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It says failed because it is failed to find this extension in the given context. According to your sip.conf default context for your SIP users is default, but all patterns that can handle 919876543210 are in general context. So Asterisk can't find them when you placing a call since the call flow doesn't go through general context. You probably want to set ...


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It answers you it's not a valid extension because it cannot find this extension in the given context. Maybe there is an error in what you're dialing vs what you got in the extension.conf For example, it looks like you're dialing number starting with 91987 when in your extensions.conf I see only pattern to handle numbers starting with 91999. Are you ...


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You haven't posted any addition details since asked for a couple of weeks ago so I'll try to answer as best I can (and recommend the question be closed): Security: If your goal is to secure your Asterisk server then a proxy server / kamailio is not the answer. See Asterisk Security for details on how to do this. Network: If your problem is NAT traversal ...


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try to rename ; Babytel [general] to ; Babytel [default]


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Your LDAP database has this suffix: dc=my-domain,dc=com You're trying to add an entry with dn: cn=1000,ou=sip,dc=digium,dc=internal LDAP is an hierarchical database, you must fix the DB suffix or the entry DN in order to fit your hierarchy needs. You can do it changing the entry dn with: cn=1000,ou=sip,dc=my-domain,dc=com But still the Organisation ...


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I don't know a lot about asterisk but I've some experience with graylog. I'd recommend starting with one of their downloadable virtual machines (thought they now have nice packages for redhat/centos/debian/ubuntu. This is very much a simplification but there are basically 2 types of input for graylog; GELF, and syslog. GELF (Graylog Extended Logging ...


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Kamailio will definitely not solve the problem with video, but it can offload some SIP processing from Asterisk, and add security. Also it allows to place multiple media servers behind one Kamailio server, thus allowing for load-balancing and failover. I saw somewhere that recent Chrome update has broken compatibility with WebRTC implementation in ...


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A pure SIP proxy let you to manipulate all SIP message, this offers you a lot of fexibility and will help you dealing with NAT issues, failover and other feature not easy to implement with asterisk. Asterisk is a great and full-features PBX but doesn't offer many options for treating SIP messages ( this by design because is projected to be a PBX, not a SIP ...


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Realtime doesnt support inlcudes within realtime. Thereby you can tweak a little with the Goto function. If you do this in your extensions.conf [context1] switch => Realtime [context2] switch => Realtime And then in the database do this: [context1] exten => _XXXXXXXXX,1,Goto(context2,${EXTEN},1) The above needs to translated to a query, I ...


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In future, please at least post a sample of the code you've tried so that we can see what you're trying to do. It helps us understand your objectives. That said, give this a try. Please understand this is untested code I pretty much wrote while having my morning coffee. You will need to alter it to suit your needs / make it work. I am being pedantic ...


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Wait for silence and amd both waiting for period when no significant sound So your channel always HAVE sound. Most likely reason - bad quality line. You can try fix that by changing silence threshold. For AMD it is last param ...


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I have found an answer. alwaysauthreject allowguests These 2 functions aren't part of pjsip.conf anymore and are standard. Furthermore you need to secure your server with a firewall and fail2ban. Security needs to be done on OS level.


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It's not clear what type of security options you are looking for - since simplistic restrictions like refusing guest, requiring registration, etc. will not reduce the load created by SIP scanners, hackers trying to get in, etc. If you are trying to secure your server check out this page www.voip-info.org for details. Solutions range from basic Asterisk ...


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-r is for a "running" Asterisk instance that you wish to connect to. When you exit the connection, Asterisk continues to run in the background. -c is for starting up Asterisk in "foreground / console" mode. As soon as you exit that console, Asterisk shuts down. As always, if this response answers your question, please 'accept' it for the benefit of ...


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The 401 Unauthorized isn't an error: is how authentication mechanism works on SIP (see this answer) The line: [Mar 23 20:02:10] NOTICE[4067] chan_sip.c: Registration from '<sip:201@192.168.0.99>' failed for '192.168.0.15:39462' - Wrong password Seems clear: the password in your SIP client is wrong and the authentication doesn't succeed. SIP ...


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I found some very strange things going on with IAX2 and PFsense. I've got the latest PFsense (2.2.1 Mar 13 2015) and this has been consistent with all my experiences prior, as well: PFsense appears to be NOT applying NAT rules to IAX2 packets. If I set up a packet dump on pppoe0 (going to my DSL carrier) I see packets coming from 192.168.1.168 port 4569, ...



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