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Hi, I have made a fresh install of asterisk-1.6.2.10 and when I register my soft phone it gives following error. Rest are default configurations. 32.454370 MY_IP -> ASTERISK_SERVER_IP SIP Request: REGISTER sip:ASTERISK_SERVER_IP 32.454505 67.19.43.202 -> MY_IP SIP Status: 401 Unauthorized (0 bindings) 36.454814 MY_IP -> ASTERISK_SERVER_IP SIP Request: REGISTER sip:ASTERISK_SERVER_IP 36.454949 67.19.43.202 -> MY_IP SIP Status: 401 Unauthorized (0 bindings) 40.467996 MY_IP -> ASTERISK_SERVER_IP SIP Request: REGISTER sip:ASTERISK_SERVER_IP 40.468132 67.19.43.202...

Hey all, I am writing a plugin in which I want to stop the rendering of an action with an unauthorized response if the user is not authorized to view the resource. I am using a before filter to achieve this and inside that before filter I do it like so: render :text => "Unauthorized!", :status => :unauthorized, :layout => false The status is properly set since I see the...

I am getting a SIP 401 unauthorized message. My public IP or PIP is being pre-routed with iptables to goto an internal IP or IIP All the polycom phones in the office point to the IIP. they work fine. I have 2 external phones that are registering to the PIP. I see the register attempt as I am getting the 401 unauthorized messag...

Yesterday we moved one of our servers to a new IP. We updated DNS and various adapters configured to register to that server registered to the new IP correctly. All seemed to be well. This evening I discovered that with one exception, all of the adapters are getting a SIP/2.0 401 Unauthorized message back from asterisk. The exception is an Innomedia adapter -- Linksys PAP2's and (I believe) one Cisco ATA-188 are getting the "Unauthorized". I have stopped, restarted, unloaded & loaded sip, and erased astdb to start from scratch... no dice. None of the config fil...

SIP SECURITY WARNING Version: v1-0 (cvs today) Problem: sip context in general section ignored - goes to default - allowing unauthorized sip devices to place calls in default context Fix [workaround]: Remove or rename "default" context in extensions.conf Notes: I am not sure what other asterisk functionality may be affected by this - review your other config files for references to the "default&qu...

...lo, I have three clients that connect to one server. The server runs tinc 1.0.11 and one client runs tinc 1.0.19. I recently upgraded the two other clients from tinc 1.0.11 to tinc 1.0.16. Since the upgrade, the server now regularly logs messages of the form Aug 26 12:17:42 ebox tinc.rath[4049]: Unauthorized request from hspc (87.173.111.136 port 42836) or Aug 25 18:39:12 ebox tinc.rath[4049]: Unauthorized request from <unknown> (80.137.207.208 port 57335) (where hspc is the name of one of the clients). However, the VPN still seems to work just fine, including connections to the two upgra...

All- Recently an Asterisk server we host was hacked and used to route some unauthorized calls. We have since improved our security measures, including installation of fail2ban. The interesting thing is the way in which this was discovered. The unauthorized calls were occurring intermittently last Thurs evening thru Sat morning. On Sat morning, some of our employees were attemp...

Hi Friends, need to help. *I have problem about sip : SIP/2.0 401 Unauthorized* Is it require to nathelper module in kamailio ? *what can i write kamailio.cfg file when kamailio and Asterisk on same network?* Scenario is like as : ----------------------------- 1) kamailio server on 172.18.100.74 kamailio.cfg ( nathelpler module ) ----------------- loadmodule "nathel...

https://bugzilla.samba.org/show_bug.cgi?id=12544 Bug ID: 12544 Summary: Confusing return codes on unauthorized connections Product: rsync Version: 3.1.3 Hardware: All OS: Linux Status: NEW Severity: normal Priority: P5 Component: core Assignee: wayned at samba.org Reporter: mruprich at redhat.com...

...=all allow=alaw canreinvite=no dtmfmode=rfc2833 However, there's also a SIP account "3333" on that same server: [3333] type=friend secret=something host=dynamic nat=yes qualify=no disallow=all allow=alaw allow=ulaw canreinvite=no context=voipin When I make that call I get "401 Unauthorized" from remote-server. When I modify the username on 100.100.94.210 to "4444" or anything else that doesn't exist on 100.100.92.16, the call connects. tcpdump: 100.100.94.210.5060 > 100.100.92.16.5060: SIP, length: 839 INVITE sip:+1234 at 100.100.92.16 SIP/2.0 Via: SIP/2...

...from 10.241.0.2 port 50858 ... 1369133834 tinc.confine[2550]: Connection with node_2 (10.241.0.2 port 50858) activated ... 1369133834 tinc.confine[2550]: Got ADD_SUBNET from node_2 (10.241.0.2 port 50858): 10 3fba6e6f node_2 fd65:fc41:c50f:2:0:0:0:0/64#10 1369133834 tinc.confine[2550]: Ignoring unauthorized ADD_SUBNET from node_2 (10.241.0.2 port 50858): fd65:fc41:c50f:2:0:0:0:0/64#10 ... 1369133834 tinc.confine[2550]: Node node_2 (10.241.0.2 port 655) became reachable The node publishes that subnet and the server knows it beforehand from the existing node host file, but as you can see it sti...

I've been trying to use net2phone's sip service at sip.net2phone.com with * but keep getting SIP/2.0 401 Unauthorized. Do you know if this should be possible? So far: I can use an ata186 to connected directly to n2p through sip.net2phone.com without any special settings. I can connect from * to iconnecthere, but, whatever I try from * to n2p produces "SIP/2.0 401 Unauthorized" (Can forward the full...

...=1543062237 To: <sip:203@192.168.1.100>;tag=1543062237 Call-ID: 2259600998@192.168.1.100 CSeq: 0 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:203@192.168.1.100> Content-Length: 0 to 192.168.1.100:5060 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1571536832 From: <sip:203@192.168.1.100>;tag=1543062237 To: <sip:203@192.168.1.100>;tag=1543062237 Call-ID: 2259600998@192.168.1.100 CSeq: 0 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: &...

Hi there, I have tried very hard to setup the x-lite with asterisk, but until now i didn't get sucess. When i start the asterisk in debug mode, i see the message: sip/2.0 401 unauthorized. I know that this problem with authentication. I put in my sip.conf as below. [2203] type=friend username=2203 auth=md5 secret=1234 reinvite=no canreinvite=no dissallow=all allow=gsm context= sip host= 192.168.10.149 -> my machine that have xlite extension.conf [sip] exten =...

...host kernel: HiSax: TeiMgr Revision 1.1.4.1 Dec 12 16:25:35 localhost kernel: HiSax: Layer3 Revision 1.1.4.1 Dec 12 16:25:35 localhost kernel: HiSax: LinkLayer Revision 1.1.4.1 Dec 12 16:25:35 localhost kernel: HiSax: Approval certification failed because of Dec 12 16:25:35 localhost kernel: HiSax: unauthorized source code changes I found this on: http://info.ccone.at/INFO/ISDN4Linux/i4lfaq-7.html#ss7.26 What to do, where to find a rpm or ?? to fix ?? Thank you! HB Norway

...w picked up & am > running the recently released solr 8.7.0. > > In a test account, I've one message -- in the 'Drafts' folder. > > On exec of fts 'index' > > ????doveadm index -u testuser at example.com -q '*' > > I get a > > ????401 unauthorized > > indexing failure, only for the one dir+message in 'Drafts', > > ????2020-11-13 05:43:21 > indexer-worker(testuser at example.com)<8MKVNHmNrl/DJgEA+IOfAw>: Info: > Indexed 0 messages in Queue > ????2020-11-13 05:43:21 > indexer-worker(testuser at example...

...;ve built a new dovecot + fts-solr instance; I've now picked up & am running the recently released solr 8.7.0. In a test account, I've one message -- in the 'Drafts' folder. On exec of fts 'index' doveadm index -u testuser at example.com -q '*' I get a 401 unauthorized indexing failure, only for the one dir+message in 'Drafts', 2020-11-13 05:43:21 indexer-worker(testuser at example.com)<8MKVNHmNrl/DJgEA+IOfAw>: Info: Indexed 0 messages in Queue 2020-11-13 05:43:21 indexer-worker(testuser at example.com)<OHfzNHmNrl/DJgEA+IOfAw>: Info: I...

Hi, I'm trying to register SIP phone with an asterisk serve, failing miserably. The server is sending "401 Unauthorized" responses to the registration attempts, but every time the phone is re-REGISTERing without authorization. I'd think this was a problem with the IP phone, except... the very same phone registers correctly (authenticated) with another asterisk box, same brand, similarly configured. Th...

??? I am having a really bad day trying to get incoming calls to work on Asterisk 13 with PJSIP.? We just migrated from Asterisk 1.8 where everything was working but there seems that something got lost in translation.? No matter what I try I always get a 401 Unauthorized message when receiving a call from the PSTN provider.? I can make calls and the registration is working.? I have tried to set the identify to an endpoint that does not have an auth defined.? Anyone using Alestra SIP trunks in Mexico? Here is what I get on the cli: <--- Received SIP req...

Hello I use Asterisk 13 with FreePBX. When I try to connect my Softphone via VPN to Asterisk I'm registered and It's show via "pjsip list contacts" Then I try to call an internal number / other extension I get the following: "SIP/2.0 401 Unauthorized". The VPN net is list in pjsip.transports.conf:local_net=10.8.0.0/24 and sip_general_additional.conf:localnet=10.8.0.0/24 I'm not sure whats wrong here. Best Regards