similar to: Asterisk register ip phone

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 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

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

Anything to do about portscans? Is there any way (should I) to see if the connection is a legit (only SIP currently) connection BEFORE my * answers? [2007-10-17 19:23:46] WARNING[4191]: chan_sip.c:6624 determine_firstline_parts: Bad request protocol 01@<ASTERISK_IP> SIP/2.0 -- Executing [s at default:1] Answer("SIP/sip.jmg.se-081dd730", "") in new stack [2007-10-17

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:

*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

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

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

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 guys still have the problem to setup the IVR correctly. I am forwarding call from ser : if (method == "INVITE") { if (uri =~ "sip:1[0-9]{10}@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("xxx.xxx.xxx.xxx:5061"); t_relay(); break; } } inside sip.conf

hello, I'm trying to use a remote SMB/CIFS share to backup a local directory tree. I use rsync "version 3.0.4 protocol version 30" and mount.cifs "1.10" with a kernel 2.6.18. The directory I'm trying to backup is quite big and has a lot of files inside. [root@linux dir]# du -hs . 50G . [root@linux dir]# find . -type f | wc -l 41929 I get the following error when

ok so my table looks like this... REATE TABLE `sip` ( `id` int(11) NOT NULL auto_increment, `name` varchar(80) NOT NULL default '', `accountcode` varchar(20) default NULL, `amaflags` varchar(7) default NULL, `callgroup` varchar(10) default NULL, `callerid` varchar(80) default NULL, `canreinvite` char(3) default 'yes', `context` varchar(80) default NULL, `defaultip`

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:

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

Asterisk-server behind Endian-firewall: SIP-aware, 5060 + RTP-ports opened and 5060 forwarded to Asterisk (192.168.2.2) Can someone see why SIP-registration fails ?? register => 092779077:XXXX at 85.119.188.3 [3starsnet] type=peer host=85.119.188.3 username=092779077 secret=XXXX fromuser=092779077 fromdomain=sip.3starsnet.com dtmfmode=rfc2833 canreinvite=no insecure=port,invite qualify=yes

Hi all, Supposedly my router is itself streaming traffic to clients. Is it possible for it to mark its packets before going through dsmark?? I have gone through the examples in iproute2/examples, i noticed that there''s always an ingress and egress of which these are 2 different dev. Is it possible for before ingress and egress be the same dev?? I tried the script below but

Hi,we have Centos 5.4 server and according to me we have strange problem. Disk size and other indormation like below.Normally,md2 partition should have 46GB free disk size but available value is zero.Why it show zero ? If you help me,I will be happy. df -h Filesystem Size Used Avail Use% Mounted on /dev/md1 19G 2.1G 16G 12% / /dev/md2 880G 834G 0

The Wine development release 5.5 is now available. What's new in this release (see below for details): - Builtin libraries use the new UCRTBase C runtime. - Compatibility mode used when reporting Windows version. - Better support for debug information in PE files. - Support for linguistic case mappings. - More attributes supported in WebServices. - Various bug fixes. The source

The Wine maintenance release 3.0.5 is now available. What's new in this release (see below for details): - Various bug fixes The source is available from the following locations: https://dl.winehq.org/wine/source/3.0/wine-3.0.5.tar.xz http://mirrors.ibiblio.org/wine/source/3.0/wine-3.0.5.tar.xz Binary packages for various distributions will be available from: