search for: asterisk_ip

...rname="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 from. # U CISCO_IP:50339 -> ASTERISK_IP:5060 REGISTER sip:ASTERISK_IP SIP/2.0..Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK75640688..From: sip:phonenumber@ASTERISK_IP..To: sip:phonenumber@ASTERISK_IP..Call-ID: 00036b09-607e0002-3377e59b-2e4ebe4d@CISCO_IP..CSeq: 41929 REGISTER..User -Agent: CSCO/6..Contact: <sip:phonenumber@CISC...

...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 from. # U CISCO_IP:50339 -> ASTERISK_IP:5060 REGISTER sip:ASTERISK_IP SIP/2.0..Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK75640688..From: sip:phonenumber@ASTERISK_IP..To: sip:phonenumber@ASTERISK_IP..Call-ID: 00036b09-607e0002-3377e59b-2e4ebe4d@CISCO_IP..CSeq: 41929 REGISTER..User -Agent: CSCO/6..Contact: <sip:phonenumber@CISC...

...ster 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 SIP/2.0. 3. Via: SIP/2.0/UDP ASTERISK_IP:5060;branch=z9hG4bK2d757382;rport. 4. From: "Unknown" <sip:Unknown at ASTERISK_IP>;tag=as5a8c9f3b. 5. To: <sip:OPENSIPS_IP>. 6. Contact: <sip:Unknown at ASTERISK_I...

...************************* Ruleset for an old reliable IPChains firewall: file: /etc/rc.d/init.d/firewall This actually opens up a few more holes for some outbound streams. Can't remember exactly why I did it this way but it works good. # VIOP - asterisk # vars $EXT_IP=your.external.ip.here $ASTERISK_IP=your.asterisk.server.ip # #chains ipmasqadm portfw -a -P udp -L $EXT_IP 5060 -R $ASTERISK_IP 5060 ipchains -A portfw -s 0/0 1024: -d $EXT_IP 5060 -p 17 -j ACCEPT ipmasqadm portfw -a -P udp -L $EXT_IP 4569 -R $ASTERISK_IP 4569 ipchains -A portfw -s 0/0 1024: -d $EXT_IP 4569 -p 17 -j ACCEPT ipmasqadm...

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 19:23:46] WARNING[4191]: chan_sip.c:6624 determine_firstline_parts: Bad request protocol 01@<ASTERISK_IP> SIP/2.0 [2007-10-17 19:23:56] WARNING[10123]: pbx.c:2505 _...

*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 Regards* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/2008...

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: 651194bd76e02f4d0126373c515685e4 at 127.0.0.1 CSeq: 104 REGISTER User-Agent: Asterisk PBX 1.6.2.5 Authorization:...

...n 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 functions seem ok). The Snom 200 I have dials "sip:asterisk@asterisk_IP" which I have an extension setup for and works like a charm. Can anyone with a 190 tell me if they also experience this problem and whether its something that needs fixing on the phone side or something I can handle on the Asterisk config side. thanks for any help Alex Dear Friends of...

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:

...ersions, 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 did a very basic res_digium_phone.conf file: [general] service_discovery_enabled=no service_name=Asterisk server registration_address=asterisk_ip registration_port=5060 userlist_auth=globalpin config_auth=globalpin globalpin=1234 config_auth=disabled file_directory=/etc/asterisk/digium_phones [network] type=network alias=home cird=0.0.0.0/0 registration_address=asterisk_ip registration_port=5060 [Home_phone] type=phone full_name=Mr Sandman...

...isk-->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 '76c64f8373def68d5088690816c67493 at Asterisk_IP'. Giving up. Please find the whole trace at http://pastebin.com/d764514b4 Can you please help me in order to find the real issue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090320/5a10eac8/atta...

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

...g call in my following context, I keep getting the below error from the perl accounting script running in background: Here is the context: [astrad] exten => 5444,1,SetVar(RADIUS_Server=IP_RADIUS) exten => 5444,2,SetVar(RADIUS_Secret=Secret_RADIUS) exten => 5444,3,SetVar(NAS_IP_Address=ASTERISK_IP) exten => 5444,4,SetAccount(${CALLERIDNUM}) exten => 5444,5,agi,radauthentic.pl|AuthorizeBy=Account&IfFailed=DoNotHangup; a copy of agi-rad-auth.pl exten => 5444,6,agi,radauthor.pl|AuthorizeBy=Account&IfFailed=DoNotHangup; a customized copy of agi-rad-auth.pl exten => 5444,7,Go...

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