similar to: 401 Unauthorized with Snom but not with Zoiper softphone

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 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"

Dear all, I'm in basic setup of my network: I try to do a call from a softphone to an other one but I got the error 603 Declined. Below the sip.conf: *[gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial* *[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial* extension.conf: *[tutorial] exten => 1234,1,Dial(SIP,gianca)* *exten

Hello! I bought a virtual IP line to my ISP to use with my asterisk but when I try to connect it to my ISP tells me I can not use and I can only use with a softphone that gives me, xlite ready configured. I use ngrep to see what information sent on xlite for communication, the User-Agent was changed so I change the User-Agent to my asterisk to the same as saying the xlite but still does not work.

I am helping a friend on one of his sip trunk and couldn't find the way to resolve his problem. His asterisk's problem is like this: 0. When incoming call to one of his sip trunk, Asterisk reply with "488 Not acceptable here". So the call get dropped. 1. Recently upgraded Elastix with Asterisk 1.4.33 2. Was working fine before the upgrade 3. There are total 4 SIP trunks

Hi, I'm running Asterisk 1.8.11.1 @office. The Broadvoice service work fine with all 1.6 version and early 1.8 behind a NAT but about 2 months ago stop working. No made changes in the firewall NAT rules. Right now I'm @home via my Xlite softphone working fine without problems Any suggestions or thoughts? Alex Celi This is the info central*CLI> sip show peers Name/username

Hi all my asterisk server, 2 sip client softphones are the same LAN asterisk ip address : 192.168.1.5 sip client 1 : 192.168.1.4 sip client 2 : 192.168.1.2 asterisk starts ok with sip setup the sip.conf [test] type=friend username=test secret=1000 host=dynamic context=cucku directmedia=yes directrtpsetup=yes [1000] type=friend username=1000 secret=1000 host=dynamic context=cucku

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

Does this help: Session Initiation Protocol (REGISTER) Request-Line: REGISTER sip:1.2.3.4;transport=TCP SIP/2.0 Method: REGISTER Request-URI: sip:1.2.3.4;transport=TCP Request-URI Host Part: 1.2.3.4 [Resent Packet: False] Message Header Via: SIP/2.0/TCP 192.168.1.15:47053 ;branch=z9hG4bK-d8754z-5e3d9f441f1de1d3-1---d8754z-;rport;transport=TCP

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|

Hi! AFAIS the incoming SUBSCRIBE is handled in the same context as INVITE - but how should I handle the SUBSCRIBE in the context? thanks klaus SUBSCRIBE sip:u+431234567 at foobar.at:5160 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.82:39982;branch=z9hG4bK-d8754z-3116e1207913aa4e-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:u+431234567 at 11.111.11.11:39982> To:

Hello, I noticed a strange thing while testing my Asterisk-Kamailio Realtime setup. In an INVITE the From and To headers contain the same number when calling through a Realtime integration setup. This happens when the INVITE leaves Asterisk. Can you guys tell me what might be causing this? I have 660 at testers.com as a websocket client and 700 at testers.com (caller) using a Zoiper client (db

Well, I thought it worked, but it actually doesn't--I am able to get the caller pick up the phone, but for some reason, I cannot hear anything on either side no matter who does the calling. Again, my two SIP phones are on the local 192.168.1.0/24 network (do not go over the Internet) and the Asterisk server is located in the same network (not accessed over the Internet). Any help is

Today I upgraded my Asterisk to the new 1.4.39.1. One of sip trunk stop working after the upgrade. Here is the sip debug: --------------------------------------------------------------------------- <--- SIP read from 208.65.xxx.xxx:5060 ---> INVITE sip:1778xxxxxxx at 10.11.22.77:5060 SIP/2.0 Via: SIP/2.0/UDP 208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-d9175178645e9146-1---d8754z-;rport Via:

Hello, what does this message mean ? [Oct 12 14:03:32] DEBUG[9064] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for public_ip:2049 I find this in my debug log file when "core set debug 25". Is something failing, or is this just informative ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, i am stuck in NAT issue on ec2 cloud computing from last 2-3 days , may be some of you are doing setup and integration on cloud. below is my setup details which may help you to suggest me solution. Asterisk version : 1.6.2.6 1) Kamailio server having public_ip as well local ip .i am using mediaproxy [also tried rtpproxy] . 2) Asterisk server having public_ip as well local ip. setup:

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

Hi list, We, an OSS advocacy group, setup a Xen 3.1 machine composed of : . a 64 bits dom0 running Debian stable amd64 . 2 hvm domUs running OpenBSD amd64 . 2 hvm domUs running NetBSD i386 This machine is to be hosted and reachable from the Internet, but it will only have one public IP. Naturally, our first tought was to port-forward using iptables / netfilter. We didn''t really

Dear All, Today I upgraded asterisk 1.6 to 1.8. As the result of this when peers trying to register to asterisk the system shows: NOTICE[24698]: chan_sip.c:23417 handle_request_register: Registration from '"50" <sip:50 at 192.168.1.109> <sip:50 at 192.168.1.109>' failed for ' 192.168.1.80:5062' - Wrong password even though on 1.6 everything was OK here is

I tried to do what I with regular SIP to Transfer a call via 302 Redirect. In asterisk 12 we need to add the Tech, or not, but in any case, there is no transfer done. The call is closed. Here is a trace. How do I do this? [Jul 9 21:39:29] DEBUG[47716][C-00000002]: pbx.c:4869 pbx_extension_helper: Launching 'Transfer' -- Executing [17274428141 at redirect:30]