Re: [asterisk-users] Polycom phones and ring no answer/302 Moved Temporarily
I have Polycom IP550. The "Forward" "No Answer" is working fine when enabled. I was looking at the sip.cfg but don't know exactly what to look for, can you give me a hint to where would i find that option? Thanks, On Wed, Dec 12, 2012 at 1:48 PM, Justin Sherrill < justin.sherr...@americanrocksalt.com> wrote: > I have several Polycom IP550 phones running UC 4.0.3, connected to > Asterisk 1.8. > > Setting forwarding for "Always" works as expected; the phone issues a 302 > Moved Temporarily, and Asterisk shifts the call to the new location. > > Setting forwarding to "No Answer" means a 302 never gets issued. It just > rings and eventually goes to voicemail. Watching with Wireshark, I never > see a 302 SIP message issued. I can't find anything in the phone settings > that look like it would disable this. > > Anyone else with a Polycom set that sees this, or does not see this and > has "forward no answer" working? > > Justin Sherrill - American Rock Salt > P: 585-991-6825 F: 585-991-6925 C: 585-298-6826 > > > -- > _ > -- 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 > -- _ -- 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
[asterisk-users] asterisk 1.8.10.1 meetme
Hello, I'm running Asterisk 1.8.10 on Linux box, when I'm in a conference(meetme) with another person, and a third person join our conference when the third person leave the conference I get disconnected from the original conference with a second party. I hope this clear. This does not happen often, is random, anybody experience something similar? or any idea how to fix this problem? Thanks, -- _ -- 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 1.8.10.1 meetme
Hello Jonathan, I thank you for prompt reply to my post. I'm using SIP trunks with Polycom sp450 devices. Also, I was wrong to mention meetme, my conference does not involve using meetme feature on Asterisk. It does not happen often, it happens random. On Thu, Feb 7, 2013 at 3:16 PM, Jonathan Rose wrote: > motty cruz wrote: > > Hello, > > I'm running Asterisk 1.8.10 on Linux box, when I'm in a > > conference(meetme) with another person, and a third person join our > > conference when the third person leave the conference I get > > disconnected from the original conference with a second party. I > > hope this clear. > > This does not happen often, is random, anybody experience something > > similar? or any idea how to fix this problem? > > Let me just start by saying that MeetMe has been touched by a rather > large number of patches in the 11 months and it's quite likely that > your problem will be fixed if you upgrade. r373242 comes to mind in > particular. > > Other than that though, it would be helpful if you added some > additional information, such as what arguments are are running meetme > with and what kinds of devices you are connecting with (SIP phones > presumably?) > > > > -- > Jonathan R. Rose > Digium, Inc. | Software Engineer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > direct +1 256 428 6139 > > Check us out at: http://digium.com & http://asterisk.org > > -- > _ > -- 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 > -- _ -- 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
[asterisk-users] monitoring Asterisk 1.8
Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -- _ -- 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] monitoring Asterisk 1.8
Thanks for the suggestion Carlos, do you have a HowTo? can you point me to one. I unsuccessfully follow one found using google. I'm using CentOs 6.0 Thanks, Motty On Thu, May 9, 2013 at 12:38 PM, Carlos Alvarez wrote: > Monitor what parts exactly? > > Right this moment I'm in the process of installing Munin and the Asterisk > plugin to monitor channel usage, SIP connections, and the like. The Munin > server is running on a separate machine with just the node software on > Asterisk. > > > > On Thu, May 9, 2013 at 12:23 PM, motty cruz wrote: > >> Hello, >> >> i'm looking for suggestions to monitor Asterisk Server? I installed >> Nagios but no success, I do prefer not to install any web server on the >> server running Asterisk. >> >> >> Thanks in advance. >> -Motty >> >> -- >> _ >> -- 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 >> > > > > -- > Carlos Alvarez > TelEvolve > 602-889-3003 > > > -- > _ > -- 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 > -- _ -- 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] monitoring Asterisk 1.8
Thanks for your help; I just want to monitor the queue, calls on hold average time, incoming out going call, I only want to monitor Asterisk, not the server Asterisk in running on. thanks, -Motty On Thu, May 9, 2013 at 1:06 PM, Carlos Rojas wrote: > http://opennms.org/wiki/Installation:Yum > > > On Thu, May 9, 2013 at 4:03 PM, Carlos Rojas wrote: > >> I'm using opennms and It's working fine. >> >> >> >> >> >> On Thu, May 9, 2013 at 3:23 PM, motty cruz wrote: >> >>> Hello, >>> >>> i'm looking for suggestions to monitor Asterisk Server? I installed >>> Nagios but no success, I do prefer not to install any web server on the >>> server running Asterisk. >>> >>> >>> Thanks in advance. >>> -Motty >>> >>> -- >>> _ >>> -- 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 >>> >> >> > > -- > _ > -- 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 > -- _ -- 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
[asterisk-users] asterisk 1.8.22 - : app_meetme.c:1248 build_conf: Unable to open DAHDI pseudo devic
Hello All, I upgraded Asterisk 1.8.10 to Asterisk 1.8.22 since upgrading I can't get meetme feature to work when dial meetme extension, can you please help? It always worked before, also I do not have dahdi installed on this machine, never did. -- Executing [104@sipphones:1] MeetMe("SIP/101-0813", "104") in new stack == Parsing '/etc/asterisk/meetme.conf': == Found [Jun 6 13:17:30] WARNING[11457]: app_meetme.c:1248 build_conf: Unable to open DAHDI pseudo device Thanks, -- _ -- 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 1.8.22 - : app_meetme.c:1248 build_conf: Unable to open DAHDI pseudo devic
Thanks Johan, I did noticed /etc/dahdi so you're right it was installed on one point, I re-install dahdi and problem went away. Thank you very much! On Thu, Jun 6, 2013 at 1:45 PM, Johan Wilfer wrote: > 2013-06-06 22:21, motty cruz skrev: > > Hello All, > > > > I upgraded Asterisk 1.8.10 to Asterisk 1.8.22 since upgrading I can't > > get meetme feature to work when dial meetme extension, can you please > help? > > > > It always worked before, also I do not have dahdi installed on this > > machine, never did. > > > >-- Executing [104@sipphones:1] MeetMe("SIP/101-0813", "104") in > > new stack > > == Parsing '/etc/asterisk/meetme.conf': == Found > > [Jun 6 13:17:30] WARNING[11457]: app_meetme.c:1248 build_conf: Unable > > to open DAHDI pseudo device > > > > Meetme is depending on DAHDI, and have always been. Asterisk would not > even compile Meetme if DAHDI header files isn't present. > > So you must have installed DAHDI at some point? > > Try run dahdi_test (after installing), you should get output like 99.9% > 99.8% ... and so on. Then you know the timing works (dahdi psuedo) > > > -- > Johan Wilfer > > > -- > _ > -- 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 > -- _ -- 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
[asterisk-users] Asterisk 1.8.22
Hello, I have a fully functional Asterisk Server, I want to configure this server to be able to process call from Skype, can someone point me to a howto? or if there are suggestions on best way to approach this problem. Thanks, -- _ -- 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 1.8.22
Hello Mitul, I did purchase a sip channel from skype, and configure it on Asterisk in Sip.conf register => motty2342342:mypass...@sip.skype.com/motty2342342 [skype] username=motty2342342 secret=mypass123 type=friend qualify=yes context=from-skype host=sip.skype.com in a testing skype account client I see is off line, in the manager daskboard in skype sip profiles was register successfuly. I don't know what else to do, any ideas? On Fri, Nov 8, 2013 at 9:30 AM, Mitul Limbani wrote: > Buy SIP Channel from Skype and you can configure it as sip trunk on > asterisk box. > > Mitul > > > On Friday, November 8, 2013, motty cruz wrote: > >> Hello, I have a fully functional Asterisk Server, I want to configure >> this server to be able to process call from Skype, can someone point me to >> a howto? or if there are suggestions on best way to approach this problem. >> >> Thanks, >> > > > -- > Regards, > Mitul Limbani, > Chief Architech & Founder, > Enterux Solutions Pvt. Ltd. > 110 Reena Complex, Opp. Nathani Steel, > Vidyavihar (W), Mumbai - 400 086. India > http://www.enterux.com/ > http://www.entvoice.com/ > email: mi...@enterux.in > DID: +91-22-71967196 > Cell: +91-9820332422 > > > > -- > _ > -- 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 > -- _ -- 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
[asterisk-users] Asterisk 1.8.22.0 Polycom ip soundpoint sp450
Hello, I'm having issues with my phone Polycom sp450 not subscribing to Asterisk server. Asterisk server is fine, firewall is not the issue because a secondary phone is working fine, my connection to the server is fine too, any ideas or suggestions are welcome. -Motty -- _ -- 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 1.8.22.0 Polycom ip soundpoint sp450
Thank you all for your prompt reply, my phone was working up until this morning it just stop subscribing to the Asterisk server. Version: 3.2.4.0244 phone is configure to download configuration via ftp, again it configure right because it was working fine. the phone icon next to the extension number is dark same as the background so that means is not subscribing to the Asterisk server. Thank you very much. On Thu, Jan 2, 2014 at 8:19 AM, Kevin Larsen < kevin.lar...@pioneerballoon.com> wrote: > asterisk-users-boun...@lists.digium.com wrote on 01/02/2014 10:03:19 AM: > > > From: motty cruz > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > , > > Date: 01/02/2014 10:02 AM > > Subject: [asterisk-users] Asterisk 1.8.22.0 Polycom ip soundpoint sp450 > > Sent by: asterisk-users-boun...@lists.digium.com > > > > Hello, I'm having issues with my phone Polycom sp450 not subscribing > > to Asterisk server. Asterisk server is fine, firewall is not the > > issue because a secondary phone is working fine, my connection to > > the server is fine too, any ideas or suggestions are welcome. > > > > -Motty > > We use Polycom 450s as the main desk phone throughout our company and have > no issues with them registering. Without seeing your configs, it is hard to > give you specific advice. The part of your configs you need to be looking > at (assuming you are provisioning from http or ftp), are the following: > > For the Polycom look at the reg section: >reg.1.auth.userId="6534" reg.1.auth.password="myPassword"> > > For Asterisk (in your sip.conf or other appropriate config file): > > [6534](polycom) > callerid="Bob SMith" <6534> > secret=myPassword > mailbox=6534 > > A couple of notes here: 6534 is the extension number for Bob Smith. > "myPassword" in the files should be replaced with whatever password you > have assigned for that phone. The (polycom) template contains all the > options needed for phones to work in my specific install, but doesn't have > anything that would affect registration. > > If you watch the asterisk console when you boot up the phone, do you get > any errors in the console? I know when I am testing/experimenting with new > setups that I often see errors when the phone goes to register. It usually > is because I have either specified a username that doesn't exist in > Asterisk or I have the phone passing an incorrect password with what is > specified in sip.conf. > -- > _ > -- 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 > -- _ -- 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 1.8.22.0 Polycom ip soundpoint sp450
Thank you all, After setting the phone to factory defaults, entered configuration parameters, phone is working again. I really don't know why all sudden stop working. at least know i have a working phone I will go thoroughly through the logs, I hope to find the answer, if I do I will post it here. Thank you again. On Thu, Jan 2, 2014 at 9:12 AM, Ryan Wagoner wrote: > > On Thu, Jan 2, 2014 at 11:13 AM, Eric Wieling wrote: > >> Which firmware version? 4.1.x is only for use with MS Link server. A >> symptom of running 4.1.x firmware with a non-MS server is the phone will >> not show buddies. >> >> > I'm running 4.1.0 on a Polycom IP 335 and IP 550 and version 4.1.5 on a > Polycom VVX 400. Buddies work on all three phones. The firmware is for both > SIP and Lync. You change the base profile option accordingly. Look in the > Polycom UC Software Admin Guide for more information. > > Ryan > > > -- > _ > -- 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 > -- _ -- 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
[asterisk-users] callerid overwrite
Hi all, I'm having issues with overwrite caller id, when I call someone my caller id should be "mycompanyinc" but instead my id shows up as my extension number 101. this is what i have in sip.conf [101] type=friend context=sipphones call-limit=99 callerid="iuser 101" disallow=all allow=ulaw allow=alaw username=101 secret=Passwd dtmfmode=rfc2833 host=dynamic mailbox=101@default nat=yes canreinvite=no this is what i have in extensions.conf [outbound] exten => _91NXXNXX,1,Set(CALLERID(num)=mycompanyinc) exten => _91NXXNXX,2,Dial(SIP/att/${EXTEN:1},80) exten => _9NXX,1,Set(CALLERID(num)=mycompanyinc) exten => _9NXX,2,Dial(SIP/att/${EXTEN:1},80) any ideas? as this happens random, -- _ -- 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] callerid overwrite
Thank you for your reply, I updated extensions.conf file to reflect your suggestion, I will monitor Asterisk for any more issues, Thanks, On Tue, Jan 28, 2014 at 11:23 AM, Andres wrote: > On 1/28/14, 1:55 PM, motty cruz wrote: > > Hi all, > I'm having issues with overwrite caller id, when I call someone my caller > id should be "mycompanyinc" but instead my id shows up as my extension > number 101. > > this is what i have in sip.conf > [101] > type=friend > context=sipphones > call-limit=99 > callerid="iuser 101" > disallow=all > allow=ulaw > allow=alaw > username=101 > secret=Passwd > dtmfmode=rfc2833 > host=dynamic > mailbox=101@default > nat=yes > canreinvite=no > > > this is what i have in extensions.conf > [outbound] > exten => _91NXXNXX,1,Set(CALLERID(num)=mycompanyinc) > > This is how we have it and it works fine on Asterisk 1.8: > Set(CALLERID(number)=insert your number here) > > exten => _91NXXNXX,2,Dial(SIP/att/${EXTEN:1},80) > exten => _9NXX,1,Set(CALLERID(num)=mycompanyinc) > exten => _9NXX,2,Dial(SIP/att/${EXTEN:1},80) > > any ideas? as this happens random, > > > > > > -- > Technical Supporthttp://www.cellroute.net > > > -- > _ > -- 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 > -- _ -- 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] callerid overwrite
look like the issue continues, I am unable to overwrite callerid from sip.conf in extensions.conf, In sip.conf under [general] trustrpid = no < should i change it to yes? Thanks On Tue, Jan 28, 2014 at 1:06 PM, motty cruz wrote: > Thank you for your reply, I updated extensions.conf file to reflect your > suggestion, I will monitor Asterisk for any more issues, > > Thanks, > > > > On Tue, Jan 28, 2014 at 11:23 AM, Andres wrote: > >> On 1/28/14, 1:55 PM, motty cruz wrote: >> >> Hi all, >> I'm having issues with overwrite caller id, when I call someone my caller >> id should be "mycompanyinc" but instead my id shows up as my extension >> number 101. >> >> this is what i have in sip.conf >> [101] >> type=friend >> context=sipphones >> call-limit=99 >> callerid="iuser 101" >> disallow=all >> allow=ulaw >> allow=alaw >> username=101 >> secret=Passwd >> dtmfmode=rfc2833 >> host=dynamic >> mailbox=101@default >> nat=yes >> canreinvite=no >> >> >> this is what i have in extensions.conf >> [outbound] >> exten => _91NXXNXX,1,Set(CALLERID(num)=mycompanyinc) >> >> This is how we have it and it works fine on Asterisk 1.8: >> Set(CALLERID(number)=insert your number here) >> >> exten => _91NXXNXX,2,Dial(SIP/att/${EXTEN:1},80) >> exten => _9NXX,1,Set(CALLERID(num)=mycompanyinc) >> exten => _9NXX,2,Dial(SIP/att/${EXTEN:1},80) >> >> any ideas? as this happens random, >> >> >> >> >> >> -- >> Technical Supporthttp://www.cellroute.net >> >> >> -- >> _ >> -- 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 >> > > -- _ -- 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
[asterisk-users] Asterisk 1.6
Hello All, my asterisk server is constantly under attack [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '"4941" ' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '"4941" ' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '"4941" ' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '"4941" ' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '"4941" ' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password is there a way to reject their registration after a three consecutive tries? Thanks, Call Send SMS Add to Skype You'll need Skype CreditFree via Skype -- _ -- 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 1.6
thank you all for your support. I am using Linux, I only have about 7 users outside our home network. I will learn fail2ban and will use it accordingly. again Thanks for your support. On Fri, Apr 4, 2014 at 7:09 AM, Mauricio Tavares wrote: > > > > On Fri, Apr 4, 2014 at 10:05 AM, Daniel Taylor wrote: > >> I don't know what platform you are on, but if you are on Linux (and >> possibly BSD) you could use "fail2ban" to block them at the network >> interface. >> >> I second fail2ban. If you need some ideas to configure it, you can > steal them from the freepbx setup. > > How many sip phones do you have outside your network? If few and in > well-known IPs, consider limiting access to only those (and the sip > provider you are using). > > >> >> On 04/04/2014 09:00 AM, motty cruz wrote: >> >> Hello All, my asterisk server is constantly under attack >> >> [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 >> handle_request_register: Registration from '"4941" ' >> failed for '194.100.46.132194.100.46.132:56714' - Wrong password >> [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 >> handle_request_register: Registration from '"4941" ' >> failed for '194.100.46.132194.100.46.132:56714' - Wrong password >> [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 >> handle_request_register: Registration from '"4941" ' >> failed for '194.100.46.132194.100.46.132:56714' - Wrong password >> [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 >> handle_request_register: Registration from '"4941" ' >> failed for '194.100.46.132194.100.46.132:56714' - Wrong password >> [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 >> handle_request_register: Registration from '"4941" ' >> failed for '194.100.46.132194.100.46.132:56714' - Wrong password >> >> is there a way to reject their registration after a three consecutive >> tries? >> >> Thanks, >> Call >> Send SMS >> Add to Skype >> You'll need Skype CreditFree via Skype >> >> >> >> >> -- >> Daniel Taylor VP OperationsVocal Laboratories, >> inc.dtay...@vocalabs.com http://www.vocalabs.com/(612)235-5711 >> >> >> -- >> _ >> -- 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 >> > > > -- > _ > -- 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 > -- _ -- 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 1.6
absolutely right A J, thanks for the heads up. I do not intent to implement that solution in production server, I hope to learn it first, build a test server and monitor for a few days or weeks. Thanks again, On Fri, Apr 4, 2014 at 7:38 AM, A J Stiles wrote: > On Friday 04 Apr 2014, Michelle Dupuis wrote: > > Take a look a SecAst from www.generationd.com< > http://www.generationd.com/> > > > > It does everything fail2ban does and more, including blocking users by > > geography (we exclude all of Asia and Africa), detection of break-in > > patterns (even if someone guessed your un/pw), detect changes in dial > > rates, etc. > > > > Grab the free version - its a BIG step up from fail2ban. > > That link points towards a precompiled binary, which could have literally > *anything* lurking in it. I politely advise you to back away slowly, and > break into a run when you think you are out of sight. > > Precompiled binaries without Source Code should be treated like a bottle of > glowing green liquid labelled "drink me", or an offer to come and look at > some > puppies. No reputable software supplier would object to showing you what > is > on the inside. > > -- > AJS > > Note: Originating address only accepts e-mail from list! If replying off- > list, change address to asterisk1list at earthshod dot co dot 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 > -- _ -- 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 1.6
Hello Ishfaq, outside users usually travel around the country and connect from different network, so it won't be possible to lock it down to specific IP. Thanks for your support. On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik wrote: > > > > On 4 April 2014 15:22, motty cruz wrote: > >> thank you all for your support. I am using Linux, I only have about 7 >> users outside our home network. I will learn fail2ban and will use it >> accordingly. >> >> again Thanks for your support. >> >> >> >> Do the 7 users outside of your home network always connect from the same > IP addresses? If so, you can just lock down your SIP port to those 7 IPs > explicitly in your IPTables configuration. > > Another option would be to change which port you're running SIP on. > > > -- > > Ishfaq Malik > Department: VOIP Support > Company: Packnet Limited > t: +44 (0)845 004 4994 > f: +44 (0)161 660 9825 > e: i...@pack-net.co.uk > w: http://www.pack-net.co.uk > > Registered Address: PACKNET LIMITED, Duplex 2, Ducie House > 37 Ducie Street > Manchester, M1 2JW > COMPANY REG NO. 04920552 > > > -- > _ > -- 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 > -- _ -- 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 1.6
that sounds feasible, Thanks Michelle, On Fri, Apr 4, 2014 at 8:25 AM, Michelle Dupuis wrote: > If you know your users are all from with your country, or state, or even > city, you could restrict geographic access in your secast.conf file like > this: > > > ruledefault=deny > ruleexceptions=NA:CA:Ontario:|NA:US:Michigan:Detroit|::Ohio:|NA > > The above would: > - By default deny all source IP's anywhere in the world > - Let in only source IP's from: > 1. North America (continent), Canada (country), Ontario (region) > 2. North America (continent), USA (country), Michigan (region), Detroit > (city) > 3. Any region called 'Ohio' anywhere in the world (not sure why you would > do that but fun example) > 4. Anywhere in North America > > So you can open up your system based solely on where you know your real > users are located. > > -=Michelle=- > > > -- > *From:* asterisk-users-boun...@lists.digium.com < > asterisk-users-boun...@lists.digium.com> on behalf of motty cruz < > motty.c...@gmail.com> > *Sent:* Friday, April 4, 2014 11:15 AM > > *To:* Asterisk Users List > *Subject:* Re: [asterisk-users] Asterisk 1.6 > > Hello Ishfaq, outside users usually travel around the country and > connect from different network, so it won't be possible to lock it down to > specific IP. > > Thanks for your support. > > > On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik wrote: > >> >> >> >> On 4 April 2014 15:22, motty cruz wrote: >> >>> thank you all for your support. I am using Linux, I only have about 7 >>> users outside our home network. I will learn fail2ban and will use it >>> accordingly. >>> >>> again Thanks for your support. >>> >>> >>> >>>Do the 7 users outside of your home network always connect from the >> same IP addresses? If so, you can just lock down your SIP port to those 7 >> IPs explicitly in your IPTables configuration. >> >> Another option would be to change which port you're running SIP on. >> >> >> -- >> >> Ishfaq Malik >> Department: VOIP Support >> Company: Packnet Limited >> t: +44 (0)845 004 4994 >> f: +44 (0)161 660 9825 >> e: i...@pack-net.co.uk >> w: http://www.pack-net.co.uk >> >> Registered Address: PACKNET LIMITED, Duplex 2, Ducie House >> 37 Ducie Street >> Manchester, M1 2JW >> COMPANY REG NO. 04920552 >> >> >> -- >> _ >> -- 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 >> > > > -- > _ > -- 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 > -- _ -- 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 1.6
that is definitely another options, thanks for the range of options provided, Thanks On Sat, Apr 5, 2014 at 4:51 PM, Duncan Turnbull wrote: > Another option we like, but i depends on your preferences is to run them > over openvpn. Works for Mac, Linux and Windows clients. > > Since all out clients are under our control we use openvpn a lot and > yealink and other phones have it built in so they can connect directly once > initially setup > > Cheers Duncan > > On 5/04/2014, at 4:36 am, motty cruz wrote: > > that sounds feasible, Thanks Michelle, > > > > > On Fri, Apr 4, 2014 at 8:25 AM, Michelle Dupuis wrote: > >> If you know your users are all from with your country, or state, or >> even city, you could restrict geographic access in your secast.conf file >> like this: >> >> >> ruledefault=deny >> ruleexceptions=NA:CA:Ontario:|NA:US:Michigan:Detroit|::Ohio:|NA >> >> The above would: >> - By default deny all source IP's anywhere in the world >> - Let in only source IP's from: >> 1. North America (continent), Canada (country), Ontario (region) >> 2. North America (continent), USA (country), Michigan (region), Detroit >> (city) >> 3. Any region called 'Ohio' anywhere in the world (not sure why you would >> do that but fun example) >> 4. Anywhere in North America >> >> So you can open up your system based solely on where you know your real >> users are located. >> >> -=Michelle=- >> >> >> -- >> *From:* asterisk-users-boun...@lists.digium.com < >> asterisk-users-boun...@lists.digium.com> on behalf of motty cruz < >> motty.c...@gmail.com> >> *Sent:* Friday, April 4, 2014 11:15 AM >> >> *To:* Asterisk Users List >> *Subject:* Re: [asterisk-users] Asterisk 1.6 >> >> Hello Ishfaq, outside users usually travel around the country and >> connect from different network, so it won't be possible to lock it down to >> specific IP. >> >> Thanks for your support. >> >> >> On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik wrote: >> >>> >>> >>> >>> On 4 April 2014 15:22, motty cruz wrote: >>> >>>> thank you all for your support. I am using Linux, I only have about 7 >>>> users outside our home network. I will learn fail2ban and will use it >>>> accordingly. >>>> >>>> again Thanks for your support. >>>> >>>> >>>> >>>>Do the 7 users outside of your home network always connect from the >>> same IP addresses? If so, you can just lock down your SIP port to those 7 >>> IPs explicitly in your IPTables configuration. >>> >>> Another option would be to change which port you're running SIP on. >>> >>> >>> -- >>> >>> Ishfaq Malik >>> Department: VOIP Support >>> Company: Packnet Limited >>> t: +44 (0)845 004 4994 >>> f: +44 (0)161 660 9825 >>> e: i...@pack-net.co.uk >>> w: http://www.pack-net.co.uk >>> >>> Registered Address: PACKNET LIMITED, Duplex 2, Ducie House >>> 37 Ducie Street >>> Manchester, M1 2JW >>> COMPANY REG NO. 04920552 >>> >>> >>> -- >>> _ >>> -- 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 >>> >> >> >> -- >> _ >> -- 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 >> > > -- > _ > -- 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 > > > > -- > _ > -- 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 > -- _ -- 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
[asterisk-users] how to hangup Local/100 channel
Hello All, one of the extensions fall into a loop, I don't know how to hangup that channel -- Executing [i@autoatten:2] Goto("Local/100@sipphones-01b2;2", "s,2") in new stack -- Goto (autoatten,s,2) -- Sent into invalid extension 's' in context 'autoatten' on Local/200@sipphones-01b2;2 -- Executing [i@autoatten:1] Playback("Local/2000@sipphones-01b2;2", "pbx-invalid") in new stack -- Playing 'pbx-invalid.gsm' (language 'en') -- Executing [i@autoatten:2] Goto("Local/200@sipphones-01b2;2", "s,2") in new stack -- Goto (autoatten,s,2) -- Sent into invalid extension 's' in context 'autoatten' on Local/200@sipphones-01b2;2 -- Executing [i@autoatten:1] Playback("Local/200@sipphones-01b2;2", "pbx-invalid") in new stack -- Playing 'pbx-invalid.gsm' (language 'en') -- Remote UNIX connection -- Executing [i@autoatten:2] Goto("Local/200@sipphones-01b2;2", "s,2") in new stack -- Goto (autoatten,s,2) -- Sent into invalid extension 's' in context 'autoatten' on Local/200@sipphones-01b2;2 -- Executing [i@autoatten:1] Playback("Local/200@sipphones-01b2;2", "pbx-invalid") in new stack -- Playing 'pbx-invalid.gsm' (language 'en') -- Executing [i@autoatten:2] Goto("Local/200@sipphones-01b2;2", "s,2") in new stack -- Goto (autoatten,s,2) -- Sent into invalid extension 's' in context 'autoatten' on Local/200@sipphones-01b2;2 -- Executing [i@autoatten:1] Playback("Local/200@sipphones-01b2;2", "pbx-invalid") in new stack -- Playing 'pbx-invalid.gsm' (language 'en') any ideas? Thanks, -- _ -- 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] how to hangup Local/100 channel
Thanks for your support, I was able to soft hangup using "hangup request Local/200@users-0001b" first, I did core show channels, after stopping this loops I was able to fixed that problem from happening again, Thanks, On Mon, May 5, 2014 at 8:56 AM, Steve Edwards wrote: > On Mon, 5 May 2014, motty cruz wrote: > > one of the extensions fall into a loop, I don't know how to hangup that >> channel >> >> -- Goto (autoatten,s,2) >> -- Sent into invalid extension 's' in context 'autoatten' on >> Local/200@sipphones-01b2;2 >> >> any ideas? >> > > If you're asking how to prevent it from happening, how about 'exten = > s,2,hangup()?' > > If you're asking how to hang up the channel while it is in a loop, what > have you tried? Does 'channel request hangup' help? > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > -- > _ > -- 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 > -- _ -- 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
[asterisk-users] Asterisk 1.8.22
Hello, recently I have seen spike in attacks on my asterisk server, this is what I get on the LCD of my phone: 201@76.220.5.205 or calls from 1000 sip1000@76.2230.5.205, have any idea on how to stop this calls? Thanks, -- _ -- 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 1.8.22
Thanks for your support, I will try your suggestions, will let you know how it goes, Thanks, On Tue, May 13, 2014 at 7:28 AM, Michelle Dupuis wrote: > Another alternative is SecAst (Asterisk intrusion detection system). > Grab the free version from www.generationd.com > > > It does everything fail2ban does, plus you have the option of blocking > IP's based on geograhic origin, detecting suspicious call patterns, etc. > > > -=M=- > > All opinions posted are my own. But as an employee of GenerationD System > my views are undoubtedly biased :) > > > -- > *From:* asterisk-users-boun...@lists.digium.com < > asterisk-users-boun...@lists.digium.com> on behalf of motty cruz < > motty.c...@gmail.com> > *Sent:* Monday, May 12, 2014 5:43 PM > *To:* Asterisk Users List > *Subject:* [asterisk-users] Asterisk 1.8.22 > > Hello, > recently I have seen spike in attacks on my asterisk server, this is what > I get on the LCD of my phone: 201@76.220.5.205 > > or calls from 1000 sip1000@76.2230.5.205, > > have any idea on how to stop this calls? > > Thanks, > > -- > _ > -- 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 > -- _ -- 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
[asterisk-users] share mailbox Asterisk 1.8.22
Hello, I want to share mailbox between two extensions Ext. 101 Ext. 102 I want the messages to go to mailbox 101, when when checked mailbox from extension 102 to be able to clear the bliking red light. here is extensions.conf exten => 102,hint,SIP/${EXTEN} exten => 102,1,Dial(SIP/101&SIP/102,20,t) exten => 102,2,Voicemail(101,u) exten => 102,102,Voicemail(101,b) exten => 102,103,Hangup sip.conf [102] type=friend context=sipphones call-limit=99 callerid="Jo" <102> disallow=all allow=ulaw allow=alaw username=102 secret=xexpasswd dtmfmode=rfc2833 host=dynamic mailbox=101 nat=yes canreinvite=no this configuration work ok but, light msg keeps bilking after checking for mesages. Any suggestions? Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] share mailbox Asterisk 1.8.22
yes they're able to hear the same msg, in /var/spool/asterisk/default/ rm -rf 102 ln -s 101 102 but it does not clear out, Thanks, On Tue, Jun 24, 2014 at 3:31 PM, Kevin Larsen < kevin.lar...@pioneerballoon.com> wrote: > I have done this for one of my users in a very similar fashion. When 102 > checks the voicemail, do they hear the correct voicemails? Ours clears just > fine in this situation. > > Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 > > asterisk-users-boun...@lists.digium.com wrote on 06/24/2014 04:37:26 PM: > > > From: motty cruz > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > , > > Date: 06/24/2014 04:37 PM > > Subject: [asterisk-users] share mailbox Asterisk 1.8.22 > > Sent by: asterisk-users-boun...@lists.digium.com > > > > Hello, I want to share mailbox between two extensions > > Ext. 101 > > Ext. 102 > > > > I want the messages to go to mailbox 101, when when checked mailbox > > from extension 102 to be able to clear the bliking red light. > > > > here is extensions.conf > > exten => 102,hint,SIP/${EXTEN} > > exten => 102,1,Dial(SIP/101&SIP/102,20,t) > > exten => 102,2,Voicemail(101,u) > > exten => 102,102,Voicemail(101,b) > > exten => 102,103,Hangup > > > > sip.conf > > [102] > > type=friend > > context=sipphones > > call-limit=99 > > callerid="Jo" <102> > > disallow=all > > allow=ulaw > > allow=alaw > > username=102 > > secret=xexpasswd > > dtmfmode=rfc2833 > > host=dynamic > > mailbox=101 > > nat=yes > > canreinvite=no > > > > this configuration work ok but, light msg keeps bilking after > > checking for mesages. > > > > Any suggestions? > > Thanks, > > > > > > __ > > 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 -- > > 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 > -- > _ > -- 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 > -- _ -- 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] share mailbox Asterisk 1.8.22
Thanks Kevin, can you provide me with example of your code? if you don't mind? Thanks, On Tue, Jun 24, 2014 at 3:46 PM, Kevin Larsen < kevin.lar...@pioneerballoon.com> wrote: > asterisk-users-boun...@lists.digium.com wrote on 06/24/2014 05:36:16 PM: > > > From: motty cruz > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > , > > Date: 06/24/2014 05:36 PM > > Subject: Re: [asterisk-users] share mailbox Asterisk 1.8.22 > > Sent by: asterisk-users-boun...@lists.digium.com > > > > yes they're able to hear the same msg, in /var/spool/asterisk/default/ > > > rm -rf 102 > > ln -s 101 102 > > > > but it does not clear out, > > > > Thanks, > > > > That is where your setup and mine are different. I have my second > extension directly check the first extensions mailbox as opposed to using a > symlink. That way, when the box is cleared, it is actually happening in the > original mailbox. > > Basically, my code to check voicemail uses the CALLERID(num) to determine > the mailbox and I have both extensions set to use the same caller ID. This > works for me as both extensions belong to the same person (one phone at the > office, one at home) and they want to always have their outbound calls show > up as if they were from their published number regardless of which phone > they use. > -- > _ > -- 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 > -- _ -- 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
[asterisk-users] Asterisk secure fine tune - stop attack
Hi All, I see this kind of attack on our Asterisk Server, do you know how to block that IP? [Sep 4 07:41:06] NOTICE[7375]: chan_sip.c:23375 handle_request_invite: Call from '' (213.136.81.166:9306) to extension '34422' rejected because extension not found in context 'default'. Thanks in advance, -Motty -- _ -- 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 secure fine tune - stop attack
Thanks, looks like fail2ban is the way to go, I would prefer a different alternatives if there is one. I tried deny=IP/netmask but did not work for me, in sip.conf. seems like fail2ban is what you all are using, so I will give it a try. Thanks, On Thu, Sep 4, 2014 at 7:58 AM, Thorsten Göllner wrote: > > Am 04.09.2014 16:44, schrieb motty cruz: > > Hi All, > I see this kind of attack on our Asterisk Server, do you know how to block > that IP? > > [Sep 4 07:41:06] NOTICE[7375]: chan_sip.c:23375 handle_request_invite: > Call from '' (213.136.81.166:9306) to extension '34422' rejected because > extension not found in context 'default'. > > > You should not invest time in blocking single IPs. Take a look at > "fail2ban". > http://www.fail2ban.org/wiki/index.php/Asterisk > > -Thorsten- > > -- > _ > -- 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 > -- _ -- 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 secure fine tune - stop attack
Hi A J, believe me, I wish i do as you suggested, however I have a few extensions outside the office with dynamic IPs, so that is not a possibility. Thanks for your suggestions, I will try fail2ban. I don't know how complicated is to implement that on production server. Thanks, -Motty On Thu, Sep 4, 2014 at 8:19 AM, A J Stiles wrote: > On Thursday 04 Sep 2014, motty cruz wrote: > > Hi All, > > I see this kind of attack on our Asterisk Server, do you know how to > block > > that IP? > > Instead of blocking unwanted IPs, you should be permitting only wanted IPs. > > -- > AJS > > Note: Originating address only accepts e-mail from list! If replying off- > list, change address to asterisk1list at earthshod dot co dot 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 > -- _ -- 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 secure fine tune - stop attack
Thank you all for your support, your suggestions are welcome. Thanks, On Thu, Sep 4, 2014 at 9:26 AM, Chris Bagnall wrote: > On 4/9/14 4:58 pm, Eric Wieling wrote: > >> If we don't need to allow access from outside the USA we block access >> from all non-ARIN IP addresses by using iptables. This takes care of at >> least 80% of attacks. >> > > Likewise here (though RIPE rather than ARIN, since we're the other side of > the pond). > > You can also take it a bit further: if, for example, you know what ISP(s) > your dynamic clients are using, you can limit connections to the IP ranges > those ISP(s) use - look up their ranges on he.net's BGP looking glass if > you need to find out what ranges they're using. > > Another thing I've been playing with of late is using iptables' string > matching functionality to block user agents of known attack vectors: > 'sipcli', 'sipvicious', 'friendly-scanner', etc. > > This seems to work remarkably well, though what impact it has on net > performance under load remains to be seen. > > Kind regards, > > Chris > -- > This email is made from 100% recycled electrons > > > -- > _ > -- 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 > -- _ -- 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
[asterisk-users] Asterisk failed to authenticate device - attack attempt.
Hi all, I continue to see the following msg on my Asterisk log: [Sep 8 15:34:37] NOTICE[7375]: chan_sip.c:23277 handle_request_invite: Failed to authenticate device 9009;tag=8dd48dd2 IP: 196.107.xx.xx is my asterisk server IP address. I don't know what it means and how to cover any holes that attacker is trying to exploit. Thanks, Motty -- _ -- 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] chan_sip.c:23647 handle_request_invite: Failed to authenticate device
Hello Deepak, 601;tag=2f498fbd is the 111.118.185.107 your server IP? it could be a client trying to authenticate as Rusty suggest or an attacker attempting to gain access, if you want to find out what IP address that request is coming from do the following command. make sure you have tcpdump installed. tcpdump -lni eth0 -f "udp port 5060" monitor your server to make sure you catch the attacker, also you could do tcpdump -w capture.cap and analyze it with wireshark. Thanks, Motty On Thu, Sep 11, 2014 at 2:28 PM, Rusty Newton wrote: > On Thu, Sep 11, 2014 at 10:14 AM, Deepak Bhatia > wrote: > > Hi, > > > > Why are we getting message in the asterisk > > > > [Sep 10 12:55:23] NOTICE[15043]: chan_sip.c:23647 handle_request_invite: > > Failed to authenticate device 601; > > tag=2f498fbd > > [Sep 10 12:55:24] NOTICE[15043]: chan_sip.c:23647 handle_request_invite: > > Failed to authenticate device 601;tag=209a8aa9 > > > > Regards > > > > Deepak Bhatia > > Deepak, > > This is commonly a wrong password on the client-side. That is, the > device attempting to call Asterisk is failing an authentication > challenge for one reason or another. > > For anyone to help you, you would probably need to post your sip.conf > configuration (full) and screenshots of your soft/hard client > configuration. > > Obviously you would want to use a fake password and sanitize IP > addresses if necessary. > > Thanks, > > -- > Rusty Newton > Digium, Inc. | Community Support Manager > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > direct: +1 256 428 6200 > > Check us out at: http://digium.com & http://asterisk.org > > -- > _ > -- 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 > -- _ -- 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
[asterisk-users] Asterisk- VoIP Phones ring from Ext. 101 but that Ext. does not exist in extensions.conf
Hello, a user outside the office regularly gets a call from ext. 101 but that extension does not exist in my extensions.conf. when the user pickup the phone no one answers. Any Idea how to fix this issue? that user uses Polycom SP 450, Thanks in advance, Motty -- _ -- 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- VoIP Phones ring from Ext. 101 but that Ext. does not exist in extensions.conf
Thanks Eric, for point to polycom instructions I will give it a try. Kevin, I am sure called is not originating from our system, Thanks for your support. On Tue, Sep 16, 2014 at 9:08 AM, Kevin Larsen < kevin.lar...@pioneerballoon.com> wrote: > > Hello, > > a user outside the office regularly gets a call from ext. 101 but > > that extension does not exist in my extensions.conf. when the user > > pickup the phone no one answers. Any Idea how to fix this issue? > > that user uses Polycom SP 450, > > First thing to look at is at the time the user receives the call, do you > show anything in your Asterisk CLI? I would make sure that the call is > actually originating from your system and track back from there. > -- > _ > -- 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 > -- _ -- 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
[asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism
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. can someone point me to a right direction to achieve this goal? Thanks, Motty -- _ -- 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
Thank you Julian, would it be possible to block calls to international calls except certain countries? I just want to make sure that if attackers try to place calls outside the states they not succeed. Thanks, Motty On Thu, Sep 18, 2014 at 12:55 PM, Julian Beach wrote: > 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 > > > -- > _ > -- 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 -- _ -- 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
Thanks Eric, for respectfully pointing that link, it is the reason why I am posting my question for lack of knowledge. I had been working on Asterisk for the last 4 years, I am always learning something knew. - Motty On Thu, Sep 18, 2014 at 2:15 PM, Eric Wieling wrote: > Your question demonstrates a fundamental lack of Asterisk concepts and > knowledge. You should start by reading http://www.asteriskdocs.org/ and > go from there.Asterisk is not something you can learn in a few days. > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *motty cruz > *Sent:* Thursday, September 18, 2014 4:52 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Asterisk prefix code to dial a high fraud > country - security mechanism > > > > Thank you Julian, > > > > would it be possible to block calls to international calls except certain > countries? I just want to make sure that if attackers try to place calls > outside the states they not succeed. > > > > Thanks, > Motty > > > > On Thu, Sep 18, 2014 at 12:55 PM, Julian Beach > wrote: > > 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 > > > -- > _ > -- 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 > > > > -- > _ > -- 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 > -- _ -- 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
absolutely not what I meant, I really meant to say thank you for respectfully pointing that out. -Motty On Thu, Sep 18, 2014 at 2:32 PM, Eric Wieling wrote: > It is unfortunate > http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#asterisk-DP-Basics-SECT-3.6 > is not helpful to you. > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *motty cruz > *Sent:* Thursday, September 18, 2014 5:27 PM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Asterisk prefix code to dial a high fraud > country - security mechanism > > > > Thanks Eric, for respectfully pointing that link, it is the reason why I > am posting my question for lack of knowledge. I had been working on > Asterisk for the last 4 years, I am always learning something knew. > > > > - Motty > > > > On Thu, Sep 18, 2014 at 2:15 PM, Eric Wieling wrote: > > Your question demonstrates a fundamental lack of Asterisk concepts and > knowledge. You should start by reading http://www.asteriskdocs.org/ and > go from there.Asterisk is not something you can learn in a few days. > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *motty cruz > *Sent:* Thursday, September 18, 2014 4:52 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Asterisk prefix code to dial a high fraud > country - security mechanism > > > > Thank you Julian, > > > > would it be possible to block calls to international calls except certain > countries? I just want to make sure that if attackers try to place calls > outside the states they not succeed. > > > > Thanks, > Motty > > > > On Thu, Sep 18, 2014 at 12:55 PM, Julian Beach > wrote: > > 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 > > > -- > _ > -- 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 > > > > > -- > _ > -- 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 > > > > -- > _ > -- 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 > -- _ -- 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
Thank you AJ, I will certainly not copy and past; I want to believe I understand the risk. I needed some kind of direction, thank you for your support. -Motty On Fri, Sep 19, 2014 at 2:51 AM, A J Stiles wrote: > On Thursday 18 Sep 2014, motty cruz 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. > > > > can someone point me to a right direction to achieve this goal? > > > > Thanks, > > Motty > > Not many people are going to want to answer this definitively, I suspect, > for > fear of being blamed if you copy what they did, it doesn't work for you and > you get landed with huge bills for calls you didn't make. Securing > Asterisk > is never as easy as you think. > > > However, if you look back through my own posts, I did post some dialplan > code > a short while ago, relating to a PIN entry. Feel free to borrow that and > play > around with it; but note, I will not accept any responsibility for it not > being as secure as you thought! > > > Another thing to consider would be only allowing overseas calls from a > particulat context; any extension that does not require the ability to call > abroad should be placed in a different default context. If you know you > will > only ever need to call a restricted range of foreign numbers, consider > giving > them "short codes" -- endpoints effectively within your own internal > numbering > scheme -- and sending calls to _00X. to a recorded message. > > [overseas-offices] > ; this context is only for phones which need the ability to call overseas > > ; 8000 is office in France > exten => 8000,1,Set(CALLERID(num)=${OUTGOING_IDENT}) > exten => 8000,n,Dial(${OUT_TRUNK}/0033251478820,180) > exten => 8000,n,Hangup() > > ; 8010 is office in India > exten => 8010,1,Set(CALLERID(num)=${OUTGOING_IDENT}) > exten => 8010,n,Dial(${OUT_TRUNK}/00918322494200,180) > exten => 8010,n,Hangup() > > ; . > > [default] > ; play suitably sarchastic announcement to chancers > _00X.,1,Play(ajs-not_allowed) > _00X.,n,Hangup() > > > Basically, be paranoid; and even then, don't forget, you probably aren't > being > paranoid enough. > > -- > AJS > > Note: Originating address only accepts e-mail from list! If replying off- > list, change address to asterisk1list at earthshod dot co dot 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 > -- _ -- 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
[asterisk-users] how to strip +1 out of incoming number
Hello, our VoIP send us caller ID +1(area)(number) for instance +16024224334 is there a way to strip +1 out of caller ID? -- Thanks for your support, Motty -- _ -- 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
[asterisk-users] adding area code
Hello, I would like to add area code if clients dial 7 digits, it that possible? currently clients dial prefix 9 plus local number, however my SIP provider is requiring to dial 10 digits. is it possible to add area code? Thanks, Motty -- _ -- 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] adding area code
Thanks for your reply, [globals] AREACODE=381 [outbound] exten => _NXX,1,Dial(SIP/SIP-Provider/1${AREACODE}${EXTEN},80) did not work for me, any ideas? Thanks, On 04/27/2015 01:59 PM, Phil Reynolds wrote: On 27 April 2015 21:32:42 BST, Motty Cruz wrote: >Hello, > >I would like to add area code if clients dial 7 digits, it that >possible? currently clients dial prefix 9 plus local number, however my > >SIP provider is requiring to dial 10 digits. is it possible to add area > >code?r Quite simple - you need to match on NXXX and when passing it to the SIP provider, present ${AREACODE}${EXTEN}, having first defined AREACODE in [globals]. -- Sent from my Android device with K-9 Mail. Please excuse my brevity. -- _ -- 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] adding area code
here is what I have: exten => _9XXX,1,Set(l_HomeAreaCode=381) exten => _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1}) exten => _9XXX,n,Dial(SIP/SIP-Provider/${dialnumber},80) not having success; "Got SIP reponse 503" Service Unavailable" On 04/27/2015 02:19 PM, Bryant Zimmerman wrote: Motty Yes From your dial plan accept 9 + 7 digits then concat your dialed number together with your areacode. This s a brief example. exten => _9XXX,1,Set(l_HomeAreaCode=555) exten => _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) ;; This line should combine your area code and the last 7 digits of your dialed phone number exten => _9XXX,n,Dial(SIP/${dialnumber},35) Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 ---- *From*: "Motty Cruz" *Sent*: Monday, April 27, 2015 4:33 PM *To*: asterisk-users@lists.digium.com *Subject*: [asterisk-users] adding area code Hello, I would like to add area code if clients dial 7 digits, it that possible? currently clients dial prefix 9 plus local number, however my SIP provider is requiring to dial 10 digits. is it possible to add area code? Thanks, Motty -- _ -- 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 -- _ -- 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] adding area code
forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS. Thanks, On 04/27/2015 02:38 PM, Motty Cruz wrote: here is what I have: exten => _9XXX,1,Set(l_HomeAreaCode=381) exten => _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1}) exten => _9XXX,n,Dial(SIP/SIP-Provider/${dialnumber},80) not having success; "Got SIP reponse 503" Service Unavailable" On 04/27/2015 02:19 PM, Bryant Zimmerman wrote: Motty Yes From your dial plan accept 9 + 7 digits then concat your dialed number together with your areacode. This s a brief example. exten => _9XXX,1,Set(l_HomeAreaCode=555) exten => _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) ;; This line should combine your area code and the last 7 digits of your dialed phone number exten => _9XXX,n,Dial(SIP/${dialnumber},35) Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 ---- *From*: "Motty Cruz" *Sent*: Monday, April 27, 2015 4:33 PM *To*: asterisk-users@lists.digium.com *Subject*: [asterisk-users] adding area code Hello, I would like to add area code if clients dial 7 digits, it that possible? currently clients dial prefix 9 plus local number, however my SIP provider is requiring to dial 10 digits. is it possible to add area code? Thanks, Motty -- _ -- 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 -- _ -- 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] adding area code
this code worked for me, here is what I did and worked for me: exten => 1381+NXX,1,Set(CALLERID(number)=3817383444) exten => 1+NXXNXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80) Thanks for you help! On 04/27/2015 02:56 PM, Matt Riddell wrote: On 27Apr, 2015, at 16:39, Motty Cruz <mailto:motty.c...@gmail.com>> wrote: forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS. Thanks, On 04/27/2015 02:38 PM, Motty Cruz wrote: here is what I have: exten => _9XXX,1,Set(l_HomeAreaCode=381) exten => _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1}) exten => _9XXX,n,Dial(SIP/SIP-Provider/${dialnumber},80) not having success; "Got SIP reponse 503" Service Unavailable” Can you send us the console output when you make the call? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] adding area code
I apologize, I coppied the wrong code, here is the code I am using: ; Adding Area code and striping 9 for local numbers exten => _9XXX,n,Set(CALLERID(all)= <3817383444>) exten => _9XXX,n,Dial(SIP/intelepeer/1381${EXTEN:1},80) Thanks, motty On 04/28/2015 11:54 AM, Chad Wallace wrote: On Tue, 28 Apr 2015 07:21:12 -0700 Motty Cruz wrote: here is what I did and worked for me: exten => 1381+NXX,1,Set(CALLERID(number)=3817383444) exten => 1+NXXNXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80) I find it hard to believe this is working. First, you don't have a leading underscore on your patterns. Your users aren't literally dialing the N's and X's are they? Second, what's with the plus in the extension? You want your users to dial that? Third, that's two different extensions, one with priority 1 and one with priority 2. The first one will set a variable and hangup, and the second there's no priority 1 for that extension... I've never tried that... I'm assuming it just won't work. -- _ -- 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
[asterisk-users] Asterisk how to setup alarm too many outgoing calls from same user
Hello, I would like to setup a mechanism to trigger an alarm if user is deal too many numbers within a very short period of time. Safeguard against users hacked accounts. can someone help? Thanks, -- _ -- 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
[asterisk-users] Asterisk pin code for out-going international calls (safeguard against fraud)
Hello All, I will like to configure Asterisk to use PIN Code for all outgoing international calls. Also, any suggestions as to when should I prompt users for code prior to dialing the number or after dialing the number? can someone provide with a example on how to accomplish this goal? I am a bit confuse by this : http://forums.digium.com/viewtopic.php?p=130936&sid=707f657f7a61dfed55e4922304925091 Thanks for your help. -- _ -- 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 pin code for out-going international calls (safeguard against fraud)
Hello, I used this guide, it worked for me: http://www.binaryheartbeat.net/2014/03/asterisk-pin-based-dialing.html Thanks, On 07/06/2015 04:54 PM, John Kiniston wrote: The Authenticate application will do this for you. https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Authenticate You can either give it a single PIN to use for all calls, Authenticate using a value in the Asterisk Database, Or use a plain text file for the PIN's On Mon, Jul 6, 2015 at 2:43 PM, Motty Cruz <mailto:motty.c...@gmail.com>> wrote: Hello All, I will like to configure Asterisk to use PIN Code for all outgoing international calls. Also, any suggestions as to when should I prompt users for code prior to dialing the number or after dialing the number? can someone provide with a example on how to accomplish this goal? I am a bit confuse by this : http://forums.digium.com/viewtopic.php?p=130936&sid=707f657f7a61dfed55e4922304925091 Thanks for your help. -- _ -- 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 -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- 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 pin code for out-going international calls (safeguard against fraud)
Here is what i have, exten => _011X,1,Set(CALLERID(number)=factory2) exten => _011X,2,Authenticate(/home/asterisk/passwds.conf,m,3) exten => _011X,3,NoOp(user has been authenticated) exten => _011X,n,Dial(SIP/VoIPSP1/${EXTEN:1},80) exten => _011X,n,HangUp() I would like to add background music if authentication failed, then after 6 minutes hangup any ideas, suggestions? On 07/07/2015 09:09 AM, Motty Cruz wrote: Hello, I used this guide, it worked for me: http://www.binaryheartbeat.net/2014/03/asterisk-pin-based-dialing.html Thanks, On 07/06/2015 04:54 PM, John Kiniston wrote: The Authenticate application will do this for you. https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Authenticate You can either give it a single PIN to use for all calls, Authenticate using a value in the Asterisk Database, Or use a plain text file for the PIN's On Mon, Jul 6, 2015 at 2:43 PM, Motty Cruz <mailto:motty.c...@gmail.com>> wrote: Hello All, I will like to configure Asterisk to use PIN Code for all outgoing international calls. Also, any suggestions as to when should I prompt users for code prior to dialing the number or after dialing the number? can someone provide with a example on how to accomplish this goal? I am a bit confuse by this : http://forums.digium.com/viewtopic.php?p=130936&sid=707f657f7a61dfed55e4922304925091 Thanks for your help. -- _ -- 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 -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- 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] Sending E-Mail from voicemail with AND without attachment
Hello, this worked for me: 500 => mysecret,Motty mailbox,mo...@domain.com,,tz=pacific,attach=yes|delete=1 Thanks, Motty On 07/10/2015 07:45 AM, Luca Bertoncello wrote: Hi again, I'm trying to send two E-Mails when a message comes in the voicemail, the first WITH the attachment, and the second WITHOUT. But I don't get it working... I wrote 004935 => SECRET,John Doe,first@emailde,sec...@email.de,attach=no but both E-Mail don't have the attachment. Writing 004935 => SECRET,John Doe,first@emailde,attach=no|sec...@email.de results in invalid E-Mail-Address for the second E-Mail. I really don't know how to solve my problem, other the write a little program on the mail server that remove the attachment from the second E-Mail... Any help is appreciated Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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
[asterisk-users] Asterisk 1.8.22.0 built - encrypt authentication
Hello, I would like to encrypt password between Asterisk servers and clients. is there an easy way to do so? I am running Asterisk 1.8.22.0 built on CentOS 6.3 Thanks, .Motty -- _ -- 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 1.8.22.0 built - encrypt authentication
Thanks Oliver, I am using this: http://www.voip-info.org/wiki/view/Asterisk+sip+md5secret however, I wanted to know if there is a way to use some sort of Certificates. http://www.voip-info.org/wiki/view/Asterisk+encryption has anybody used this feature? Thanks, .Motty On 07/29/2015 03:40 PM, Oli-net wrote: Hello, Try MD5SUM like that in a terminal echo -n "myverysecretpassword" | md5sum It will return you something like that 22ea6cf875d66b15d275684427275dfdf witch is your password in an MD5 format. Hope this help Oliver Le 29/07/2015 16:30, Motty Cruz a écrit : Hello, I would like to encrypt password between Asterisk servers and clients. is there an easy way to do so? I am running Asterisk 1.8.22.0 built on CentOS 6.3 Thanks, .Motty -- _ -- 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
[asterisk-users] dahdi auto-call multiple destinations
Am trying to get a script to call multiple destinations, for instance I have the following file: Channel: dahdi/g1/6078880 CallerID: "Room Tempeture" <800579> MaxRetries: 2 RetryTime: 60 WaitTime: 20 This works great for me, however I am trying to add a secondary number, is that possible to add dahdi/g1/6078880&dahdi/g1/7068880 ? -- Thanks for your support, Motty -- _ -- 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
[asterisk-users] Asterisk 1.8 secure SIP session only
Hello, I am trying to secure SIP session with TLS on Asterisk Server 1.8. I keep getter an error, == Problem setting up ssl connection: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca [2016-05-04 09:31:17] WARNING[30032]: tcptls.c:254 handle_tcptls_connection: FILE * open failed! I tried both signed and self-signed cert to no avail. Here is my Configuration: Sip.conf tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/keys/box1.pem tlscapath=/etc/asterisk/keys tlscipher=ALL tlsclientmethod=tlsv1 sip.conf ext. [5006] type=peer context=sipext call-limit=3 trustrpid=no callerid="Rec" <5006> disallow=all allow=ulaw allow=alaw username=5006 secret=9fcbb025200881850526bc57d59885c3 dtmfmode=rfc2833 host=dynamic mailbox=5006 nat=yes canreinvite=no transport=tls == Problem setting up ssl connection: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca [2016-05-04 09:31:17] WARNING[30032]: tcptls.c:254 handle_tcptls_connection: FILE * open failed! Any ideas? -- _ -- 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 1.8 secure SIP session only
Thank you Markos, finally was able to secure SIP session with TLS between server & client. Thanks for you support! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markos Vakondios Sent: Wednesday, May 04, 2016 10:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 secure SIP session only Your CA cert is missing. Add in sip.conf: tlscafile=/etc/asterisk/keys/ca.crt You don't need: tlscapath=/etc/asterisk/keys On 4 May 2016 at 19:43, Motty Cruz wrote: Hello, I am trying to secure SIP session with TLS on Asterisk Server 1.8. I keep getter an error, == Problem setting up ssl connection: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca [2016-05-04 09:31:17] WARNING[30032]: tcptls.c:254 handle_tcptls_connection: FILE * open failed! I tried both signed and self-signed cert to no avail. Here is my Configuration: Sip.conf tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/keys/box1.pem tlscapath=/etc/asterisk/keys tlscipher=ALL tlsclientmethod=tlsv1 sip.conf ext. [5006] type=peer context=sipext call-limit=3 trustrpid=no callerid="Rec" <5006> disallow=all allow=ulaw allow=alaw username=5006 secret=9fcbb025200881850526bc57d59885c3 dtmfmode=rfc2833 host=dynamic mailbox=5006 nat=yes canreinvite=no transport=tls == Problem setting up ssl connection: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca [2016-05-04 09:31:17] WARNING[30032]: tcptls.c:254 handle_tcptls_connection: FILE * open failed! Any ideas? -- _ -- 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 -- _ -- 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
[asterisk-users] Asterisk Secure SIP session TLS port 5061
I finally secure SIP session between Asterisk server and a remote client. My questions is the following; do I need to open port 5061 UDP on my firewall or just port 5061 TCP for SIP sessions.? I am not interested in securing RTP only SIP sessions. Thanks for your help! -- _ -- 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
[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
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] Asterisk 13.11.2 unable to register on Centos 7 64bit
Thank you for your help! Centos 7 firewall was enable. systemctl stop firewalld issue fixed. Thanks, On Thu, Oct 13, 2016 at 3:54 PM, Victor Villarreal wrote: > 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 > -- Thanks for your support, 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] Asterisk 1.8 to Asterisk 13.11 appending area code to local numbers
Hello, I would like to add area code to local numbers, it worked like a charm on Asterisk 1.8 but does not work on Asterisk 13.11. Extensions.conf; worked before on Asterisk 1.8 ; Adding Area code to local numbers exten => _9XXX,n,Set(CALLERID(all)="$CallerID" <3818008000>) exten => _9XXX,n,Dial(SIP/voip-truck/1381${EXTEN:1},80) Any ideas? Thanks, Motty -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 to Asterisk 13.11 appending area code to local numbers
It does not append the area code: 5007-0047","SIP/truck1-0048","Dial","SIP/truck1/6052736,80","2016-11 -06 18:49:41",,"2016-11-06 18:49:45",4,0,"NO ANSWER","DOCUMENTATION","1478458181.738","" 5007-0049","SIP/truck1-004a","Dial","SIP/truck1/6052736,80","2016-11 -06 18:52:26",,"2016-11-06 18:52:34",7,0,"NO ANSWER","DOCUMENTATION","1478458346.826","" 5007-004b","SIP/truck1-004c","Dial","SIP/truck1/6052736,80","2016-11 -06 18:54:02",,"2016-11-06 18:54:23",20,0,"NO ANSWER","DOCUMENTATION","1478458442.914","" Our Sip provider takes 10 digits, it should have append 381 to 6052736 number. Thanks, Mottyh -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Sunday, November 06, 2016 12:15 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8 to Asterisk 13.11 appending area code to local numbers On Sun, Nov 6, 2016, at 03:35 PM, Motty Cruz wrote: > Hello, I would like to add area code to local numbers, it worked like > a charm on Asterisk 1.8 but does not work on Asterisk 13.11. > > > > Extensions.conf; worked before on Asterisk 1.8 ; Adding Area code to > local numbers > > exten => _9XXX,n,Set(CALLERID(all)="$CallerID" <3818008000>) > > exten => _9XXX,n,Dial(SIP/voip-truck/1381${EXTEN:1},80) This should also work fine. You'll need to provide the console output of an attempt. Different logic may be executed. -- Joshua Colp Digium, Inc. | Senior 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 -- 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 -- _ -- 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] T1 -Asterisk server - Analog lines
Hello All, The problem I'm facing is the following: two machines that require analog signal. However, connection to the world is setup as follow: T1 connection to Asterisk server Any suggestions on how to convert digital signal to analog? Type of card on my Asterisk server is wct4xxp Thanks for your support! Motty -- _ -- 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] Polycom SoundStation IP 6000 does not register
can you provide the configuration on sip.conf file? Do you have the following settings under the account number or ext number? host=dynamic nat=yes for instance my configuration sip.conf file is as follow: [1005] type=friend context=sip-phone call-limit=1 trustrpid=no callerid="iuser 1005" disallow=all allow=ulaw allow=alaw username=1005 auth=md5 secret=819c8ebd2d1525235235325235 dtmfmode=rfc2833 host=dynamic nat=yes canreinvite=no Thanks, Motty From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Monday, December 19, 2016 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom SoundStation IP 6000 does not register 2016-12-19 16:26 GMT+01:00 Yves : Hi, I am pulling my hair for days now... I can´t get a PolyCom SoundStation IP 6000 (Conferencephone) to register with my Asterisk. There are no SIP Packets arriving at my asterisk at all... and it has nothing to do with a firewall or similar... Simple Question: Does anybody have a running SoundStation IP 6000 registerd with asterisk? yes, I've got several of them running. If so... would you please be so kind to tell me whats wrong with my setup? AsteriskServer: 192.168.1.211 SIP-user: 165 (the SIP-Settings on asterisk-side are OK, tested with a normal Softphone... registering and placing calls is no problem...) The phone-log only says: "Registration failed User: 165, Error Code:480 Temporarily not available" I tried with newest firmware, resetting to factory 100 times, using a provisionig file (which the SoundStation correctly downloads) but it is always the same... the SoundStation does not contact the asterisk for registering... 1. Do you have any switch able to mirror traffic sent and received by Polycom phone ? Capturing such traffic would help to understand what's happening. 2. Some phones support zero touch config with which they download their config files from the Internet. Are you sure this doesn't happen ? 3. Is SNTP/NTP correctly configured on the phone ? Phoneversion: Telefoninformationen Telefonmodell SoundStation IP 6000 Teilenummer 3111-15600-001 Rev:W MAC-Adresse 00:04:F2:07:0C:D3 IP-Adresse 192.168.0.13 UC-Softwareversion 4.0.11.0583 BootROM-Softwareversion 5.0.5.2324 I can ping the phone from the asterisk, the phone can reach the asterisk server (as it downloads the tftp files, if used with a provisioning profile), so the route and everything is correct... I even connected another Hardphone on the same cable that stuck in the Polycom... no problem... the other phone can register and works, so there is really no cable or firewall related problem here... it must be a setting! thank you -- _ -- 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 -- _ -- 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] Polycom SoundStation IP 6000 does not register
Yves, I have a SoundStation IP 6000: My sip.conf [1006] type=friend context=sip-phone call-limit=1 trustrpid=no callerid="Conference Room3" <1006> disallow=all allow=ulaw allow=alaw username=1006 secret=secret1 dtmfmode=rfc2833 host=dynamic mailbox=1000 nat=yes canreinvite=no Asterisk server IP XX.XX.42.16 (Public) Client IP: 192.168.1.56 I would look for typos in your configuration! -Motty From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Thursday, December 22, 2016 6:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom SoundStation IP 6000 does not register Do you have any LLDP or CDP enabled anywhere ? 2016-12-21 19:50 GMT+01:00 Victor Villarreal : Hi Yves, Maybe your switch put your Polycom inside a Voice VLAN, based on the MAC of the phone. Maybe with the snom this not happen because your switch don't see the MAC of the Snom as a "supperted IP Phone". 2016-12-21 13:59 GMT-03:00 Yves : sorry... typo the problematic phone has the 192.168.0.13 the asterisk has 192.168.1.211 when i connect a snom phone on the cable that was in the soundstation 6000 before and configure the phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP... it would be helpful if someone, that has a running soundstation ip 6000 could send the configuration... :-/ regards, yves Am 21.12.2016 um 15:13 schrieb Mauricio Tavares: On Wed, Dec 21, 2016 at 7:50 AM, Yves wrote: Hi Mark, yes, you are right... these are different VLANs I configured the other phone to use the same IP (192.168.1.13)... and it worked flawlessly... on the SAME Networkcable in the same plug... so it must have something to do with the polycom phone config... remember... when I use tcp the phone tries to register, but does not even try with udp... thank you, yves I am a bit confused: is your problematic phone's IP 192.168.0.13 (what the error log is reporting below) or 192.168.1.13? Am 21.12.2016 um 13:34 schrieb Mark Wiater: Yves, Didn't you say that AsteriskServer: 192.168.1.211 SIP-user: 165 ? On 12/21/2016 4:24 AM, Yves wrote: . It is sure for 100% that there is no firewall or something else mangeling in between... another Hardphone works as expected using the same Netzworkcable on the same Networkplug with UDP on Port 5060... This other hardphone, what IP does it have? 50.848|cfg |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask 255.255.255.0 The line above suggests to me that your phone and your asterisk server are on a different network, there has to be something that routes between those two networks. Often what routes, can firewall. 000122.941|sip |4|03|Registration failed User: 165, Error Code:480 Temporarily not available Mark -- _ -- 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 -- _ -- 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 -- GnuPG Key ID: 0x39BCA9D8 https://www.github.com/mefhigoseth ...:::[ God Rulz ! ]:::... -- _ -- 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 -- _ -- 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] how to add area code to outgoing number in Asterisk 13.13
Hi, my SIP provider requires 10 digits for all outgoing calls; Users dial 7 digits for outgoing. Here is how I added the area code to all outgoing calls in Asterisk 1.8 Extensions.conf ; Adding Area code and striping 7 for local numbers exten => _7XXX,n,Set(CALLERID(all)="My ID" ) exten => _7XXX,n,Dial(SIP/mySIPprovider/1731${EXTEN:1},80) This syntax does not work in Asterisk 13.13. has anybody dealt with this issue? Thanks Motty -- _ -- 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] how to add area code to outgoing number in Asterisk 13.13
Thank you Carlos, you’re right I am using PJSIP. Should I not use it? Thanks, Motty From: Carlos Chavez [mailto:cur...@telecomab.mx] Sent: Saturday, December 31, 2016 5:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Motty Cruz Subject: Re: [asterisk-users] how to add area code to outgoing number in Asterisk 13.13 On 2016-12-29 12:44, Motty Cruz wrote: Hi, my SIP provider requires 10 digits for all outgoing calls; Users dial 7 digits for outgoing. Here is how I added the area code to all outgoing calls in Asterisk 1.8 Extensions.conf ; Adding Area code and striping 7 for local numbers exten => _7XXX,n,Set(CALLERID(all)="My ID" ) exten => _7XXX,n,Dial(SIP/mySIPprovider/1731${EXTEN:1},80) This syntax does not work in Asterisk 13.13. has anybody dealt with this issue? Thanks Motty The syntax has not changed so it should work. You are still using chan_sip right? Maybe you are using PJSIP which does have a different syntax? It is probably better if you send the output from the cli so we can see what error the dialplan is throwing at you. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez dCAP #1349 +52 (55)8116-9161 -- _ -- 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] T1 -Asterisk server - Analog lines
Thanks for your suggestion! After some research I purchased the AdTran Total Access 904 2nd. We have a T1 connection to our Asterisk server and from my Asterisk server to AdTran. I am in dead in the water, I cant figure out how to configure AdTran to talk to the Asterisk server. PRI à Asterisk àAdTran (total access 904 2nd gen) Calls are being forward to AdTran from Asterisk! Thanks, Motty From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Tuesday, December 06, 2016 3:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 -Asterisk server - Analog lines On 12/06/2016 06:45 PM, Motty Cruz wrote: Any suggestions on how to convert digital signal to analog? I do this with an ADIT 600 channel bank that I got off of ebaY many years ago. A quick search shows one for around $98 http://www.ebay.com/itm/like/191565965269?lpid=82 <http://www.ebay.com/itm/like/191565965269?lpid=82&chn=ps&ul_noapp=true> &chn=ps&ul_noapp=true Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 13.13.1
Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are starting to complaint about packets loss, conversations are choppy! I don't even know where to start looking! Choppy conversations happened within users. I am using sip.conf [1091] type=friend context=sip-phone call-limit=2 trustrpid=no callerid="dev1" <1091> disallow=all allow=ulaw allow=alaw username=1091 secret=X dtmfmode=rfc2833 host=dynamic mailbox=10091@default nat=force_rport,comedia canreinvite=no extensions.conf exten => 1091,hint,SIP/${EXTEN} exten => 1091,1,Dial(SIP/${EXTEN},15,t) exten => 1091,2,Voicemail(${EXTEN}@default,u) exten => 1091,102,Voicemail(${EXTEN}@default,b) exten => 1091,103,Hangup [2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission 7c803889-63e1b3fe-c2b5ef77@192.168.0.191 for seqno 156 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response any ideas? Thanks! Motty -- _ -- 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 13.13.1
Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from here: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz I continue to see errors like this: [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission 56849706-ba96a6d9-817305d0@192.168.125.173 for seqno 109 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission 6e3dd238-911e2ac3-f1260152@192.168.125.152 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission ed38f9c8-295a9db-c23f5242@192.168.125.144 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission ef497d11-a81b1c00-8bfbd3bf@192.168.1.244 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Before upgrading to this new server, Asterisk version 1.8 on CentOS 5.9 hardware on both servers were similar in CPU, Memory Any support on this matter is appreciated! Thanks, Motty From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of kambiz sharifi Sent: Saturday, January 28, 2017 5:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13.13.1 On Wed, Jan 25, 2017 at 16:00 Olivier wrote: What did you exactly upgade ? Asterisk only ? Asterisk and OS ? How did you installed Asterisk 1.8 and 13 ? From source or from package ? I would be curious to see what would happen after downgrading back to 1.8. 2017-01-24 21:03 GMT+01:00 Motty Cruz : Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are starting to complaint about packets loss, conversations are choppy! PkEI don’t even know where to start looking! Choppy conversations happened within users. I am using sip.conf [1091] type=friend context=sip-phone call-limit=2 trustrpid=no callerid="dev1" <1091> disallow=all allow=ulaw allow=alaw username=1091 secret=X dtmfmode=rfc2833 host=dynamic mailbox=10091@default nat=force_rport,comedia canreinvite=no extensions.conf exten => 1091,hint,SIP/${EXTEN} exten => 1091,1,Dial(SIP/${EXTEN},15,t) exten => 1091,2,Voicemail(${EXTEN}@default,u) exten => 1091,102,Voicemail(${EXTEN}@default,b) exten => 1091,103,Hangup [2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission 7c803889-63e1b3fe-c2b5ef77@192.168.0.191 for seqno 156 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response any ideas? Thanks! Motty -- _ -- 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 -- _ -- 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 -- _ -- 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 13.13.1
I thought it was a firewall issues. I disabled IP Tables & Selinux, but the problem persist! I have not made changes on our firewall since the upgrade! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Monday, January 30, 2017 9:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13.13.1 >>> On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.c...@gmail.com wrote: >>> Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from here: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar .gz >>> I continue to see errors like this: >>> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission 56849706-ba96a6d9-817305d0@192.168.125.173 for seqno 109 (Critical Request) -- See >>> >>> Firewall? Doug -- _ -- 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 -- _ -- 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] asterisk 13.13.1 Everyone is busy-congested at this time (1:1/0/0)
Hi, my server is running a fresh install of Asterisk 13.13.1 on CentOS 7. My extensions.conf file was mostly copied from server running Asterisk 1.8. That being said! If I dial a number and get a busy signal I get the following error: -- SIP/voipeer-084b redirecting info has changed, passing it to SIP/1007-084a -- SIP/voipeer-084b is busy == Everyone is busy/congested at this time (1:1/0/0) -- Timeout on SIP/1007-084a -- Executing [t@phones:1] Playback("SIP/1007-084a", "goodbye") in new stack > 0x7f6a62146640 -- Probation passed - setting RTP source address to 191.96.18.41:62568 -- Playing 'goodbye.slin' (language 'en') > 0x7f6a62146640 -- Probation passed - setting RTP source address to 191.96.18.41:62568 -- Executing [t@phones:2] Hangup("SIP/1007-084a", "") in new stack Sip.conf [1007] type=friend context=sip-phone call-limit=2 trustrpid=no callerid="dev1" <1007> disallow=all allow=ulaw allow=alaw username=1007 secret=X dtmfmode=rfc2833 host=dynamic mailbox=1007@default nat=force_rport,comedia Is it a codec issue? Or missed configuration? Asterisk does not know how to translate busy signal. Your help is appreciated! Thanks, -- _ -- 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] Soft SIP phones that support TLS - Asterisk version 13.13.1
Hello, I have a user that prefers Soft SIP phone install on his laptop, for security reasons I have enable TLS on our Asterisk server to support TLS authentication, It works well with hard phones. Has anybody in this forum use SIP Soft phones with TLS authentication enabled? Any suggestions? Thanks, Motty -- _ -- 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] fail2ban Asterisk 13.13.1
Hello, fail2ban does not ban offending IP. NOTICE[29784] chan_sip.c: Registration from '"user3"' failed for 'offending-IP:53417' - Wrong password NOTICE[29784] chan_sip.c: Registration from '"user3"' failed for ‘offending-IP:53911' - Wrong password systemctl status fail2ban ● fail2ban.service - Fail2Ban Service Loaded: loaded (/usr/lib/systemd/system/fail2ban.service; enabled; vendor preset: disabled) Active: active (running) since Wed 2017-03-01 00:40:43 PST; 470min ago Docs: man:fail2ban(1) jail.local [DEFAULT] # "bantime" is the number of seconds that a host is banned. bantime = -1 # A host is banned if it has generated "maxretry" during the last "findtime" # seconds. findtime = 300 # "maxretry" is the number of failures before a host get banned. maxretry = 3 [asterisk-iptables] enable = true port = 5060,5061 filter = asterisk action = iptables-allports[name=ASTERISK, protocol=all] sendmail[name=ASTERISK, dest=mo...@email.com, sender=fail2...@asterisk-ip.com] #action = %(banaction)s[name=%(__name__)s-tcp, port="%(port)s", protocol="tcp", chain="%(chain)s", actname=%(banaction)s-tcp] %(banaction)s[name=%(__name__)s-udp, port="%(port)s", protocol="udp", chain="%(chain)s", actname=%(banaction)s-udp] %(mta)s-whois[name=%(__name__)s, dest="%(destemail)s"] logpath = /var/log/asterisk/messages maxretry = 3 findtime = 300 bantime = -1 in filter.d asterisk.conf failregex = ^%(__prefix_line)s%(log_prefix)s Registration from '[^']*' failed for '(:\d+)?' - (Wrong password|Username/auth name mismatch|No matching peer found|Not a local domain|Device does not match ACL|Peer is not supposed to register|ACL error \(permit/deny\)|Not a local domain)$ ^%(__prefix_line)s%(log_prefix)s Call from '[^']*' \(:\d+\) to extension '[^']*' rejected because extension not found in context ^%(__prefix_line)s%(log_prefix)s Host failed to authenticate as '[^']*'$ ^%(__prefix_line)s%(log_prefix)s No registration for peer '[^']*' \(from \)$ ^%(__prefix_line)s%(log_prefix)s Host failed MD5 authentication for '[^']*' \([^)]+\)$ ^%(__prefix_line)s%(log_prefix)s Failed to authenticate (user|device) [^@]+@\S*$ ^%(__prefix_line)s%(log_prefix)s hacking attempt detected ''$ ^%(__prefix_line)s%(log_prefix)s SecurityEvent="(FailedACL|InvalidAccountID|ChallengeResponseFailed|InvalidPa ssword)",EventTV="([\d-]+|%(iso8601)s)",Severity="[\w]+",Service="[\w]+",Eve ntVersion="\d+",AccountID="(\d*|)",SessionID=".+",LocalAddress="IPV [46]/(UDP|TCP|WS)/[\da-fA-F:.]+/\d+",RemoteAddress="IPV[46]/(UDP|TCP|WS)//\d+"(,Challenge="[\w/]+")?(,ReceivedChallenge="\w+")?(,Response="\w+",Ex pectedResponse="\w*")?(,ReceivedHash="[\da-f]+")?(,ACLName="\w+")?$ ^%(__prefix_line)s%(log_prefix)s "Rejecting unknown SIP connection from "$ ^%(__prefix_line)s%(log_prefix)s Request (?:'[^']*' )?from '[^'] *' failed for '(?::\d+)?'\s\(callid: [^\)]*\) - (?:No matching endpoint found|Not match Endpoint(?: Contact)? ACL|(?:Failed|Error) to authenticate)\s*$ failregex = NOTICE.* .*: Registration from '.*' failed for '' - Wrong password NOTICE.* .*: Registration from '.*' failed for ':.*' - No matching peer found NOTICE.* .*: Registration from '.*' failed for '' - No matching peer found NOTICE.* .*: Registration from '.*' failed for '' - Username/auth name mismatch NOTICE.* .*: Registration from '.*' failed for '' - Device does not match ACL NOTICE.* .*: Registration from '.*' failed for '' - Peer is not supposed to register NOTICE.* .*: Registration from '.*' failed for '' - ACL error (permit/deny) NOTICE.* .*: Registration from '.*' failed for '' - Device does not match ACL NOTICE.* failed to authenticate as '.*'$ NOTICE.* .*: No registration for peer '.*' \(from \) NOTICE.* .*: Host failed MD5 authentication for '.*' (.*) NOTICE.* .*: Failed to authenticate user .*@.* NOTICE.* .*: Sending fake auth rejection for device .*\\>;tag=.* NOTICE.* .*: Registration from '\".*\".*' failed for '' - No matching peer found NOTICE.* .*: Registration from '\".*\".*' failed for '' - Wrong password ignoreregex = Thanks Motty -- _ -- 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] CDR reporting solution
Hello, I am looking for CDR reporting solution? Any suggestions? I am using Asterisk 13.13.1 I would like a report on number of calls per extension. Thanks, Motty -- _ -- 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] CDR reporting solution
I installed CDR-Stats on Debian 8.7 http://cdr-stats.readthedocs.io/en/latest/installation/install-cdr-stats.html I am trying to figure out how to import flat CSV file to CDR-Stats On Wed, Mar 22, 2017 at 6:28 PM, Bruce Ferrell wrote: > How about CDR to either MySQL or cdrlite and a quickie sql query? I could > have added postgres, but I'm a DB bigot. That would work too. > > > On 03/22/2017 01:46 PM, Motty Cruz wrote: > >> >> Hello, I am looking for CDR reporting solution? Any suggestions? I am >> using Asterisk 13.13.1 >> >> I would like a report on number of calls per extension. >> >> Thanks, >> Motty >> >> >> >> > > -- > _ > -- 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 > -- Thanks for your support, Motty -- _ -- 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] how to hangup this channel "Message/ast_msg_queu
omega*CLI> core show channels Channel Location State Application(Data) Message/ast_msg_queu 4002@sipphones:2 Up VoiceMail(4002@default,u) "Message/ast_msg_queu" it's been up for the last day, how to hangup this channel? -- _ -- 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] Asterisk 13.13.1 use_callids = yes Extensions ID as CallerID
Hello, In Master.csv Asterisk is loggin the Company ID set in Extensions.conf, but I configured logger.conf to log the EXT ID. For instance, the SRC in the following line should be my ext. number. Does it make sense? From my extension 4007 I called 78079745, yet in the log below the first number is 2318001800 which is the main company's number set in Extensions.conf. 2318001800 78079745 phones "ITadmin" <2318001800> SIP/4007-00015c0a SIP/voip1-00015c0b Dial SIP/voip1/78079745,80 4/3/2017 15:30 4/3/2017 15:31 2 0 NO ANSWER DOCUMENTATION 1.49E+09 Logger.conf [general] dateformat=%F %T ; ; Customize the display of debug message time stamps ; this example is the ISO 8601 date format (-mm-dd HH:MM:SS) ; ; see strftime(3) Linux manual for format specifiers. Note that there is also ; a fractional second parameter which may be used in this field. Use %1q ; for tenths, %2q for hundredths, etc. ; ;dateformat=%F %T ; ISO 8601 date format ;dateformat=%F %T.%3q ; with milliseconds dateformat = %F %T.%3q ; ISO 8601 date format with milliseconds ; ; ; This makes Asterisk write callids to log messages ; (defaults to yes) use_callids = yes -- _ -- 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 13.13.1 use_callids = yes Extensions ID as CallerID
Hello, I am trying to match SRC in CSV file to clid - Caller ID to user extension number for stats purposes, however, in CSV file the SRC is the company number as set in Extensions.conf exten => _7XXX,1,Set(CALLERID(number)="Company Inc" <3788001800>) exten => _7XXX,2,Dial(SIP/voip1/13781${EXTEN:1},80) exten => _7XXX,n,Congestion() exten => _7XXX,n,Hangup() how would I change it? I have look in cdr.conf and logger.conf Thanks, From: Motty Cruz [mailto:motty.c...@gmail.com] Sent: Monday, April 03, 2017 3:52 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: motty.c...@gmail.com Subject: [asterisk-users] Asterisk 13.13.1 use_callids = yes Extensions ID as CallerID Hello, In Master.csv Asterisk is loggin the Company ID set in Extensions.conf, but I configured logger.conf to log the EXT ID. For instance, the SRC in the following line should be my ext. number. Does it make sense? From my extension 4007 I called 78079745, yet in the log below the first number is 2318001800 which is the main company's number set in Extensions.conf. 2318001800 78079745 phones "ITadmin" <2318001800> SIP/4007-00015c0a SIP/voip1-00015c0b Dial SIP/voip1/78079745,80 4/3/2017 15:30 4/3/2017 15:31 2 0 NO ANSWER DOCUMENTATION 1.49E+09 Logger.conf [general] dateformat=%F %T ; ; Customize the display of debug message time stamps ; this example is the ISO 8601 date format (-mm-dd HH:MM:SS) ; ; see strftime(3) Linux manual for format specifiers. Note that there is also ; a fractional second parameter which may be used in this field. Use %1q ; for tenths, %2q for hundredths, etc. ; ;dateformat=%F %T ; ISO 8601 date format ;dateformat=%F %T.%3q ; with milliseconds dateformat = %F %T.%3q ; ISO 8601 date format with milliseconds ; ; ; This makes Asterisk write callids to log messages ; (defaults to yes) use_callids = yes -- _ -- 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] upgrading asterisk 13.13.1 to latest version best practices
Hello, Best practices examples to upgrade Asterisk 13.13.1 to latest version? Any suggestions? Thanks, Motty -- _ -- 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] choppy calls version Asterisk 13.17 on CentOS 7
Hello, A few months ago we upgraded our server from Asterisk 1.8.22.0 on CentOS 5.9 to Asterisk 13.13.1 on CentOS 7. We are still using SIP not PJSIP. Since the upgrade our remote users' conversions are choppy. Here is what my sip.conf looks like for the users with most problems: [user-name] type=friend context=sip-phone trustrpid=no call-limit=2 callerid="Mike" <1008> disallow=all allow=g726 allow=g722 allow=g723 allow=ulaw allow=alaw username=user-name secret= dtmfmode=rfc2833 host=dynamic mailbox=1008 nat=force_rport,comedia canreinvite=no any ideas? Monitoring using CLI, I noticed the device always select ulaw for codec. Thanks, Motty -- _ -- 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