similar to: Portscans and Asterisk

Hi all, I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy is not replying back...The issue is the UNKNOWN username that reside in the OPTIONS packet as you can see in the captured packets as you can see below: 1. U Asterisk_IP:5060 -> OPENSIPS_IP:5060 2. OPTIONS sip:OPENSIPS_IP

Hi all!! I have a really weird problem. I upgraded 2 Asterisk 1.2 boxes to Asterisk 1.4.18. Both are home PBX's and both boxes register to a SIP DID at exactly same provider. One box runs without errors on the console, the other box keeps repeating : [Feb 11 23:40:29] WARNING[11292]: chan_sip.c:6705 determine_firstline_parts: Bad request protocol Packet When i set debug on, it seems to

Hi list, After a recent upgrade to Asterisk v1.4.14, my message log is now filling up with the following error messages: <-------------> [Dec 29 17:24:52] WARNING[10655]: chan_sip.c:6645 determine_firstline_parts: Bad request protocol Packet --- (1 headers 0 lines) --- bitis*CLI> <--- SIP read from 82.101.62.99:5060 ---> Cirpack KeepAlive Packet <-------------> Seeing

*Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 > Asterisk_IP.5060: UDP, length 345 What i need to know please what TTL means specifically and what is the best value og TTL and what is the lengh vale mean

Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on

Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on host it's comming

Hi all, I got a couple of Snom 190's through this week and after some initial foolishness I have them both setup no problems. But here comes the except. When there is voicemail waiting the softbutton appears but the phone always dials its own extension. No matter what I put into the "mailbox" parameter on the line settings. (Phone registers correctly with * and all standard

Just responding in case this may be of help to somebody with firewalling issues. Not sure if this is off on a tangent to the original question... Here are three different forms of common firewall scripts and ways of getting SIP to work behind them. The third one has some additional stuff beyond just SIP although I can't remember why I wrote it that way. I've been having no fun using

Any recommendations on how I can get mathematical equations into my resulting html? (Besides using x^2^ type markup.) I figure I could parse out equation text from my main text before RedCloth ever sees it... and then maybe create png''s somehow from it, putting the links to them back into the main text before handing it over to RC. But I don''t know of a util that will generate

Hi all, I appologise in advance if this is a little OT, but I am building a box that will serve as firewall and router for a small ''internet cafe / netcafe'' and am using CentOS... So here it is: What are the best tools to be used for keeping the potential script kiddies from ''harming the Internet'' :) ? I specifically want to be able to detect and prevent

The SIP trunking service that I am trying to set up keeps saying that my registration from Asterisk is invalid. Asterisk registration: REGISTER sip:{registration_ip} SIP/2.0 Via: SIP/2.0/UDP {asterisk_ip}:5060;branch=z9hG4bK5c2eb10c;rport Max-Forwards: 70 From: <sip:{registration_user}@{registration_ip}>;tag=as5579cc0c To: <sip:{registration_user}@{registration_ip}> Call-ID:

Was in the process of upgrading a 7960 to SIP and in advertently applied a skinny image (P003G302.bin), now no matter what i put in OS79XX.TXT and OS7960.TXT it simply wont upgrade. The phone is pulling down OS79XX.TXT from my TFTP server but then goes on to repetedly ask for SEP<MACaddr>.cnf. Help!! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Fails because libusb_close does not have a prototype early enough in libusb.c. Adding it before the call fixes it. I haven't started running it yet, but at least it now compiles. -- John-Mark Gurney <jmg at cryptography.com> Cryptography Research, Inc. +1 415 397 0123 ext. 332

Hi, all What is the use of astdb? Is it used to store realtime values like sip etc. Regards, Bhrugu Mehta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100127/b13d6a62/attachment.htm

Greetings all, After a long day of fighting with GTalk and having it finally working, I wanted to setup DPMA on my Digium phone. So first of all, I had to reinstall it all and reconfigure it all, since it works only on certified versions, and my installation was not from the certified branch. It took a long time of recompiling, testing, adding missing stuff, but I got it straight. Now, I

Dear All, I'm trying to send FAX to an endpoint Behind NAT...The scenario i the following: PSTN_GW-->Asterisk-->asterisk-->OpenSIPS-->Endpoint behind NAT.. The FAX is failed and I got the following error log on asterisk: Mar 20 09:21:09] WARNING[20388]: chan_sip.c:12409 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog

I was about to copy wmmintrin.h for the FreeBSD project, but I noticed that the file (clang/lib/Headers/wmmintrin.h) does not have a copyright statement in it, though it says: * The above copyright notice and this permission notice shall be included in * all copies or substantial portions of the Software. Also, the license is significantly different than what is contained in LICENSE.TXT, as the

Hello, i use shorewall for a very long time (2 years or so) and i use it for nat and as firewall....i now use portsentrys to detect portscans but there is one problem...i use the HOWTO from the shorewall mailing list to make portsentry and shorewall work together....but there is one prob portscans get detected and a drop rule is added to shorewall for example shorewall drop 62.178.xxx.xx

Dear all, I have installed asterisk 1.0.5 on redhat 9 I have installed also, asterisk-oh323 0.6.5 module (successfully compiled and installed) Now When I am trying to get asterisk communicate with a Radius (in my case: it's the VoiceMaster Radius) I was able to do the following: After installing all recommended to download and install radius client for asterisk

Hi All Just received my first 7905G from a distributer here in Sweden. According to the spec this phone should be able to use SIP. Now I been looking on Ciscos home pages for several hours trying to find a "SIP image" for this phone. No luck at all, need special access to be able to download software to this phone. Is it the fact, that I have to pay for a contract of some kind to be