Re: [asterisk-users] Failed to authenticate
Hello Antony, Monday, August 9, 2021, 4:14:11 PM, you wrote: > You want to look for firewall rules which will allow UDP in both directions > on > ports 1 - 3 (typically, may vary a bit, but something like that), or > alternatively, look for any rules which would block this, and remove them. This. Check that the port range in rdp.conf matches that in your firewall UDP settings. Even a slight mismatch in the ranges can result in a surprising number of calls with one-way audio which seem to happen in clusters. -- Best regards, Julianmailto:jb_s...@trink.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 13 takes over an hour to clear the MWI light
Hello Thelma, Thursday, December 24, 2020, 9:26:53 AM, the...@sys-concept.com wrote: > In astersik-11 MWI light was cleared as soon as I checked the message. > In asterink-13 it takes about 20min to set the light ON and the light > takes over an hour to clear. I had this problem following an upgrade between releases of Asterisk 13 last year, but I upgraded to Asterisk 16 and the problem went away without any need for configuration changes. Julian -- Best regards, Julianmailto:jb_s...@trink.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] which linux for asterisk?
Title: Re: [asterisk-users] which linux for asterisk? On Wednesday, December 9, 2020, 11:07:55 AM, Antony Stone wrote: > Upgrading a Debian server to the next release is a whole lot easier than doing > a CentOS one. It will be slightly easier for me having all Debian-based boxes, but my main concern will be the Sangoma drivers for the telephony cards, which are developed and tested on Centos and why I use Centos for my Asterisk server now. Centos 7 is supported until 2024, so there is a bit of time to think about it. -- Best regards, Julian mailto:jb_s...@trink.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Block Spam Calls
Hello Jeff,
Friday, December 13, 2019, 7:42:38 PM, you wrote:
> Mind posting your dialplan code? I was thinking the same thing - very
> much like an old spam control program I used to use whose name now
> escapes me. First time senders would have to respond to an auto-reply,
> then were added to a whitelist.
This is not the neatest of code segments, as there are a few checks I
make before I let calls through to the phones:
(1) It checks for blacklisted callers, and sends them off to a go-away
message
(2) It checks whether there is Caller ID information, and sends sends
anon calls to another handler (again asking callers to press 1 or
leave a message) but there is also a switch to just send them to
voicemail: I had a problem with 1 anon caller phoning repeatedly a few
years ago, and that was the solution!
(3) For calls with a CID number, it checks whether these have called
before and if not, adds them to the previous caller db then sends them
to an IVR to play them a message and ask them to press a key (this
means that if they call again, they will always go through to the phones, as
a number of genuine callers cannot, or won't, dial 1 - if they turn
out to be a robocall, they are blacklisted)
(4) Calls that pass all the tests go through to the phones
I am sure that I could make the code much more efficient, but it has
not been a priority and I have been adding features as I need them,
rather than re-writing!
[from-pstn] (initial handler for PSTN calls)
exten => s,1,Log(NOTICE, Incoming PSTN Call from CDR ${CDR(src)} and CID
${CALLERID(name)})
exten => s,n,GotoIf(${DB_EXISTS(blacklist/${CDR(src)})}?block) ; Check whether
caller blacklisted
exten => s,n,Set(DB(callers/lastcall)=${CDR(src)})
exten => s,n,Set(GLOBAL(CALLEDLINE)=L1-)
exten => s,n,GotoIf($["${CDR(src)}" != ""]?toincoming) ; if there is valid CID
go to IncomingHome
exten => s,n,GotoIf($["${CALLERID(name)}" = "WITHHELD"]?nocid) ; if anon call...
exten => s,n,GotoIf($["${CALLERID(name)}" = "INTERNATIONAL"]?nocid)
exten => s,n,GotoIf($["${CALLERID(name)}" = "UNAVAILABLE"]?nocid)
exten => s,n,GotoIf($["${CALLERID(name)}" = "PAYPHONE"]?nocid)
exten => s,n,GotoIf($["${CDR(src)}" = ""]?drop)
exten => s,n(toincoming),GotoIf(${DB_EXISTS(previous/${CDR(src)})}?:unknown) ;
Check whether caller has rung before
exten => s,n,GoSub(IncomingHome,s,1) ; answer the call in the macro
exten => s,n(nocid),GotoIf($[ $[ "${DB(variables/ANONTOVM)}" = "1" ] ]?vm) ;
...check whether anon goes straight to vm
exten => s,n,GoSub(IncomingHome,s,1) ; answer the call in the macro
exten => s,n(block),GoSub(Handler-MarketingCall,s,1) ; deal with blacklisted
callers
exten => s,n(vm),Log(NOTICE, No CID data or AnontoVM set) ; send anon callers
to AnonCall IVR
exten => s,n,Goto(AnonCall-menu,s,1)
exten => s,n(unknown),Log(NOTICE, Unknown caller to IVR) ; send unknown callers
to UnknownCaller IVR
exten => s,n,Set(DB(previous/${CDR(src)})=1) ; add number to previous caller DB
exten => s,n,GoSub(UnknownCaller-menu,s,1,(${CDR(src)}))
exten => s,n(drop),Log(NOTICE, No CID data, call dropped) ; drop calls with no
CID info
exten => s,n,Wait(15)
exten => s,n,Hangup()
[UnknownCaller-menu] ; The Unknown caller IVR - with some logging of
call volumes
exten => s,1,Log(NOTICE,Unknown Caller ${ARG1} - Caller to IVR)
exten => s,n,Verbose(2,Anon Calls = ${DB(unknown/total)} Total,
${DB(unknown/valid)} Valid, ${DB(unknown/invalid)} Invalid)
exten => s,n,Set(unknown=$[${DB(unknown/total)} + 1])
exten => s,n,Set(DB(unknown/total)=${unknown})
exten => s,n,Verbose(2,Unknown Calls set to ${DB(unknown/total)})
exten => s,n,Answer(500)
exten => s,n(loop),Background(unknown)
exten => s,n,WaitExten()
; 1 - Call to incoming call handler
exten => 1,1,NoOp(Real caller progressing)
exten => 1,n,Log(NOTICE,Unknown Call - Caller dialled to continue to phones)
exten => 1,n,Set(valid=$[${DB(unknown/valid)} + 1])
exten => 1,n,GoSub(IncomingHome,s,1)
exten => 1,n,Hangup()
; 2 - invalid entry
exten => i,1,Playback(invalid)
exten => i,n,Log(NOTICE,Unknown Call - Caller dialled invalid number)
exten => i,n,Set(invalids=$[${DB(unknown/invalid)} + 1])
exten => i,n,Set(DB(unknown/invalid)=${invalids})
exten => i,n,Goto(s,loop)
; 3 - timeout
exten => t,1,Playback(invalid)
exten => t,n,Log(NOTICE,Unknown Call - Caller timed out)
exten => t,n,Set(invalids=$[${DB(unknown/invalid)} + 1])
exten => t,n,Set(DB(unknown/invalid)=${invalids})
exten => t,n,GoSub(subMessaging,s,1,(${ARG1}),("Unknown"))
exten => t,n,Hangup()
--
Best regards,
Julianmailto:jb_s...@trink.co.uk
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Re: [asterisk-users] Block Spam Calls
Hello Doug, Friday, December 13, 2019, 11:03:37 AM, you wrote: >> This is exactly what I do - “press 1 for a human” >> Works great > I do this as well, but I also do a database lookup to see if the number > is on our speeddial list and if so, pass the call directly on without > the IVR prompts. I do something similar for calls without caller ID, but I was still getting robocalls with spoofed caller ID. I have now changed the dialplan slightly so that the first time people call they are asked to dial 1. After the first call, they are added to a known caller list and get straight through, and any robocalls at that point are blacklisted manually. I have found that most robocallers spoof the Caller ID so rarely call from the same number twice. It means that legitimate callers who cannot dial 1 just have to dial again to get through to the phones - there is a recorded message telling them to dial 1 or call back. I haven't had a robocall since! The hardest thing about this was extracting all the numbers of previous callers from the CDR and adding it to the Previous_Callers AstDB for the lookup. I didn't want to make existing callers go through the initial learning process. -- Best regards, Julianmailto:jb_s...@trink.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] clarification on gosub, macros and AEL
Hello Joshua, Wednesday, October 16, 2019, 10:39:27 AM, you wrote: > The module is still present, it just isn't built by default. It > requires explicit enabling using menuselect. I was obviously too concerned about the rebuilding of wanpipe with a new kernel version to notice anything about macros in menuselect! As I said, it is not hard to convert from macro to gosub, once I had worked out why my dialplan wasn't working. -- Best regards, Julianmailto:jb_s...@trink.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] clarification on gosub, macros and AEL
Hello Doug, Tuesday, October 15, 2019, 5:07:45 PM, you wrote: > Personally, I don't think MACROS are going anywhere any time soon, > so I have not bothered looking into a substitution. Haven't they gone in Asterisk 16? I've just upgraded from Asterisk 13 and had to do some hasty re-writing of my dialplan to convert the Macros into Gosubs because the Macros did not work (I had noticed the deprecation notice previously, but forgot about it). I didn't have many, so I did them by hand, but it would be easy to script. Julian -- Best regards, Julianmailto:jb_s...@trink.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
Hello Thelma, Friday, February 16, 2018, 2:16:02 AM, you wrote: > Contact: "sip:pstn-" > And it found in sip.conf only: > Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 > Is perhaps the name effected by the special character "-" (dash) that is > why it only matches "pstn" and take the first one it found. Will it > make a difference if I rename the port to pstn_ in configuration files. If the type=friend then it matches on IP Address and Port Number, not the user name. It will then use the first entry in the sip.conf - it does not take any notice of the name. If you change the order that the two entries appear, all the calls will appear to come from [pstn-9998] even if they come from [pstn-]. I used to set user=peer, which solved the problem for me, but I now direct all the calls to a single context in extensions.conf and then send them to their own contexts based on the DNID. -- Best regards, Julianmailto:jb_s...@trink.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Centos 7 & Sangoma Wanpipe
Hello, Following the recent discussions about CentOS, I am interested to hear whether anyone is successfully running Sangoma's Wanpipe (for an analog telephony card) using Centos 7? Upgrades of Wanpipe have been tricky at best in the past, and I don't want to make the leap to CentOS 7 without knowing whether Wanpipe will operate correctly. It looks like it should work for recent kernel versions, so it might be OK. Thanks! -- Best regards, Julian mailto:jb_s...@trink.co.uk-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunks going to the wrong context
On Thursday, December 14, 2017, 10:05:23 PM, Tony Mountifield (t...@softins.co.uk) wrote: > So I think you really do need to have a single peer section for all sipgate > calls, pointing to one sipgate context in your dialplan that contains all > your various extensions like se2489, sj0151, etc. That is what I do - all my incoming calls (which originate from the same IP and Port) go into a single context in extensions.conf, from where they are directed into individual call handlers depending on the DCID [incoming calls] exten => 4420,n,Goto(handler-a,s,1) exten => 4400,n,Goto(handler-b,s,1) exten => 4421,n,Goto(handlet-c,s,1) exten => 4422,n,Goto(handler d,s,1) -- Best regards, Julianmailto:jb_s...@trink.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wanpipe 7 Driver problem with Debian 8.5 (Sangoma B600DE)
Hello Guillermo, On Tuesday, September 13, 2016, 7:10:29 PM, you wrote: > I have tried to install Wanpipe 7.0.20 in a fresh-installed Debian > 8.5 (Jessie) server, but when I reach to the point where I have to > enter the Linux Headers directory (installed in default with > “apt-get install linux-headers-$(uname -r)”), I get an error, the > package refuses to compile and installation aborts. Wanpipe installs fill me with dread! Sangoma say that they only support CentOS 6.x installs, and after many struggles to get Wanpipe to work on Fedora versions, that is what I run it on (CentOS 6.7). Even then, install issues are not uncommon, and it seems to be particularly sensitive to kernel issues - I avoid kernel updates on my Asterisk box! I am currently running with kernel 2.6.32. Sangoma says that Wanpipe will work on 4.3 kernels, but if your kernel is newer than 4.3 (November 2015) then that might be your problem. "setup_drv_compile.log" is the place to look for clues. You should find it in the directory where your Wanpipe source files are. -- Best regards, Julianmailto:jb_s...@trink.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Including doesn't have any effect
Title: Re: [asterisk-users] Including doesn't have any effect Hello Frank, Monday, June 6, 2016, 9:46:47 PM, you wrote: > As far as I know, Asterisk's database/blacklist function only supports > exact match of caller ID. > If you want to block a specific area code or a block of numbers (eg. > 321-654-8XXX) the blacklist function is useless. Ah, possibly. I only need to block specific incoming numbers. Julian -- Best regards, Julian mailto: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] Including doesn't have any effect
Title: Re: [asterisk-users] Including doesn't have any effect
On Monday, June 6, 2016, 4:55:12 PM, AJS wrote:
> Are you lazy enough to edit a text file and reload your dialplan, *every single
> time* someone calls you, that you don't want to have to speak to ever again?
> Not sure about you, but that sounds way too much like hard work for me! :)
Indeed, too much for me too...
exten => s,n,GotoIf(${DB_EXISTS(blacklist/${CDR(src)})}?block) ; Check whether caller blacklisted
[snip additional logging]
exten => s,n,Dial(SIP/phones,30)
exten => s,n,Voicemail(x@work,su)
exten => s,n,Hangup()
exten => s,n(block),GoSub(Handler-MarketingCall,s,1) ; deal with blacklisted callers
[snip]
This is linked to couple of simple routines that allow me to add the current (##666) or last caller (*32) to the blacklist or manually input any other number (*30).
--
Best regards,
Julian mailto:jb_s...@trink.co.uk
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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 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
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, 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, 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 calls might come in from. so I simplified the setup. I now have just one context in sip.conf to handle incoming 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 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] Patched Res_Musiconhold.So module
Hello Daniel, Wednesday, November 25, 2015, 7:02:08 PM, you wrote: > It still asks for a client certificate. > See this screen shot, hopefully it showswhat I mean. > http://firestar-hosting.com/clientcert.png It is probably a Keychain issue, and not a problem with the Issue Tracker website. Have a look at http://thedesignspace.net/MT2archives/001022.html for some instructions on resolving it. -- 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] Call forwarding in Asterisk
Hello Kantharuban, Friday, September 4, 2015, 8:19:28 AM, you wrote: > Thanks for your info, What is the impact of the following line in > dialpla Dial(SIP/19201/19202,300) It does not look like a valid format. If you are trying to dial two SIP devices (19201 and 19202) with a timeout of 300 seconds, the command would be Dial(SIP/19201/19202,300) and you might want to consider some of the option Dial options depending on what you do with the call after it has been answered. Have a look at http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial for details of the dial command, and the options or have a look at Asterisk: The Definitive Guide which will tell you more about Originate and Local Channels, which you might also find useful. http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/index.html J -- 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] Asterisk prefix code to dial a high fraud country - security mechanism
Hello motty,
Thursday, September 18, 2014, 6:35:40 PM, you wrote:
Hello, I would to allow users to place calls overseas such as India
and Malaysia but only with a security code. if they don't have a
security code I want to be able to drop the calls.
I use this
exten = _0041,1,Log(NOTICE,Pin Code for Switzerland calls)
same = n,Playback(silence/1)
same = n,Authenticate(9084,,4)
same = n,Macro(outgoingTrunk,${EXTEN})
same = n,Hangup()
It uses a fixed PIN number which calls a macro which deals with the
actual dialling, but a standard Dial command would work here too.
Quick and easy, but there are lots of options. If the correct PIN is
not entered, the call is not made.
--
Best regards,
Julianmailto:jb_s...@trink.co.uk
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Re: [asterisk-users] gsm codec compile
Hello Doug, Monday, March 3, 2014, 7:13:52 PM, you wrote: I was successful in compiling asterisk in raspbien except for the following error If I enable the gsm codec. It appears there is something in the Makefile n this directory that needs to be changed. Probably involving optimization. Not sure why it does not recognize the processor since it is one that is mentioned in the Makefile. Any help would be appreciated. [snip] Error: unknown architecture `armv6l' [snip] Here are the lines in the Makefile - ifeq (, $(findstring $(shell uname -m) , ppc ppc64 alpha armv4l arm5b armv5b armv61 armv7l s390 )) ^^ I am not sure whether it is part of the problem, but the architecture on the machine appears to be armv6l (lower case L) whilst the Makefield has armv61 (the number one). A typo in the Makefile? -- 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] iax2: two users can't authenticate from same ip address
Hello Sean, Sunday, September 8, 2013, 11:25:24 PM, you wrote: The problem is that once a phone has used the server, no other phone can use it. The servers sees all the phones as having the same ip address (though different ports). This sounds like the Peer v Friend problem I have had in the past. Try setting user=friend which will match on the username and not IP address. I found that asterisk was matching to the first account in the list in IAX.CONF and authentication was then failing (or in the case of incoming calls, ending up in the wrong context). http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer 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
