[asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate
I have service with both VoIPtalk.org and Andrews & Arnold (aa.net.uk). VoIPtalk calls are unauthenticated and reach me fine, but Andrews & Arnold calls are authenticated. The last call I successfully received was on Tuesday afternoon. Initially, A&A were for some odd reason not sending calls to my server, but that has been resolved. The problem now is that the calls fail to authenticate, and are therefore rejected - error 403 is presented to them, and I see this in Asterisk's console: [Apr 23 11:53:19] NOTICE[27398][C-0004]: chan_sip.c:25535 handle_request_invite: Failed to authenticate device "X XX" ;tag=201604231153191 I have checked that the username and password in my config agree both ends, and have even tried changing them. The bulk of my calls come in on A&A, so I am obviously trying to find out what has gone wrong. No-one else is seeing any problem. What do I need to do to track this down? -- Phil Reynolds mail: phil-aster...@tinsleyviaduct.com Web: http://phil.tinsleyviaduct.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate
Hello Phil, On Saturday, April 23, 2016, 12:19:15 PM, you wrote: > I have checked that the username and password in my config agree both > ends, and have even tried changing them. > The bulk of my calls come in on A&A, so I am obviously trying to find > out what has gone wrong. No-one else is seeing any problem. What do I > need to do to track this down? I have a couple of lines with A&A, and I have not been having any problems recently. When I have had similar problems in the past, it has been an issue with the SIP config. I originally had a number of contexts set up in sip.conf to handle the lines coming in (such as [aa-line1], [aa-line2]) each with their own username and password settings. The type=user setting was critical, because all the calls came from the same IP address, and using type=peer caused matching problems which resulted in authentication failures. This got too complex to manage once I added in all the IP addresses A&A calls might come in from. so I simplified the setup. I now have just one context in sip.conf to handle incoming A&A calls, with the same username for all lines, and type=peer. Calls are then sent to extensions.conf, where the calls are directed to the correct call-handler for the line based on the CID. Here is the setup in sip.conf for A&A calls: --- sip.conf [aa-incoming](!) type=peer context=aa-incoming insecure=invite transport=udp disallow=all allow=alaw trustrpid=yes sendrpid=yes ; IPv4 hostnames [voiceless-1](aa-incoming) host=a4.voiceless.aa.net.uk [voiceless-2](aa-incoming) host=b4.voiceless.aa.net.uk [voiceless-3](aa-incoming) host=c4.voiceless.aa.net.uk [voiceless-4](aa-incoming) host=d4.voiceless.aa.net.uk [voiceless-5](aa-incoming) host=e4.voiceless.aa.net.uk [voiceless-6](aa-incoming) host=f4.voiceless.aa.net.uk [voiceless-7](aa-incoming) host=g4.voiceless.aa.net.uk [voiceless-8](aa-incoming) host=h4.voiceless.aa.net.uk [voiceless-9](aa-incoming) host=i4.voiceless.aa.net.uk [voiceless-10](aa-incoming) host=j4.voiceless.aa.net.uk --- The trustrpid and sendrpid settings were important. --- extensions.conf (DNIDs changed) === [aa-incoming] exten => 4401,1,Goto(from-aa-line1,s,1) exten => 4402,1,Goto(from-aa-line2,s,1) exten => 4403,1,Goto(from-aa-line3,s,1) --- Hope this helps. Julian -- Best regards, Julianmailto:jb_s...@trink.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate
On Sat, 23 Apr 2016 22:45:32 +0100 Julian Beach wrote: > Hello Phil, > > I have a couple of lines with A&A, and I have not been having any > problems recently. When I have had similar problems in the past, it > has been an issue with the SIP config. I originally had a number of > contexts set up in sip.conf to handle the lines coming in (such as > [aa-line1], [aa-line2]) each with their own username and password > settings. The type=user setting was critical, because all the calls > came from the same IP address, and using type=peer caused matching > problems which resulted in authentication failures. This got too > complex to manage once I added in all the IP addresses A&A calls might > come in from. so I simplified the setup. > > I now have just one context in sip.conf to handle incoming A&A calls, > with the same username for all lines, and type=peer. Calls are then > sent to extensions.conf, where the calls are directed to the correct > call-handler for the line based on the CID. Here is the setup in > sip.conf for A&A calls: Actually, this is now sorted. It turns out the latest recommended configs on the A&A wiki had peer vs. user confusion. On correcting this, all was well. -- Phil Reynolds mail: phil-aster...@tinsleyviaduct.com Web: http://phil.tinsleyviaduct.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate
Hello Phil, On Saturday, April 23, 2016, 11:11:29 PM, you wrote: > Actually, this is now sorted. It turns out the latest recommended > configs on the A&A wiki had peer vs. user confusion. On correcting > this, all was well. I'm glad you found it. It look me a while to track down that problem when I had it. The one that was hardest for me to track down was a slight mis-match between the RTP ports in Asterisk and the corresponding ports open on a firewall, which resulted in about 1 in 10 calls having no audio! Doh! -- Best regards, Julianmailto:jb_s...@trink.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users