Re: [asterisk-users] Asterisk 14.0.2 opens a high numbered UDP port
On Thu, Oct 13, 2016 at 12:35 PM, Brandon B. wrote: > What part of Asterisk 14.0.2 opens the random, high numbered (33094 > currently) UDP port? This port is opened even without any channel drivers > loaded. > > $ sudo netstat -ltunp | grep asterisk > udp0 0 0.0.0.0:51488 0.0.0.0:* > 13830/asterisk > udp0 0 0.0.0.0:5060 0.0.0.0:* > 13830/asterisk > udp0 0 :::42516 :::* > 13830/asterisk > > > Those ports are used by the underlying pjproject DNS resolver. The resolver is always listening on those ports for DNS query responses. 1 for IPV4 and 1 for IPV6. Your firewall should only be allowing responses to flow through to those ports that match outgoing requests. > > -- > _ > -- 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 > -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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] Asterisk 13.11.2 unable to register on Centos 7 64bit
Ok. Please, note that 192.168.1.37 (I suspect) is the internal LAN address Of the Polycom hardphone. If this is true, then you have NAT issues. The REGISTER message are received by your PBX, but when respond, Asterisk send the next SIP message to the IP informed by the phone, that is the internal LAN address. The messages do not reach back to the hardphone. You need to setup a STUN server in the Polycom hardphone settings. Please, check the manual. Search in Google some public STUN server to put in the settings. Last, the idea behind the "sip set debug" command was view the complete SIP messages conversation, not search for an error. On NAT escenarios, remember: * The NATed phones need to know the public IP of the NATing router. Either by manual setting or by STUN protocol. * Reduce the time between REGISTERs attempt, if the client have a dynamic IP connection. * Use the "localnet" SIP settings in Asterisk, so the PBX can distingish what Network need contacted via NAT and what not. Cheers. -- _ -- 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] Asterisk 13.11.2 unable to register on Centos 7 64bit
Hello Victor, I did set debug on, but I don’t see any errors. I did tcpdump, client is trying to register: here is the header of a udp packet User Datagram Protocol, Src Port: 55300, Dst Port: 5060 Session Initiation Protocol (REGISTER) Request-Line: REGISTER sip:pbx.mydomain.com:5060 SIP/2.0 Method: REGISTER Request-URI: sip:pbx.mydomain.com:5060 [Resent Packet: True] [Suspected resend of frame: 14] Message Header Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK7b2855394DB988BE Transport: UDP Sent-by Address: 192.168.1.37 Sent-by port: 5060 Branch: z9hG4bK7b2855394DB988BE From: "1006" ;tag=2859342B-CBC71460 SIP Display info: "1006" SIP from address: sip:1...@pbx.mydomain.com SIP from tag: 2859342B-CBC71460 To: SIP to address: sip:1...@pbx.mydomain.com SIP to address User Part: 1006 SIP to address Host Part: pbx.mydomain.com CSeq: 1 REGISTER Call-ID: 6cbe37bb-cca69d70-85d0431d@192.168.1.37 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.10.0689 Accept-Language: en Max-Forwards: 70 Expires: 90 Content-Length: 0 Sip.conf [1006] type=friend username=1006 secret=mysecret context=sip-phone call-limit=5 callerid="iuser" <1006> disallow=all host=dynamic allow=all nat=yes Is NAT value set to yes OK? Servers is on public IP, client is on private network. Thanks, Motty From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Victor Villarreal Sent: Thursday, October 13, 2016 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit Hi Motty, Please, set Verbose to 3 and Debug to 3 At Asterisk CLI. Then "sip set debug on". Now try to register again. At last, " sip de debug off". Examine tour console or full log file to find some clue ir send me back some trace. Cheers. El oct. 13, 2016 1:45 PM, "Motty Cruz" escribió: Hello, fresh install of Asterisk 13.11.2, client unable to register. For now I have IPtables disabled, also selinux is disabled [1006] type=friend username=1006 secret=mysecret context=sip-phone call-limit=1 callerid="iuser" <1006> disallow=all host=dynamic allow=all any ideas? Thanks, Motty -- _ -- 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 -- _ -- 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] Openfile Issue
On Thu, 13 Oct 2016, Dovid Bender wrote: 50771 is the PID. I am talking about the user. for instances if running as root (which you should never do) then: lsof -u root | wc -l Wouldn't sudo lsof -c asterisk | wc --lines be more direct and accurate? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- 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
[asterisk-users] Asterisk 14.0.2 opens a high numbered UDP port
What part of Asterisk 14.0.2 opens the random, high numbered (33094 currently) UDP port? This port is opened even without any channel drivers loaded. $ sudo netstat -ltunp | grep asterisk udp0 0 0.0.0.0:51488 0.0.0.0:* 13830/asterisk udp0 0 0.0.0.0:5060 0.0.0.0:* 13830/asterisk udp0 0 :::42516 :::* 13830/asterisk -- _ -- 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] Openfile Issue
50771 is the PID. I am talking about the user. for instances if running as root (which you should never do) then: lsof -u root | wc -l On Thu, Oct 13, 2016 at 1:31 PM, Ahmed Munir wrote: > > [root@abc asterisk]# lsof -u 50771 | wc -l > 0 > > BTW, I'm using CentOS 6.5 > > > >> >> Date: Thu, 13 Oct 2016 10:20:19 -0400 >>> From: Dovid Bender >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> >>> Subject: Re: [asterisk-users] Openfile Issue >>> Message-ID: >>> >> ail.com> >>> Content-Type: text/plain; charset="utf-8" >>> >>> Strange. What's the output of: >>> lsof -u | wc -l >>> >>> I know that on some CentOS7 setups we needed to remove the >>> file 90-nproc.conf as well as add to the Asterisk init script: >>> ulimit -s unlimited >>> ulimit -n 65535 >>> ulimit -Hn 65535 >>> ulimit -u 65535 >>> ulimit -Hu 65535 >>> >>> >>> >>> On Thu, Oct 13, 2016 at 9:59 AM, Ahmed Munir >>> wrote: >>> >>> > >>> > See below; >>> > >>> > [root@abc asterisk]# cat /proc/50771/limits >>> > Limit Soft Limit Hard Limit >>> Units >>> > Max cpu time unlimitedunlimited >>> seconds >>> > Max file size unlimitedunlimited >>> bytes >>> > Max data size unlimitedunlimited >>> bytes >>> > Max stack size10485760 unlimited >>> bytes >>> > Max core file sizeunlimitedunlimited >>> bytes >>> > Max resident set unlimitedunlimited >>> bytes >>> > Max processes 256389 256389 >>> > processes >>> > Max open files225000 >>> files >>> > Max locked memory 6553665536 >>> bytes >>> > Max address space unlimitedunlimited >>> bytes >>> > Max file locksunlimitedunlimited >>> locks >>> > Max pending signals 256389 256389 >>> signals >>> > Max msgqueue size 819200 819200 >>> bytes >>> > Max nice priority 00 >>> > Max realtime priority 00 >>> > Max realtime timeout unlimitedunlimitedus >>> > >>> > >>> > Date: Thu, 13 Oct 2016 09:37:34 -0400 >>> >> From: Dovid Bender >>> >> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> >> >>> >> Subject: Re: [asterisk-users] Openfile Issue >>> >> Message-ID: >>> >> >> >> ail.com> >>> >> Content-Type: text/plain; charset="utf-8" >>> >> >>> >> What do you get when you do: >>> >> cat /proc//limits ? >>> >> >>> >> On Thu, Oct 13, 2016 at 9:30 AM, Ahmed Munir >> > >>> >> wrote: >>> >> >>> >> -- > Regards, > > Ahmed Munir Chohan > > > -- > _ > -- 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 > -- _ -- 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] Openfile Issue
[root@abc asterisk]# lsof -u 50771 | wc -l 0 BTW, I'm using CentOS 6.5 > > Date: Thu, 13 Oct 2016 10:20:19 -0400 >> From: Dovid Bender >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> >> Subject: Re: [asterisk-users] Openfile Issue >> Message-ID: >> > gmail.com> >> Content-Type: text/plain; charset="utf-8" >> >> Strange. What's the output of: >> lsof -u | wc -l >> >> I know that on some CentOS7 setups we needed to remove the >> file 90-nproc.conf as well as add to the Asterisk init script: >> ulimit -s unlimited >> ulimit -n 65535 >> ulimit -Hn 65535 >> ulimit -u 65535 >> ulimit -Hu 65535 >> >> >> >> On Thu, Oct 13, 2016 at 9:59 AM, Ahmed Munir >> wrote: >> >> > >> > See below; >> > >> > [root@abc asterisk]# cat /proc/50771/limits >> > Limit Soft Limit Hard Limit >> Units >> > Max cpu time unlimitedunlimited >> seconds >> > Max file size unlimitedunlimited >> bytes >> > Max data size unlimitedunlimited >> bytes >> > Max stack size10485760 unlimited >> bytes >> > Max core file sizeunlimitedunlimited >> bytes >> > Max resident set unlimitedunlimited >> bytes >> > Max processes 256389 256389 >> > processes >> > Max open files225000 >> files >> > Max locked memory 6553665536 >> bytes >> > Max address space unlimitedunlimited >> bytes >> > Max file locksunlimitedunlimited >> locks >> > Max pending signals 256389 256389 >> signals >> > Max msgqueue size 819200 819200 >> bytes >> > Max nice priority 00 >> > Max realtime priority 00 >> > Max realtime timeout unlimitedunlimitedus >> > >> > >> > Date: Thu, 13 Oct 2016 09:37:34 -0400 >> >> From: Dovid Bender >> >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> >> >> >> Subject: Re: [asterisk-users] Openfile Issue >> >> Message-ID: >> >> > >> ail.com> >> >> Content-Type: text/plain; charset="utf-8" >> >> >> >> What do you get when you do: >> >> cat /proc//limits ? >> >> >> >> On Thu, Oct 13, 2016 at 9:30 AM, Ahmed Munir >> >> wrote: >> >> >> > -- Regards, Ahmed Munir Chohan -- _ -- 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] Asterisk 13.11.2 unable to register on Centos 7 64bit
Hi Motty, Please, set Verbose to 3 and Debug to 3 At Asterisk CLI. Then "sip set debug on". Now try to register again. At last, " sip de debug off". Examine tour console or full log file to find some clue ir send me back some trace. Cheers. El oct. 13, 2016 1:45 PM, "Motty Cruz" escribió: > Hello, fresh install of Asterisk 13.11.2, client unable to register. For > now I have IPtables disabled, also selinux is disabled > > > > [1006] > > type=friend > > username=1006 > > secret=mysecret > > context=sip-phone > > call-limit=1 > > callerid="iuser" <1006> > > disallow=all > > host=dynamic > > allow=all > > > > any ideas? > > > > Thanks, > > Motty > > -- > _ > -- 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 > -- _ -- 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] Asterisk inside network. What phone works well?
I think you had asked what phone works well with VPN's. I've had very good experiences with Yealink using OpenVPN, never an issue. I think I've heard that Snom does OpenVPN as well. Mark -- _ -- 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
[asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit
Hello, fresh install of Asterisk 13.11.2, client unable to register. For now I have IPtables disabled, also selinux is disabled [1006] type=friend username=1006 secret=mysecret context=sip-phone call-limit=1 callerid="iuser" <1006> disallow=all host=dynamic allow=all any ideas? Thanks, Motty -- _ -- 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
[asterisk-users] Wildcard AEX800 digium card asterisk configuration
hello, i recently purchased a Wildcard AEX800 digium card. Ive installed asterisk 13 and all prerequistses on ubuntu serv14.04 LTS. Dahi is the driver am using; ive configured all but when i call from PSTN through fxo port an not getting anything in logs or to extensions. below are my config please help system.conf loadzone = za defaultzone =za fxsks=1 echocanceller=mg2,1 fxsks=2 echocanceller=mg2,2 fxsks=3 echocanceller=mg2,3 fxsks=4 echocanceller=mg2,4 fxoks=5 echocanceller=mg2,5 fxoks=6 echocanceller=mg2,6 fxoks=7 echocanceller=mg2,7 fxoks=8 echocanceller=mg2,8 chan dahdi.conf [trunkgroups] ; No trunk groups are needed in this configuration. [channels] #include /etc/asterisk/dahdi-channels.conf ; The channels context is used when defining channels using the ; older deprecated method. Don't use this as a section name. ;[phone](!) ; ; A template to hold common options for all phones. ; usecallerid = yes hidecallerid = no callwaiting = yes threewaycalling = yes transfer = yes echocancel = yes echocancelwhenbridged = yes ;immediate = no rxgain = 0.0 txgain = 0.0 ;FXS Modules group = 1 echocancel = yes signalling = fxo_ks context = Internal channel = 1-4 ;FXO Modules group = 2 echocancel = yes signalling = fxs_ks context = Incoming channel = 5-8 voicemail.conf [default] 1234 => 4242,chris kamutumwa,root@localhost 1000 => 1234,chris kamutumwa,chriskamutu...@gmail.com 2000 => 1234,chris utumwa,ch...@crystaline.co.zm ~ ~ extension.conf [Internal] exten => 1000,1,Dial (DAHDI/1,20,rt) exten => 1000,2,Voicemail (1000,u) exten => 1000,102,Voicemail (1000,b) exten => 2000,1,Dial (DAHDI/2,20,rt) exten => 2000,2,Voicemail (2000,u) exten => 2000,102,Voicemail (2000,b) exten => 8500,1,VoiceMailMain exten => 8501,1,MusicOnHold exten => _9.,1,Dail (DAHDI/g2/www${EXTEN:1} ) exten => _9.,2,Congestion [Incoming] exten => s,1,Answer exten => s,2,Dial(DAHDI/g1,20,rt) exten => s,3,Voicemail(1000,u) exten => s,103,Voicemail(1000,b) root@ubuntu:/etc/asterisk# lsdahdi ### Span 1: WCTDM/0 "Wildcard AEX800" (MASTER) 1 FXO RED 2 FXO RED 3 FXO RED 4 FXO RED 5 FXS 6 FXS 7 FXS 8 FXS root@ubuntu:/etc/asterisk# dahdi_cfg -vvv DAHDI Tools Version - 2.11.1 DAHDI Version: 2.11.1 Echo Canceller(s): Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 07: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 07) Channel 08: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 08) 8 channels to configure. Setting echocan for channel 1 to mg2 Setting echocan for channel 2 to mg2 Setting echocan for channel 3 to mg2 Setting echocan for channel 4 to mg2 Setting echocan for channel 5 to mg2 Setting echocan for channel 6 to mg2 Setting echocan for channel 7 to mg2 Setting echocan for channel 8 to mg2 -- _ -- 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] Asterisk inside network. What phone works well?
> I have Asterisk running well inside our network. I did some > experiments exposing it to internet but had some issues: > 1. NAT issues (voice one way, etc). From what I understand double- > NAT users will always have something like this > 2. Immediately I see people trying to hack into. I did configure > Fail2Ban and it works somewhat, but not 100%. Erroneous logs, etc > > So.. I ended up closing network. Currently most users inside > network. My home router have GRE tunnel to office so phone works just fine. > Another user uses VPN and soft phone and it works good too. > > Now I need to setup some users with actual phone devices and none of > those solutions will work. So, I did some research and found > that some phones have VPN capability built in. > > Right now I use Cisco SPA504G phones. We have auto-provisioning for > them, works well. But I don’t think they have VPN capability. > > > What I found it that Cisco 525g2 has AnyConnect functionality (SSL > VPN) but not sure if this is what I need. > > We have Mikrotik router. Can I setup VPN on router and have this > Cisco phone auto-dial VPN and then connect to Asterisk? I’m asking > to see if this will work before I go in and buy that phone. > Or maybe there is other devices/solutions you suggest? I’d like to > stay with Cisco because I’m somewhat familiar with provisioning those.. I haven't done this myself, but I think what you need to look at is phones that can do IPSEC vpn setups. For the Mikrotik router, this may be helpful to start investigating: http://wiki.mikrotik.com/wiki/L2TP_%2B_IPSEC_between_Mikrotik_router_and_a_PC __ This email has been scanned by the Symantec Email Security.cloud service. For more information please visit http://www.symanteccloud.com __ -- _ -- 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
[asterisk-users] Asterisk inside network. What phone works well?
Hello list, I have Asterisk running well inside our network. I did some experiments exposing it to internet but had some issues: 1. NAT issues (voice one way, etc). From what I understand double-NAT users will always have something like this 2. Immediately I see people trying to hack into. I did configure Fail2Ban and it works somewhat, but not 100%. Erroneous logs, etc So.. I ended up closing network. Currently most users inside network. My home router have GRE tunnel to office so phone works just fine. Another user uses VPN and soft phone and it works good too. Now I need to setup some users with actual phone devices and none of those solutions will work. So, I did some research and found that some phones have VPN capability built in. Right now I use Cisco SPA504G phones. We have auto-provisioning for them, works well. But I don’t think they have VPN capability. What I found it that Cisco 525g2 has AnyConnect functionality (SSL VPN) but not sure if this is what I need. We have Mikrotik router. Can I setup VPN on router and have this Cisco phone auto-dial VPN and then connect to Asterisk? I’m asking to see if this will work before I go in and buy that phone. Or maybe there is other devices/solutions you suggest? I’d like to stay with Cisco because I’m somewhat familiar with provisioning those.. Thank you Ivan -- _ -- 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] Openfile Issue
acl.c: Cannot create >> socket >> > >> > [2016-10-13 08:18:40] ERROR[49913][C-05b7] cdr_csv.c: Unable to >> > re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many >> > open files >> > >> > [2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket >> > >> > [2016-10-13 08:18:40] ERROR[49833][C-059c] cdr_csv.c: Unable to >> > re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many >> > open files >> > >> > [2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket >> > >> > [2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket >> > >> > Further added, I'm using CentOS 6.5 as OS. >> > >> > Please advise what changes required for permanently fixing this random >> > issue. >> > >> > >> > -- >> > Regards, >> > >> > Ahmed Munir Chohan >> > >> > >> > -- >> > _ >> > -- 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 >> > >> -- next part -- >> An HTML attachment was scrubbed... >> URL: <http://lists.digium.com/pipermail/asterisk-users/attachment >> s/20161013/dd80ddc4/attachment.html> >> >> -- >> >> ___ >> --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 >> >> End of asterisk-users Digest, Vol 147, Issue 11 >> *** >> > > > > -- > Regards, > > Ahmed Munir Chohan > > > -- > _ > -- 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 > -- _ -- 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] Openfile Issue
ently fixing this random > > issue. > > > > > > -- > > Regards, > > > > Ahmed Munir Chohan > > > > > > -- > > _____ > > -- 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 > > > -- next part -- > An HTML attachment was scrubbed... > URL: <http://lists.digium.com/pipermail/asterisk-users/attachment > s/20161013/dd80ddc4/attachment.html> > > -- > > ___ > --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 > > End of asterisk-users Digest, Vol 147, Issue 11 > *** > -- Regards, Ahmed Munir Chohan -- _ -- 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] Openfile Issue
What do you get when you do: cat /proc//limits ? On Thu, Oct 13, 2016 at 9:30 AM, Ahmed Munir wrote: > Hi all, > > Now a days getting openfile issues on asterisk quite often even setting > system soft limit to 2 and hard limit to 25000 and issue usually > occurs during openfile socket consumed by system and asterisk is quite > smaller than the soft or hard limit. See below system and asterisk logs; > > 2016:10:13_08:19:01 | Too many LOG file moved successfully - messages > > 2016:10:13_08:19:01 | Asterisk openfile count: 1252 > > 2016:10:13_08:19:01 | Total system open files count: 4091 > > 2016:10:13_08:19:01 | SIP channels: 93 active calls|1563 calls processed > > 2016:10:13_08:19:01 | Asterisk SIP peers: 366 > > 2016:10:13_08:19:01 | Asterisk service uptime: System uptime: 4 days, 4 > hours, 12 minutes, 33 seconds > > Last reload: 4 days, 4 hours, 12 minutes, 33 seconds > > Privilege escalation protection disabled! > > See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. > > 2016:10:13_08:19:01 | Socket Summary > > Total: 648 (kernel 758) > > TCP: 20 (estab 8, closed 5, orphaned 0, synrecv 0, timewait 1/0), ports > 14 > > > > Transport Total IPIPv6 > > * 758 - - > > RAW 0 0 0 > > UDP 422 419 3 > > TCP 15141 > > INET 437 433 4 > > FRAG 0 0 0 > > 2016:10:13_08:19:01 | Logged successfully all the required details > > > [2016-10-13 08:18:41] ERROR[50690][C-062a] res_timing_timerfd.c: > Failed to create timerfd timer: Too many open files > > [2016-10-13 08:18:41] ERROR[50690][C-062a] acl.c: Cannot create socket > > [2016-10-13 08:18:41] ERROR[50690][C-062a] res_timing_timerfd.c: > Failed to create timerfd timer: Too many open files > > [2016-10-13 08:18:41] ERROR[50690][C-062a] acl.c: Cannot create socket > > [2016-10-13 08:18:41] ERROR[50689][C-0629] res_timing_timerfd.c: > Failed to create timerfd timer: Too many open files > > [2016-10-13 08:18:41] ERROR[50689][C-0629] acl.c: Cannot create socket > > [2016-10-13 08:18:40] ERROR[49913][C-05b7] cdr_csv.c: Unable to > re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many > open files > > [2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket > > [2016-10-13 08:18:40] ERROR[49833][C-059c] cdr_csv.c: Unable to > re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many > open files > > [2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket > > [2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket > > Further added, I'm using CentOS 6.5 as OS. > > Please advise what changes required for permanently fixing this random > issue. > > > -- > Regards, > > Ahmed Munir Chohan > > > -- > _ > -- 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 > -- _ -- 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
[asterisk-users] Openfile Issue
Hi all, Now a days getting openfile issues on asterisk quite often even setting system soft limit to 2 and hard limit to 25000 and issue usually occurs during openfile socket consumed by system and asterisk is quite smaller than the soft or hard limit. See below system and asterisk logs; 2016:10:13_08:19:01 | Too many LOG file moved successfully - messages 2016:10:13_08:19:01 | Asterisk openfile count: 1252 2016:10:13_08:19:01 | Total system open files count: 4091 2016:10:13_08:19:01 | SIP channels: 93 active calls|1563 calls processed 2016:10:13_08:19:01 | Asterisk SIP peers: 366 2016:10:13_08:19:01 | Asterisk service uptime: System uptime: 4 days, 4 hours, 12 minutes, 33 seconds Last reload: 4 days, 4 hours, 12 minutes, 33 seconds Privilege escalation protection disabled! See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. 2016:10:13_08:19:01 | Socket Summary Total: 648 (kernel 758) TCP: 20 (estab 8, closed 5, orphaned 0, synrecv 0, timewait 1/0), ports 14 Transport Total IPIPv6 * 758 - - RAW 0 0 0 UDP 422 419 3 TCP 15141 INET 437 433 4 FRAG 0 0 0 2016:10:13_08:19:01 | Logged successfully all the required details [2016-10-13 08:18:41] ERROR[50690][C-062a] res_timing_timerfd.c: Failed to create timerfd timer: Too many open files [2016-10-13 08:18:41] ERROR[50690][C-062a] acl.c: Cannot create socket [2016-10-13 08:18:41] ERROR[50690][C-062a] res_timing_timerfd.c: Failed to create timerfd timer: Too many open files [2016-10-13 08:18:41] ERROR[50690][C-062a] acl.c: Cannot create socket [2016-10-13 08:18:41] ERROR[50689][C-0629] res_timing_timerfd.c: Failed to create timerfd timer: Too many open files [2016-10-13 08:18:41] ERROR[50689][C-0629] acl.c: Cannot create socket [2016-10-13 08:18:40] ERROR[49913][C-05b7] cdr_csv.c: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many open files [2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket [2016-10-13 08:18:40] ERROR[49833][C-059c] cdr_csv.c: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many open files [2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket [2016-10-13 08:18:40] ERROR[2983] acl.c: Cannot create socket Further added, I'm using CentOS 6.5 as OS. Please advise what changes required for permanently fixing this random issue. -- Regards, Ahmed Munir Chohan -- _ -- 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] Match one OR two digit extension not working as expected without using "dangerous" _. pattern (Ast 14)
In article , Jonathan H wrote: > Back to basics here. I want to match on one OR two digits. > > The following two both work, but ONLY for more than one digit, which > is not as expected from the docs (see below). > > exten => _X.,1,SayNumber(${EXTEN}) > exten => _[0-9].,1,SayNumber(${EXTEN}) > > > This next one will ONLY match 2 digits, as expected, but the first two > SHOULD match one or more, right? > > exten => _XX,1,SayNumber(${EXTEN}) > > The following pattern works, but I thought it was "dangerous" and to > be discouraged? > exten => _.,1,SayNumber(${EXTEN}) > > So, again, if someone dials 1 and a one second delay passes, I want it to say > 1. > If someone dials 1 then another 1 within a second then I want it to be > 11, and 111 should be invalid. > > (I've Set(TIMEOUT(digit)=1) ) > > Yes, I can do this with multiple lines, but the docs suggest this > should be easily do-able in 1 line, and I don't want to double the > amount of dialplan (there'll be a few of these!). When matching an extension being dialled, Asterisk is only concerned about priority 1, so that's the only priority you need to double. You should be able to use ! safely in priority 2 upwards: exten => _X,1,NoOp(Matching single digit) exten => _X.,1,NoOp(Matching multiple digits) exten => _X!,2,SayNumber(${EXTEN}) exten => _X!,3,Etc.. Disclaimer: I haven't tested this. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- 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] Softphone with by-default AutoAnswer active?
On Thu, Oct 13, 2016 at 12:19 PM, Mandar Khire wrote: > Hi, > I know that This is not Asterisk related question but then also I ask this > question here due to Asterisk users know about softphones & here lots of > user present. > > Question:- > > I am looking for Softphone which work on Windows platform. > > Softphone must have 'default AutoAnswer on'. > > Means example:- When I install softphone, it does not have any SIP > registration. But it has some by-default settings. > > In that I am looking active AutoAnswer option. > > I tried Ekiga, Linphone, mizuphone, x-lite etc but all have default > AutoAnswer off. > > I have to click on settings & active it. > > So all these softphones not useful for me. > > Need help. > Thanks, > Mandar P. Khire > +919769419340 > > > -- > _ > -- 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 > >From what I remember Bria allows this either by a setting on the OS or via a configuration server which provisions the settings on login -- _ -- 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
[asterisk-users] Softphone with by-default AutoAnswer active?
Hi, I know that This is not Asterisk related question but then also I ask this question here due to Asterisk users know about softphones & here lots of user present. Question:- I am looking for Softphone which work on Windows platform. Softphone must have 'default AutoAnswer on'. Means example:- When I install softphone, it does not have any SIP registration. But it has some by-default settings. In that I am looking active AutoAnswer option. I tried Ekiga, Linphone, mizuphone, x-lite etc but all have default AutoAnswer off. I have to click on settings & active it. So all these softphones not useful for me. Need help. Thanks, Mandar P. Khire +919769419340 -- _ -- 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] Match one OR two digit extension not working as expected without using "dangerous" _. pattern (Ast 14)
Sorry, I should have said, I already tried "!". It matches immediately and doesn't wait for a second digit. On 13 October 2016 at 12:05, Jean Aunis wrote: > You can use the "!" character : > > exten => _X!,1,SayNumber(${EXTEN}) > > > Best regards > > Jean Aunis > > > Le 13/10/2016 à 12:54, Jonathan H a écrit : >> >> Back to basics here. I want to match on one OR two digits. >> >> The following two both work, but ONLY for more than one digit, which >> is not as expected from the docs (see below). >> >> exten => _X.,1,SayNumber(${EXTEN}) >> exten => _[0-9].,1,SayNumber(${EXTEN}) >> >> >> This next one will ONLY match 2 digits, as expected, but the first two >> SHOULD match one or more, right? >> >> exten => _XX,1,SayNumber(${EXTEN}) >> >> The following pattern works, but I thought it was "dangerous" and to >> be discouraged? >> exten => _.,1,SayNumber(${EXTEN}) >> >> So, again, if someone dials 1 and a one second delay passes, I want it to >> say 1. >> If someone dials 1 then another 1 within a second then I want it to be >> 11, and 111 should be invalid. >> >> (I've Set(TIMEOUT(digit)=1) ) >> >> Yes, I can do this with multiple lines, but the docs suggest this >> should be easily do-able in 1 line, and I don't want to double the >> amount of dialplan (there'll be a few of these!). >> >> Here are my references: >> >> --- >> >> https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching >> >> The letter X or x represents a single digit from 0 to 9. >> The period character (.) at the end of a pattern matches one or more >> remaining characters. You put it at the end of a pattern when you want >> to match extensions of an indeterminate length. >> >> --- >> >> Page 141 of the Asterisk Definitive Guide 4th Edition: >> >> . (period) >> Wildcard match; matches one or more characters, no matter what they are. >> If you’re not careful, wildcard matches can make your dialplans do >> things you’re not expecting (like matching built-in extensions such >> as i or h). You should use the wildcard match in a pattern only after >> you’ve matched as many other digits as possible. For example, the >> following pattern match should probably never be used: >> _. >> In fact, Asterisk will warn you if you try to use it. Instead, if you >> really need a catchall pattern match, use this one to match all strings >> that start with a digit followed by one or more characters (see ! if >> you want to be able to match on zero or more characters): >> _X. >> Or this one, to match any alphanumeric string: >> _[0-9a-zA-Z]. >> >> --- >> >> http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns >> Do not use a pattern of _. as this will match everything including >> Asterisk special extensions like i, t, h, etc. Instead use something >> like _X. or _X which will not match __special__ extensions.. >> So what do you use instead of _. ? Many examples use this construct, >> but if you use it you may see a warning message in the log advising >> you to change _. to _X. >> >> --- >> > > > -- > _ > -- 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 -- _ -- 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] Match one OR two digit extension not working as expected without using "dangerous" _. pattern (Ast 14)
You can use the "!" character : exten => _X!,1,SayNumber(${EXTEN}) Best regards Jean Aunis Le 13/10/2016 à 12:54, Jonathan H a écrit : Back to basics here. I want to match on one OR two digits. The following two both work, but ONLY for more than one digit, which is not as expected from the docs (see below). exten => _X.,1,SayNumber(${EXTEN}) exten => _[0-9].,1,SayNumber(${EXTEN}) This next one will ONLY match 2 digits, as expected, but the first two SHOULD match one or more, right? exten => _XX,1,SayNumber(${EXTEN}) The following pattern works, but I thought it was "dangerous" and to be discouraged? exten => _.,1,SayNumber(${EXTEN}) So, again, if someone dials 1 and a one second delay passes, I want it to say 1. If someone dials 1 then another 1 within a second then I want it to be 11, and 111 should be invalid. (I've Set(TIMEOUT(digit)=1) ) Yes, I can do this with multiple lines, but the docs suggest this should be easily do-able in 1 line, and I don't want to double the amount of dialplan (there'll be a few of these!). Here are my references: --- https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching The letter X or x represents a single digit from 0 to 9. The period character (.) at the end of a pattern matches one or more remaining characters. You put it at the end of a pattern when you want to match extensions of an indeterminate length. --- Page 141 of the Asterisk Definitive Guide 4th Edition: . (period) Wildcard match; matches one or more characters, no matter what they are. If you’re not careful, wildcard matches can make your dialplans do things you’re not expecting (like matching built-in extensions such as i or h). You should use the wildcard match in a pattern only after you’ve matched as many other digits as possible. For example, the following pattern match should probably never be used: _. In fact, Asterisk will warn you if you try to use it. Instead, if you really need a catchall pattern match, use this one to match all strings that start with a digit followed by one or more characters (see ! if you want to be able to match on zero or more characters): _X. Or this one, to match any alphanumeric string: _[0-9a-zA-Z]. --- http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns Do not use a pattern of _. as this will match everything including Asterisk special extensions like i, t, h, etc. Instead use something like _X. or _X which will not match __special__ extensions.. So what do you use instead of _. ? Many examples use this construct, but if you use it you may see a warning message in the log advising you to change _. to _X. --- -- _ -- 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
[asterisk-users] Match one OR two digit extension not working as expected without using "dangerous" _. pattern (Ast 14)
Back to basics here. I want to match on one OR two digits. The following two both work, but ONLY for more than one digit, which is not as expected from the docs (see below). exten => _X.,1,SayNumber(${EXTEN}) exten => _[0-9].,1,SayNumber(${EXTEN}) This next one will ONLY match 2 digits, as expected, but the first two SHOULD match one or more, right? exten => _XX,1,SayNumber(${EXTEN}) The following pattern works, but I thought it was "dangerous" and to be discouraged? exten => _.,1,SayNumber(${EXTEN}) So, again, if someone dials 1 and a one second delay passes, I want it to say 1. If someone dials 1 then another 1 within a second then I want it to be 11, and 111 should be invalid. (I've Set(TIMEOUT(digit)=1) ) Yes, I can do this with multiple lines, but the docs suggest this should be easily do-able in 1 line, and I don't want to double the amount of dialplan (there'll be a few of these!). Here are my references: --- https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching The letter X or x represents a single digit from 0 to 9. The period character (.) at the end of a pattern matches one or more remaining characters. You put it at the end of a pattern when you want to match extensions of an indeterminate length. --- Page 141 of the Asterisk Definitive Guide 4th Edition: . (period) Wildcard match; matches one or more characters, no matter what they are. If you’re not careful, wildcard matches can make your dialplans do things you’re not expecting (like matching built-in extensions such as i or h). You should use the wildcard match in a pattern only after you’ve matched as many other digits as possible. For example, the following pattern match should probably never be used: _. In fact, Asterisk will warn you if you try to use it. Instead, if you really need a catchall pattern match, use this one to match all strings that start with a digit followed by one or more characters (see ! if you want to be able to match on zero or more characters): _X. Or this one, to match any alphanumeric string: _[0-9a-zA-Z]. --- http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns Do not use a pattern of _. as this will match everything including Asterisk special extensions like i, t, h, etc. Instead use something like _X. or _X which will not match __special__ extensions.. So what do you use instead of _. ? Many examples use this construct, but if you use it you may see a warning message in the log advising you to change _. to _X. --- -- _ -- 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] Asterisk 12 error when installing
Are those numbers correct? Asterisk 12 stopped being supported almost 2 years ago and became "do not use" on 2015-12-20 https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions Ubuntu 14 may still be supported, if you're on 14.0.4.5 https://wiki.ubuntu.com/Releases You could try make distclean and start again. What happens when you compile with a current, supported version of Asterisk? On 13 October 2016 at 08:33, christopher kamutumwa wrote: > Hello, > > Am trying to install asterisk 12 on ubuntu 14.04lts and am getting the > below error after a MAKE any hints how to go round this? > > bedit.a -> asterisk > asterisk.o: file not recognized: File truncated > collect2: error: ld returned 1 exit status > make[1]: *** [asterisk] Error 1 > make: *** [main] Error 2 > > > regards > > chris > > -- > _ > -- 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 -- _ -- 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
[asterisk-users] Asterisk 12 error when installing
Hello, Am trying to install asterisk 12 on ubuntu 14.04lts and am getting the below error after a MAKE any hints how to go round this? bedit.a -> asterisk asterisk.o: file not recognized: File truncated collect2: error: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make: *** [main] Error 2 regards chris -- _ -- 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