Re: [asterisk-users] is g729 codec free? or under license???
Darryl Moore moores.ca> writes: > > > I'll explain. > > The g.729 compression algorithm is not protected by copyright, though > specific instances may be. It is protected by a patent. > > http://www.sipro.com/G-729.html > > An open source version is available here: > > http://asterisk.hosting.lv/ > > What stops you from using this, or even your own implementation isn't > copyright, but patent protection. It is the right to use the patented > technology that you are licensing, not the particular copyrighted coded > that implements it. > > Here you will find the various G.729 patents which were all granted in > 1996. > > https://www.itu.int/ITU-T/recommendations/related_ps.aspx?id_prod=3334 > > I had thought these expired next year because I was thinking patents > were only 18 years. Turns out they are now 20 years, so they really do > not expire til some time in 2016. My bad. > > So in countries that honour software patents, you need to have a license > until some time in 2016. In countries which do not, you are free to use > these open source codes now. > > cheers. Darrel- The G729 essential patents were *granted* in 1996, but applied for prior to June 8 1995. That means their lifespan is either 20 years from their application date, or 17 years from their grant date, whichever is greater (http://www.uspto.gov/main/faq/p120013.htm). Either way, they expire in 2014. -Jeff -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 Trunk Encryption
Any ideas? Still hoping..
On Sun, Apr 6, 2014 at 12:03 AM, Elliott W wrote:
> I have.
>
> On the receiving side I had gotten:
> [2014-04-05 23:28:12] WARNING[1832] chan_iax2.c: Rejected connect attempt.
> No secret present while force encrypt enabled.
>
> I had no secret because I was using RSA authentication and didn't think I
> needed it, so I added EXACTLY the same line on both sides (copy/paste).
> Now I get:
> [2014-04-05 23:30:42] NOTICE[1832] chan_iax2.c: Call Terminated, Incoming
> call is unencrypted while force encrypt is enabled.
>
> On the sending side I really get nothing useful:
> [2014-04-05 23:30:42] VERBOSE[2795][C-0002] pbx.c: -- Executing
> [s@macro-dialout-trunk:22] Dial("SIP/comp-in-ch01-0001", "
> IAX2/ch01_ch02/1234,300,Ttr") in new stack
> [2014-04-05 23:30:42] VERBOSE[2795][C-0002] app_dial.c: -- Called
> IAX2/ch01_ch02/1234
> [2014-04-05 23:30:43] VERBOSE[2795][C-0002] chan_iax2.c: -- Hungup
> 'IAX2/ch01_ch02-17634'
> [2014-04-05 23:30:43] VERBOSE[2795][C-0002] app_dial.c: == Everyone is
> busy/congested at this time (1:0/0/1)
> I modified the extension and the trunk name for security reasons, but
> without force encryption calls flow back and forth easily.
>
> These three directives exist on both sides:
> encryption=yes
> forceencryption=yes
> secret=mysecretcode
>
> So I'm kind of at a loss, I can see the options set, I can see:
> [2014-04-05 23:59:32] VERBOSE[1832] chan_iax2.c: -- Accepting
> AUTHENTICATED call from xxx.yyy.zzz.aaa:
> when I DON'T have the force encryption set, so I can't see what else I
> need to do..
>
> CEW
>
>
>
>
> On Fri, Apr 4, 2014 at 7:07 PM, Steve Totaro <
> stot...@totarotechnologies.com> wrote:
>
>> Have you enabled IAX2 debugging and tried some test calls?
>>
>> Thanks,
>> Steve T
>>
>>
>>
>> On Fri, Apr 4, 2014 at 6:59 PM, Elliott W wrote:
>>
>>> That answered my question as to whether it WAS encrypted, I think, and
>>> the answer is no, the credentials are but all the rest is not. That just
>>> leaves the question of what I need to do to get it encrypted..
>>>
>>> Thanks.
>>>
>>>
>>> On Fri, Apr 4, 2014 at 12:59 PM, Steve Totaro <
>>> stot...@totarotechnologies.com> wrote:
>>>
Wireshark.
On Fri, Apr 4, 2014 at 11:13 AM, Elliott W >>> > wrote:
> Ok, I think I am 90%+ there.
>
> Note: the configuration or status is the same on both sides unless
> otherwise noted.
>
> I am using RSA keys for authentication and the calls are coming
> through as authenticated so I'm sure that part works.
>
> The peer shows the "(E)" next to the status in Asterisk Info for the
> IAX2 peers
>
> The trunk configuration contains:
> encryption=yes
>
> So here is my question, Calls stop flowing when I use the directive:
> forceencryption=yes
> At the trunk level or higher does not matter, same effect.
>
> So my question comes down to, are my calls getting encrypted and why
> does this directive cause them to fail, AND how can I tell.
>
> Thanks.
>
>
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>>>
>>>
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>>
>>
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>
>
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Re: [asterisk-users] asterisk-users Digest, Vol 117, Issue 7
Hi Patrick, Thanks a lot for your quick help. Yes, I configured the NAT options in sip.conf. BTW, I am using 12.1.1, will try 11.8.1 and see if I can make it work. Thanks again, William === Date: Sat, 05 Apr 2014 23:38:32 +0200 From: Patrick Laimbock To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk and SRTP Message-ID: <534077d8.7000...@laimbock.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 04/05/2014 07:56 PM, William Wu wrote: >Hi experts, > > I am trying Asterisk SRTP in my environment, and find that when >Asterisk is behind a NAT, the audi/video UDP ports opened for SRTP relay >by Asterisk are local ports on the Asterisk server, media from the two >clients out of the NAT (for example from Internet) can not reach the >ports, and thus the two client can not establish the secure call via >Asterisk. I have set up a STUN server and configured in rtp.conf, but >seems Asterisk does not do STUN before it opens ports for SRTP. BTW, >Non-SRTP call can work though. > >Anyone can give advice on how to make SRTP work in such an env? I have no problems with a TLS/SRTP call between a GSM with CSipSimple and Asterisk 11.8.1 behind NAT. Have you configured the NAT options in sip.conf? externip=... localnet=... nat=... You may also need to add/change the options below. Check the sip.conf example file to see what these options do and use what's best for your situation. canreinvite=no directmedia=no directrtpsetup=no HTH, Patrick > -- _ -- 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] Need to hire recordings for an IVR
On 07/04/2014 11:29 AM, CDR wrote: I wonder if anybody know how to hire Alice or some professional voice-artist. I need to record 12 messages for a customer. We have had good success with a local sound studio that uses radio personalities for recording. I like radio announcers for the following: - good quality - fast turnaround - can read and understand a script and get it right the first time - ability to find the talent again if you need re-recording. - neutral accent Ron -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Need to hire recordings for an IVR
> I wonder if anybody know how to hire Alice or some professional > voice-artist. I need to record 12 messages for a customer. Assuming you mean Allison, her information is here: http://www.digium.com/en/products/ivr/allison-smith-- _ -- 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] Need to hire recordings for an IVR
I wonder if anybody know how to hire Alice or some professional voice-artist. I need to record 12 messages for a customer. -- _ -- 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] additional range parameter for sip peer
Am 29.03.2014 11:12, schrieb Thomas Rechberger: Many ITSP are using loadbalancers, so if somebody registers on a sip peer with specific dns host, an incoming call may be received from a different ip and the host value in peer section doesnt match, so it will go to default context. For example Telekom or 1&1, biggest providers in Germany are using too many different addresses that its not practical to define them all (up to 50 hosts and they still add!), as this will also generate too much traffic (especially with qualify and multiple registrations) and they may even lock you out as untrusted, which may even result in that they will block asterisk permanently for everybody. Thats not really desirable. I think its also not recommended in terms of security to use default context with allowguest=yes and sort the incoming calls by header, because this can be faked easily. From my understanding the permit/deny parameters are only used for incoming calls if host is set to dynamic and then there will be no outgoing registration to remote peer possible. permit/deny is used for access, not for matching. How about an additional parameter where an range of ip addresses can be defined in peer section, which will be used for matching calls? hostmatchrange=x.x.x.x/24 anyone here? What do you think about using permit/deny for host matching? -- _ -- 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
Re: [asterisk-users] how to stop asterisk using a call
thanks a lot it works correctly 2014-04-07 12:08 GMT+00:00 Andres : > On 4/7/14, 4:53 AM, Salaheddine Elharit wrote: > > hello list, > > i have a question i don't know if there is any possibility to stop > asterisk using a call for exp: > > when i call a number 0522xx i want to excute a script or any idea to > stop asterisk automatically > > Sure, try something like: > [custom-stop] > exten => 052212345,1,System(sudo /usr/sbin/service asterisk stop) > > (you need to give the asterisk owner permission to execute 'service' > comand via sudo) > > i use asterisk 1.4.43 > > NB: with mysql using a database i can insert into table using php > without issue. but now with SSH how can i do > > thanks and regards. > > > > > -- > 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] how to stop asterisk using a call
On 4/7/14, 4:53 AM, Salaheddine Elharit wrote: hello list, i have a question i don't know if there is any possibility to stop asterisk using a call for exp: when i call a number 0522xx i want to excute a script or any idea to stop asterisk automatically Sure, try something like: [custom-stop] exten => 052212345,1,System(sudo /usr/sbin/service asterisk stop) (you need to give the asterisk owner permission to execute 'service' comand via sudo) i use asterisk 1.4.43 NB: with mysql using a database i can insert into table using php without issue. but now with SSH how can i do thanks and regards. -- Technical Support http://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
[asterisk-users] how to stop asterisk using a call
hello list, i have a question i don't know if there is any possibility to stop asterisk using a call for exp: when i call a number 0522xx i want to excute a script or any idea to stop asterisk automatically i use asterisk 1.4.43 NB: with mysql using a database i can insert into table using php without issue. but now with SSH how can i do thanks and regards. -- _ -- 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
