similar to: chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for public_ip:2049

Hello list, how come on my Asterisk 1.6.2.11, I have no help available ?! asterisk*CLI> core show application Dial -= Info about application 'Dial' =- [Synopsis] Not available [Description] Not available [Syntax] Not available [Arguments] Not available [See Also] Not available Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed...

Hello friends: I am facing cutoffs randomly when negotiating calls. The PBX dials the destination, the provider (softswitch) receives the request *[1]* and sudenly the PBX hangs up the call* [2]* while the provider is still dialing it, as a consequence the remote peer receives a ghost call. Along the atempt I could see six times a messages regarding NAT isuues *[3]* I hope anyone can give me an

Hi, I noted on my 4.7 machines that when a ssh conection is made, the following PTR query happens (10.11.1.11 is the src address in the example): 13:23:21.120290 PUBLIC_IP.4523 > PUBLIC_IP.53: 52788+ PTR? 11.1.11.10.in-addr.arpa. (41) 13:23:21.120517 PUBLIC_IP.4524 > PUBLIC_IP.53: 52788+ PTR? 11.1.11.10.in-addr.arpa. (41) 13:23:21.120683 PUBLIC_IP.4525 > PUBLIC_IP.53: 52788+ PTR?

Hello list, has anyone else also noticed spontaneous reboots ?! I noticed this today and also yesterday. Can't really see if there is a fixed time between the reboots. Normally al of my SIP peers are registered. When I put up the CLI today I saw that a lot of SIP accounts where UNREACHABLE and needed to register again (what they slowly did). These are realtime SIP peers that reside on

Hello All, my asterisk server is constantly under attack [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '"4941" <sip:4941 at public_ip>' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '"4941"

Hello, I'm having difficulty with registering a SIP account in a Snom 320 IP-phone. This is what sip debug tells me : [Oct 7 13:28:42] VERBOSE[20314] chan_sip.c: [Oct 7 13:28:42] <--- SIP read from UDP:public_ip:58697 ---> REGISTER sip:sip.domain.tld SIP/2.0 Via: SIP/2.0/UDP 192.168.114.200:2048;branch=z9hG4bK-vj1xvbdnp4dw;rport From: <sip:test3 at

Hello. I'm using asterisk 1.4.30. I've found this patch for app_queue.c : https://issues.asterisk.org/view.php?id=11700 Can I easily implement this by issuing : */wget 'https://issues.asterisk.org/file_download.php?file_id=17192&type=bug' -O - | patch -p0/* ?? Does this mean I have a "patched" asterisk ? (I ask this because some applications require a

Hello list, whenever I press the #-key I hear a voice saying 'transfer'. How can I use the #-key without this voice-message or without having it the function of unattended transfer ?! The T or t option is not set in my Dial()-command so I don't know where this transfer is coming from in the first place. Kind regards, Jonas. -------------- next part -------------- An HTML

Hello list, my CLI is spammed with : [Sep 13 08:31:38] > doing dnsmgr_lookup for 'ssw6.itsp.tld' [Sep 13 08:31:38] > doing dnsmgr_lookup for 'ssw6.itsp.tld' [Sep 13 08:31:47] > doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep 13 08:31:48] > doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep 13 08:31:49] > doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep

Hello, I have this in my dialplan : exten => s,n,Set(vgLabel=vg(${number}+1)) exten => s,n,GoTo(${vgLabel}) But in stead of vgLabel becoming the SUM of 2 numbers, it is just a string : [Nov 3 16:17:27] -- Executing [s at macro-f:43] Set("SIP/test-00000002", "vgLabel=vg(1+1)") in new stack [Nov 3 16:17:27] -- Executing [s at macro-f:44]

Hello, I read on the wiki : Asterisk 1.8 will allow to read SIP response codes in the dialplan via *${HASH(SIP_CAUSE,<channel-name>)}*. Additionally make sure you're using the destination channel, not the source channel. But when I use this in my dialplan, this 'variable' is empty. Dialplan : exten => h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,${CHANNEL})}) exten =>

I''d like to convert some simple, general ruby scripts I have into rails apps just for testing purposes. Many of these apps are not DB driven, so the whole CRUD concept does not apply to them... here''s a sample: require ''socket'' server = TCPServer.new(''12345'') while (session = server.accept) Thread.new(session) do |this_session|

Hello, can I use include-statements in the calendar.conf configuration file ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131113/8aaffcd8/attachment.html>

I''ve had some issues with my network, and I''ve had to reconfigure my Gibraltar CD. It runs shorewall 1.4.8, and I have a 2-interface setup, so I downloaded the relevant files from the install page. Masq and such works, but I''m having a problem with my port forwarding. It works for port 22, but it doesn''t seem to work for any other port. I''ve turned

Just done a clean install of rc5 on a totally new machine and found the following:- /etc/init.d/asterisk start errors on line 109 - there is no 0 before $VERBOSITY as in the other lines. More interesting is that after make samples I have no iax2 available. Dave Cotton

Andrew, Just passing the public IP of our samba to samba_dnsupdate is enough to replicate the data correctly? samba_dnsupdate --verbose --all-names --current-ip=PUBLIC_IP On Fri, Sep 25, 2020 at 6:39 AM Andrew Bartlett <abartlet at samba.org> wrote: > On Thu, 2020-09-24 at 14:59 -0300, Elias Pereira via samba wrote: > > Hello, > > > > Has anyone already installed

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1"> </head> <body text="#000000" bgcolor="#ffffff"> <font size="+1">Does anyone have links to the most recent audiocodes

I downloaded the latest 1.8 (27922) but got some segmentation faults. The first one was when it loaded cdr_odb, and so I changed menuselect not to compile that one, but the second one was when it tried to load chan_agent and so I stopped there to see if anyone else was seeing this. The agents.conf is all commented out except for [general] . Anyone know what is happening? Thanks. P.S. I deleted

Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only

How can I let asterisk immediately dials a trunk when off hook?