempt .*\@HOST\:
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[asterisk-users] DTMF detection problem with analog card
Hi all. I have a DTMF detection problem by my new analog card (ATCOM 2 FXO port). When i`m playing a voice with 'GET DATA' AGI command, sometimes asterisk do not receive DTMF from caller while the voice is playing. But if user waits to the end of playing voice, there is no problem. I`m using Asterisk 10.3.1, dahdi-2.6.1 on CentOS.6.4. Could you please help me? Here is my configs: system.conf: fxsks=1 fxsks=2 loadzone = nl defaultzone = nl chan_dahdi.conf: -- [channels] ;=== ;General options ;=== usecallerid = yes hidecallerid = no busydetect=yes busycount=3 ;=== ;FXO Modules ;=== group = 1 signalling = fxs_ks context = my-context channel = 1,2-- _ -- 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] Capture Media IP in CDR
I am not proxying the media, but never the less I am forced to store the source media IP in my CDR, for regulatory reasons. Asterisk gets that information when the reinvite comes, but how do I store it? If I don't figure this out my next email will be from Federal Prison. Kindly help me stay away from those guys. Eventually we all need to save that information or we shall not be able to stay in 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] Capture Media IP in CDR
On Fri, Oct 11, 2013 at 9:05 PM, CDR vene...@gmail.com wrote: I am not proxying the media, but never the less I am forced to store the source media IP in my CDR, for regulatory reasons. Asterisk gets that information when the reinvite comes, but how do I store it? If I don't figure this out my next email will be from Federal Prison. Kindly help me stay away from those guys. Eventually we all need to save that information or we shall not be able to stay in business. You can add custom fields to your CDR records using Set(CDR(customfieldname)=foobar). I don't know the name of the variable you want that specifically contains the source media IP, but I imagine you can pull it with the SIP_HEADER function, or possibly the CHANNEL(recvip) function. -- Thanks, Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.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
