[asterisk-users] problem getting dahdi-linux to work with kernel 6.1.0-10
Hi. I have run into a problem compiling dahdi-linux in kernel 6.1.0-10. Apparently there was a change, so I found a patch to fix stdbool.h but now I have an implicit declaration of pci_alloc_consistent in drivers/dahdi/wct4xxp/base.c I don't see any other references to that name anywhere. I am using version from git 5c840cf43838e0690873e73409491c392333b3b8 . So, the question, how to fix, so I can get the tompile to work? Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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 IP PBX VoIP Servers Hacked by Hackers
I am using freepbx latest 16 version -- am I subject to this problem? I am not using elastics, but I installed on a Debian bullseye server, so this is of definite concern to me. Thanks. On Mon, 18 Jul 2022 06:45:41 -0400, Joshua C. Colp wrote: > > [1 ] > [1.1 ] > On Mon, Jul 18, 2022 at 7:43 AM Turritopsis Dohrnii Teo En Ming < > c...@teo-en-ming.com> wrote: > > > > > Dear Joshua Colp, > > > > Noted with thanks. So the vulnerability is not related to the Asterisk > > open source project at all? > > > > It is not. The vulnerability mentioned is regarding FreePBX and Elastix, > which do use Asterisk but the vulnerability has nothing to do with Asterisk > itself. > > -- > Joshua C. Colp > Asterisk Project Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > [1.2 ] > [2 ] > -- > _ > -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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 detect which confbridge user is talking or muted
Hi. Is there a way in confbridge where I can enquire if a channel is muted, or if the channel is talking? There seems to be nothing except ami events, but I would just like to check a channel to see if he is talking or muted at a particular time and display that information on the console. I have been using meetme and there you can just display the list of users and you get that information. Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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] [External] a couple of problems with confbridge
OK, thanks, that is what I was hoping for. On Fri, 01 Jul 2022 12:02:46 -0400, Dan Cropp wrote: > > I believe the answer #2 depends on the user options for each participant. > > If all participants have user options with wait for marked set to true there > will be no conference/recording until at least one marked user joins. > If any participants have user options with wait for marked set to false, when > they join the conference bridge it is actually going. Thus, if the bridge > options had the record enabled it would start recording. > If only marked user joins first, it's met the criteria and will conference > and start recording. > > Dan > > -Original Message- > From: asterisk-users On Behalf Of > John Covici > Sent: Tuesday, June 28, 2022 6:28 PM > To: asterisk-users@lists.digium.com > Subject: [External] [asterisk-users] a couple of problems with confbridge > > Hi. I have been using meetme for years, but I wanted to try > confbridge as meetme is going away soon.I am having a few > problems/questions doing this. > > 1. When I list the confbridge users in a bridge, I only get the caller id > number -- I have a number of contacts in contact manager and I am using > superfecta, but the name does not appear. I do need the name to see who is > on there. > > 2. I will be using a conference with a marked user -- and I would like to > record the conference -- when does the recording start -- when the first user > comes on or when the marked user joins? > > 3. In the sample file it says you cannot have more than one user profile on > a bridge, but I need two, one for the marked user and another one for regular > users -- how do I work around this? > > Thanks in advance for any suggestions. > > > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici wb2una > cov...@ccs.covici.com > > -- > _ > -- 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 > -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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] a couple of problems with confbridge
On Tue, 28 Jun 2022 19:54:11 -0400, Joshua C. Colp wrote: > > [1 ] > On Tue, Jun 28, 2022 at 8:28 PM John Covici wrote: > > > Hi. I have been using meetme for years, but I wanted to try > > confbridge as meetme is going away soon.I am having a few > > problems/questions doing this. > > > > 1. When I list the confbridge users in a bridge, I only get the > > caller id number -- I have a number of contacts in contact manager and > > I am using superfecta, but the name does not appear. I do need the > > name to see who is on there. > > > > You'll need to be specific on how you are listing. The AMI action provides > all of the information. ... I was using the confbridge list command from the console and that only gives the number -- any way to fix or is there some other way I could get this information on the console? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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] a couple of problems with confbridge
Hi. I have been using meetme for years, but I wanted to try confbridge as meetme is going away soon.I am having a few problems/questions doing this. 1. When I list the confbridge users in a bridge, I only get the caller id number -- I have a number of contacts in contact manager and I am using superfecta, but the name does not appear. I do need the name to see who is on there. 2. I will be using a conference with a marked user -- and I would like to record the conference -- when does the recording start -- when the first user comes on or when the marked user joins? 3. In the sample file it says you cannot have more than one user profile on a bridge, but I need two, one for the marked user and another one for regular users -- how do I work around this? Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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] GET DATA on AGI
I thought one of the arguments to the read command was the terminator, is that the command you have in your agi? On Sun, 27 Feb 2022 12:26:50 -0500, Tom Ray wrote: > > [1 ] > [1.1 ] > I believe that # in the default terminator for GET DATA and I don’t think > that can be disabled. But I’m not a 100% as I’ve always used # as the > terminator. > > > > From: asterisk-users On Behalf Of > Dovid Bender > Sent: Sunday, February 27, 2022 11:01 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [asterisk-users] GET DATA on AGI > > > > Hi, > > > When using GET DATA in an AGI it seems that the # key ends the input. So if > say I want the user to input 123#456 the system will return 123. I did not > see this in the documentation. Is this a bug, lack of documentation or do I > have a bug in my AGI? > > > > TIA. > > > > Dovid > > > > [1.2 ] > [2 ] > -- > _ > -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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] a few confbridge questions
Hi. I am using meetme application and I am interested in switching to confbridge, but there seems to be no way to do certain things in the dialplan with confbridge. How would I get the number of users in a particular conference? I want the leader to only start the recording when there are sufficient participants, which I will give him in an ivr. How would I increase or decrease the volume for a particular user in a conference? I can do these things using meetme, so I don't want to lose functionality when going to confbridge. Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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] strange sound on conference call
Hi. I am having a problem with a conference call on my server which a vps in the cloud. I am using chan_sip and meetme. What I get is a bit of a staticy or robotic sound, but it goes away if the user lowers the volume a bit which we can do with *4 in meetme. So, is the problem with the chan_sip, meetme or something else entirely? Nothing relevant in the logs. Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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 escape the & in BackGround
I have been using system commands in my dialplan for years and the &
goes through and puts the process in background like it should,
asterisk does not do anything, so you are left with what the shell
does.
On Thu, 27 Jan 2022 17:48:46 -0500,
Dovid Bender wrote:
>
> [1 ]
> [1.1 ]
> I tried tinyURL and that did not work. I got an error of:
> file.c:789 ast_openstream_full: File https://tinyurl.com/bdfye5ts9 does not
> exist in any format (URL changed to hide aws key). I tried adding
> \;foo=wav. but that did not work either.
>
>
> On Thu, Jan 27, 2022 at 3:32 PM Kingsley Tart wrote:
>
> > Does asterisk follow HTTP redirects? If so can you use something like
> > tinyurl.com to produce an alternative URL?
> >
> > Or, base64 encode the URL, and then set a variable with
> > Set(url=${BASE64_DECODE(${encodedURL})) ?
> >
> > Cheers,
> > Kingsley.
> >
> > On Wed, 2022-01-26 at 16:56 -0500, Dovid Bender wrote:
> > > I tried but it seems it does not.
> > >
> > >
> > > On Tue, Jan 18, 2022 at 2:57 PM John Runyon
> > > wrote:
> > > > ${SPRINTF(%c,38)}
> > > > or
> > > > %26
> > > >
> > > > should work, I think.
> > > >
> > > > On Sun, 16 Jan 2022 at 13:21, Dovid Bender
> > > > wrote:
> > > > > Hi,
> > > > >
> > > > > I am trying to play a sound file from AWS S3. The URL is
> > > > > something like this http://example.org?foo=bar=b. The issue
> > > > > seems to be that as soon as Asterisk see's the & it assumes there
> > > > > is a new file and the a=b is not sent along. I tried doing \& but
> > > > > that did not work. Does anyone know a way of telling Asterisk
> > > > > that & is part of the URL and to pass it along as a string?
> >
> >
> > --
> > _
> > -- 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
> [1.2 ]
> [2 ]
> --
> _
> -- 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
--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?
John Covici wb2una
cov...@ccs.covici.com
--
_
-- 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 and maybe a freepbx question
OK, that tells me something, I will disable pjsit for now, learn about it and try again. On Sun, 09 Jan 2022 06:39:55 -0500, John Harragin wrote: > > [1 ] > [1.1 ] > You can also set up multiple physical or vlan(ed) interfaces and bind sip > to one and pjsip to the other - then you have to set up the appropriate > interface routing too for both inbound and outbound packets which takes a > good understanding of your network topology and the locations of your > respective devices. You might be able to do it with multiple addresses on > your interface too (although I haven't tried it). > > All of the packets have to be presented to the appropriate channel > otherwise get discarded. You can't set it up so if a packet is from a > device not registered with pjsip, it gets passed to chan_sip to try. > > For me, I had both channel types running on production machines while I > migrated to pjsip or when not being able to figure out how to set up some > property in pjsip that you had running in sip. Each time I've had to do > this, eventually I was able get it all running within pjsip. I also already > had multiple vlans configured for my servers (with voip exclusive to one). > > The short story is that it is easier to learn how to get things working > within pjsip than learning the tricky networking setup. > > > On Sun, Jan 9, 2022 at 2:49 AM Duncan Turnbull > wrote: > > > > > > > > > > > > On 9/01/2022, at 7:11 PM, John Covici wrote: > > > > > > On Sat, 08 Jan 2022 19:17:57 -0500, > > > Antony Stone wrote: > > >> > > >>> On Sunday 09 January 2022 at 00:50:27, John Covici wrote: > > >>> > > >>> Hi. I am using asterisk 18.3 and freepbx. > > >> > > >> Hm, which version of FreePBX uses Asterisk 18.3? > > >> > > >>> How can both sip and pjsip be listening at port 5060 at the same time > > >> > > >> They can't. > > >> > > >> One might be on TCP and the other on UDP, but you can't have them both > > >> listening on the same port with the same protocol. > > > > In freepbx you enable chan sip or pjsip or both and set what ports they use > > > > The choices are either in advanced settings or sip settings > > > > Disable pjsip and reset the chan_sip port to 5060 or use pjsip. With them > > both enabled sometimes odd things happen but it will still work. You will > > get lots of error messages though > > > > > > >> > > >>> for instance I get: > > >>> > > >>> [2022-01-08 17:08:59] SECURITY[244351] res_security_log.c: > > >>> > > SecurityEvent="FailedACL",EventTV="2022-01-08T17:08:59.957-0500",Severity=" > > >>> > > Error",Service="PJSIP",EventVersion="1",AccountID="anonymous",SessionID="20 > > >>> 25076022",LocalAddress="IPV4/UDP/166.84.7.53/5060 > > ",RemoteAddress="IPV4/UDP/ > > >>> 45.134.144.118/5823 > > ",ACLName="registrar_attempt_without_configured_aors" > > >> > > >> What makes you think chan_sip and pjsip are both listening on UDP 5060? > > >> > > >>> I would like pjsit not to listen,till I figure out how to configure > > >>> the thing, so my logs don't fill up with messages. > > >>> > > >>> Thanks in advance for any suggestions. > > >> > > >> As far as I recall using FreePBX, there is a selector for the SIP > > protocol to > > >> tell it whether you want it to use pjsip or chan_sip. I don't think it > > even > > >> supports using both at the same time, so simply make sure that is set > > to > > >> chan_sip and you should be fine. > > >> > > >> On the other hand, why do you need to learn "how to configure the > > thing" if > > >> you're using FreePBX? Part of the whole point is that it does the > > fiddly > > >> techie sutff in the background for you, and you just need to use the > > personnel- > > >> department-friendly web GUI. > > > > > > This is what I thought as well, I just generated one trunk using the > > > old chan_sip and expected nothing from pjsit, yet I get all kinds of > > > errors like > > > [2022-01-08 17:08:59] WARNING[487628] res_pjsip_registrar.c: Endpoint > > > 'anonymous' (45.134.144.118:5823) has no configured AORs > > > > > > so I am very confused a
Re: [asterisk-users] asterisk and maybe a freepbx question
On Sat, 08 Jan 2022 19:17:57 -0500, Antony Stone wrote: > > On Sunday 09 January 2022 at 00:50:27, John Covici wrote: > > > Hi. I am using asterisk 18.3 and freepbx. > > Hm, which version of FreePBX uses Asterisk 18.3? > > > How can both sip and pjsip be listening at port 5060 at the same time > > They can't. > > One might be on TCP and the other on UDP, but you can't have them both > listening on the same port with the same protocol. > > > for instance I get: > > > > [2022-01-08 17:08:59] SECURITY[244351] res_security_log.c: > > SecurityEvent="FailedACL",EventTV="2022-01-08T17:08:59.957-0500",Severity=" > > Error",Service="PJSIP",EventVersion="1",AccountID="anonymous",SessionID="20 > > 25076022",LocalAddress="IPV4/UDP/166.84.7.53/5060",RemoteAddress="IPV4/UDP/ > > 45.134.144.118/5823",ACLName="registrar_attempt_without_configured_aors" > > What makes you think chan_sip and pjsip are both listening on UDP 5060? > > > I would like pjsit not to listen,till I figure out how to configure > > the thing, so my logs don't fill up with messages. > > > > Thanks in advance for any suggestions. > > As far as I recall using FreePBX, there is a selector for the SIP protocol to > tell it whether you want it to use pjsip or chan_sip. I don't think it even > supports using both at the same time, so simply make sure that is set to > chan_sip and you should be fine. > > On the other hand, why do you need to learn "how to configure the thing" if > you're using FreePBX? Part of the whole point is that it does the fiddly > techie sutff in the background for you, and you just need to use the > personnel- > department-friendly web GUI. This is what I thought as well, I just generated one trunk using the old chan_sip and expected nothing from pjsit, yet I get all kinds of errors like [2022-01-08 17:08:59] WARNING[487628] res_pjsip_registrar.c: Endpoint 'anonymous' (45.134.144.118:5823) has no configured AORs so I am very confused as to why this is happening. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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 and maybe a freepbx question
Hi. I am using asterisk 18.3 and freepbx. How can both sip and pjsip be listening at port 5060 at the same time, for instance I get: [2022-01-08 17:08:59] SECURITY[244351] res_security_log.c: SecurityEvent="FailedACL",EventTV="2022-01-08T17:08:59.957-0500",Severity="Error",Service="PJSIP",EventVersion="1",AccountID="anonymous",SessionID="2025076022",LocalAddress="IPV4/UDP/166.84.7.53/5060",RemoteAddress="IPV4/UDP/45.134.144.118/5823",ACLName="registrar_attempt_without_configured_aors" I would like pjsit not to listen,till I figure out how to configure the thing, so my logs don't fill up with messages. Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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] 18.7.1 - can't load res_fax, can't stop app_fax
I always empty /usr/lib/asterisk/modules if I am going to do an install with a different version, or better to do it always. On Wed, 03 Nov 2021 09:14:37 -0400, Kingsley Tart wrote: > > > Is the app_fax.so module still in /usr/lib/asterisk/modules? If so - > > if you remove it do things work. > > Is app_fax.so explicitly being loaded in modules.conf? > > Thanks. > > I was already waiting for it to finish recompiling after Doug's > suggestion but yes, app_fax.so was still in there and removing it then > let me remove the noload => res_fax.so line from modules.conf and > everything started fine. > > At the end of the re-compile it was nice to see it point this out > actually: > > --8<-- > WARNING WARNING WARNING > > Your Asterisk modules directory, located at > /usr/lib/asterisk/modules > contains modules that were not installed by this > version of Asterisk. Please ensure that these > modules are compatible with this version before > attempting to run Asterisk. > > app_fax.so > > WARNING WARNING WARNING > --8<-- > > > No, modules.conf didn't mention app_fax. > > Thanks. All sorted. Now to work on the next one ;) > > -- > Cheers, > Kingsley. > > > -- > _ > -- 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 > -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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] Stir Shaken
On Mon, 13 Jul 2020 15:44:12 -0400, Matthew Fredrickson wrote: > > On Mon, Jul 13, 2020 at 2:34 PM Saint Michael wrote: > >> > >> There is a big confusion here about Stir Shaken. It is NOT a provider > >> issue. Un fact, all providers are whasing their hands and modifying their > >> swihtches to pass-through the Signature. They cannot sign the call because > >> then the become the responsible party for the call before the FCC, and > >> liable for any illegal call. Every owner of a PBX that sends calls to the > >> network, except if you use a trunk for the likes of Vonage, needs to sign > >> their calls. So if you send calls with any kind of dialer and use DIDs, > >> real or "borrowed", you need to get the signature service urgently or your > >> business will stop terminating calls. You cannot self-sign, you cannot get > >> around it, the calls will either go to straight to voicemail or fail. Even > >> worse, the carries wil play a fake voicemail and charge you a fee, > >> something that some already a are doing when they detect robocallig. > > > > Don't even think about Transnexus, because they use 302 Redirect with a > > header, and no version of Asterisk supports it. I am the only game in the > > world for Stir-Shaken and Asterisk. I know it sounds arrogant but it is > > literally true. If you need to sign your calls to get through, with > > Asterisk, you need to connect to my service. I am an approved Service > > Provider from the FCC. If you keep thinking this is not happening, it is, > > and your business will disappear overnight. > > The issue is that Vicidial, for example, does not provide res_odbc and > > func_odbc, so you need to solve that first with Vicidial. Then you can > > apply the code I provided earlier and your calls with have a legal, binding > > signature. The carriers verify each signature and discard the ones that > > fail the cryptography test. > > Sounds like you're trying to sell/direct people towards a service that > you've created. Feel free to do so on the -biz list but the -users > list isn't the right place for that sort of thing. But the question is, are his statements correct that we need some service -- not necessarily his -- to sign the call before sending it to our normal carrier, or will the normal carrier -- whoever -- sign the call if they know the number? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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] Length of dial string
Or you could just increase MAX_EXTENSION and recompile. On Fri, 01 May 2020 06:25:36 -0400, Paddy Grice wrote: > > [1 ] > [1.1 ] > Hi Dovid > > Yes was one of the options but as the required list is dynamic becomes very > messy - and all combinations problem - where as "call all workers job xxx" > is what is needed so the ability to call 20+ numbers is what is needed - agi > does a database search for all jobx workers and constructs a dialstring with > SIP, DAHDI and Local devices. > > Can someone tell me where to find maximum string length for the dial data in > the DIAL command > > Paddy > > _ > > From: Dovid Bender [mailto:do...@telecurve.com] > Sent: 01 May 2020 10:26 > To: pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial > Discussion > Subject: Re: [asterisk-users] Length of dial string > > > Paddy, > > Why not use local extensions? You can do something like this. > Exten => > s,1,Dial(Local/set1@call_all/set2@call_all/set3@call_all) > > [call_all] > Exten => set1,1,Dial(SIP/100/101/102/103/104/105 > Exten => set1,1,Dial(SIP/106/107/108/109/110/111 > Exten => set1,1,Dial(SIP/112/113/114/1015/116/117 > > > On Fri, May 1, 2020 at 3:22 AM Paddy Grice wrote: > > > Hi all > > as per the new release notice for 13.33.0 received today - can anyone advise > me the max limit of the string to the Dial Command - see > * [ASTERISK-27946 > https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - > dial (API): Storage of dialed target uses AST_MAX_EXTENSION > when it shouldn't > > I have been fighting with this issue for months trying to find a solution I > need to call 20+ devices at the same time so dial strings are very long I > cant really use a queue(ringall) which was my original idea as the customer > needs different groups for virtually every call some of which are simple sip > devices and others have to be local devices (Internal and External CLIs). > > Paddy Grice > > > > > > -- > _ > -- 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 > > [1.2 ] > [2 ] > -- > _ > -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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] audio problem with asterisk and meetme conference
On Thu, 26 Mar 2020 09:18:24 -0400, Doug Lytle wrote: > > >>> Can I adjust the talk or listen volume for another user? > > I've never used the volume controls, but it would appear. > > https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration > > Doug According to this document, there is no way for me to change the volume(s) for another user, whereas meetme allows me to do this by specifying the conference number and user number. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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] audio problem with asterisk and meetme conference
On Thu, 26 Mar 2020 06:54:37 -0400, Doug Lytle wrote: > > >>> I never moved to confbridge because they don't have an option for > >>> controlling the volume of other > >>> participants audio > > I have menu options in my confbridge configs that has increase and decrease > conference volume. > > I'd still configure a small confbridge and test if you still have the issue, > since meetme is no longer being developed. Can I adjust the talk or listen volume for another user? If I could do that I would switch, but otherwise I have to stay with meetme. And I wonder if its a meetme issue? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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] audio problem with asterisk and meetme conference
On Wed, 25 Mar 2020 12:42:00 -0400, Doug Lytle wrote: > > > >>> he problem is that there is some sort of distortion in the audio > > Has been been going on for a while or is this a new setup? Do you have a > timing source? > > I bit the bullet around a year ago and moved to CONFBRIDGE; it wasn't as > horrible as I thought it would be to setup. Well, this has been going on for quite a while, my timing source is internal according to asterisk.conf. I never moved to confbridge because they don't have an option for controlling the volume of other participants audio, meetme has this feature which I use frequently. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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] audio problem with asterisk and meetme conference
Hi. I have a problem with my audio in meetme conference under asterisk 13 using Debian buster compiled from source. The problem is that there is some sort of distortion in the audio -- a workaround is always to lower the listen volume (*4). I see nothing in the log and so I wonder what is happening. I have dahdi loaded so I can record the conferences. Thanks in advance for any suggestions and let me know if you need any more information. I know 13 is old, I am working on upgrading. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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] setting up ODBC for cdr logging into MariaDB
I think you are missing a package, you need the odbc driver from mariadb, downloaded from their git repository -- if you build this using the default installation on a Debian type system, you would get /usr/local/lib64/libmaodbc.so as the driver file. On Fri, 11 Oct 2019 22:12:08 -0400, Fourhundred Thecat wrote: > > Hello, > > I am trying to set up cdr logging into MariaDB through ODBC. > > I have installed unixodbc unixodbc-dev and now I am struggling with > configuring /etc/odbcinst.ini > > All the examples online use non-existent libraries, ie: > > [MySQL] > Description = MySQL ODBC MyODBC Driver > Driver = /usr/lib/x86_64-linux-gnu/odbc/libmaodbc.so > Setup = /usr/lib/x86_64-linux-gnu/odbc/libodbcmyS.so > FileUsage = 1 > > I have these odbc related libraries on my system. Which of those do I > have to use for `Driver =` ? > > /usr/lib/x86_64-linux-gnu/libodbc.so > /usr/lib/x86_64-linux-gnu/libodbccr.so > /usr/lib/x86_64-linux-gnu/libodbcinst.so > > /usr/lib/x86_64-linux-gnu/odbc/libesoobS.so > /usr/lib/x86_64-linux-gnu/odbc/libmimerS.so > /usr/lib/x86_64-linux-gnu/odbc/libnn.so > /usr/lib/x86_64-linux-gnu/odbc/libodbcdrvcfg1S.so > /usr/lib/x86_64-linux-gnu/odbc/libodbcdrvcfg2S.so > /usr/lib/x86_64-linux-gnu/odbc/libodbcminiS.so > /usr/lib/x86_64-linux-gnu/odbc/libodbcmyS.so > /usr/lib/x86_64-linux-gnu/odbc/libodbcnnS.so > /usr/lib/x86_64-linux-gnu/odbc/libodbcpsqlS.so > /usr/lib/x86_64-linux-gnu/odbc/libodbctxtS.so > /usr/lib/x86_64-linux-gnu/odbc/liboplodbcS.so > /usr/lib/x86_64-linux-gnu/odbc/liboraodbcS.so > /usr/lib/x86_64-linux-gnu/odbc/libsapdbS.so > /usr/lib/x86_64-linux-gnu/odbc/libtdsS.so > > I have tries many possible permutations, but none worked. > > 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 > -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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] problem with new install with asterisk 15.7.4
hmmm, is asterisk 16 long term support? I thought only the od numbered releases were long term support. On Mon, 07 Oct 2019 08:02:51 -0400, George Joseph wrote: > > [1 ] > [2 ] > Oh, I forgot to mention that Asterisk 15 went End-Of-Life last Thursday. :) > You should use Asterisk 16. > > On Mon, Oct 7, 2019 at 5:58 AM George Joseph wrote: > > On Fri, Oct 4, 2019 at 1:19 PM John Covici wrote: > > Hi. I am trying to install asterisk 15.7.4 from git onto a Debian 10 > system and I am running into the following problem. I need to install > meetme (I know its old), and I have dahdi installed and the configure > script answers yes to all the edahdi questions, but the app_meetme > says depends on dahdi (e). I did not install libpri as I have no > hardware of that type. > > The (E) means "external" not "error". Does the app_meetme entry in > menuselect have "[ ]" before it or "XXX"? > If "[ ]" you should be able to select it and build. > > > I installed dahdi from git and have the kernel sources and it > installed without errors. > > How can I fix? > > Thanks in advance for any suggestions. > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici wb2una > cov...@ccs.covici.com > > -- > _ > -- 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 > > -- > George Joseph > Digium - A Sangoma Company | Software Developer | Software Engineering > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > direct/fax: +1 256 428 6012 > Check us out at: https://digium.com · https://sangoma.com > > * > > -- > George Joseph > Digium - A Sangoma Company | Software Developer | Software Engineering > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > direct/fax: +1 256 428 6012 > Check us out at: https://digium.com · https://sangoma.com > > * -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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] problem with new install with asterisk 15.7.4
Hi. I am trying to install asterisk 15.7.4 from git onto a Debian 10 system and I am running into the following problem. I need to install meetme (I know its old), and I have dahdi installed and the configure script answers yes to all the edahdi questions, but the app_meetme says depends on dahdi (e). I did not install libpri as I have no hardware of that type. I installed dahdi from git and have the kernel sources and it installed without errors. How can I fix? Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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] trying to upgrade asterisk and Debian -- not working (John Covici)
But I delete all the modules before the make install. I got no such warning. On Thu, 24 Jan 2019 09:46:13 -0500, Floimair Florian wrote: > > You need to run > make uninstall_all > while you still have 13.24.0-rc1 checked out. > Then checkout the previous version, rebuild it and make install. > 13.15.0 doesn't know anything about modules added by 13.24.0. > You usually would get a warning when running make install that there are > modules present that were not compiled with the current version. > > > > With best regards > > Florian Floimair > Innovation - Software-Development > > COMMEND INTERNATIONAL GMBH > A-5020 Salzburg, Saalachstraße 51 > http://www.commend.com <http://www.commend.com/> > > Security and Communication by Commend > > FN 178618z | LG Salzburg > > Am 24.01.19, 08:52 schrieb "asterisk-users im Auftrag von John Covici" > cov...@ccs.covici.com>: > > I checked out 13.15.0, ./configure, make delete all modules, followed > by make install. > > On Thu, 24 Jan 2019 01:17:32 -0500, > Stefan Viljoen wrote: > > > > What procedure did you follow to revert back to the old version? > > > > It sounds like your binary has been revereted, but the modules it needs > to load are still the 13.24.0-rc1 modules... > > > > --- > > Hi. I am trying to upgrade my asterisk from 13.15 to the latest of > asterisk 13 which seems to be 13.24.0-rc1. At the same time I want to go > from Debian 8 to DEbian 9 to get a more recent operating system and > applications. > > > > I ran in to the following problems when trying to do this. > > > > When trying to use asterisk 13.24.0-rc1, I ran into some strange > problems with some of my custom scripts. > > > > It seems the following statement immediately disconnects the user exten > => s,n,Read(digit,,1,,1,20) ; read a digit > > > > In the log after that statement it says user disconnected. I have an > agi which speaks some text before the read and that agi does not even say > anything, although it does complete. > > > > Now, if I try to go back to 13.15.0, it does not work at all because it > keeps telling in my log that modules support is not available, so no modules > get loaded. > > > > Any ideas on thispuzzle would be appreciated. > > > > > > -- > > Your life is like a penny. You're going to lose it. The question is: > > How do > > you spend it? > > > > John Covici wb2una > > cov...@ccs.covici.com > > > > > > > > -- > > > > Subject: Digest Footer > > > > ___ > > --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 > > > > -- > > > > End of asterisk-users Digest, Vol 173, Issue 21 > > *** > > > > > > -- > > _ > > -- 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 > > > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici wb2una > cov...@ccs.covici.com > > -- > _____ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.o
Re: [asterisk-users] trying to upgrade asterisk and Debian -- not working (John Covici)
I checked out 13.15.0, ./configure, make delete all modules, followed by make install. On Thu, 24 Jan 2019 01:17:32 -0500, Stefan Viljoen wrote: > > What procedure did you follow to revert back to the old version? > > It sounds like your binary has been revereted, but the modules it needs to > load are still the 13.24.0-rc1 modules... > > --- > Hi. I am trying to upgrade my asterisk from 13.15 to the latest of asterisk > 13 which seems to be 13.24.0-rc1. At the same time I want to go from Debian > 8 to DEbian 9 to get a more recent operating system and applications. > > I ran in to the following problems when trying to do this. > > When trying to use asterisk 13.24.0-rc1, I ran into some strange problems > with some of my custom scripts. > > It seems the following statement immediately disconnects the user exten => > s,n,Read(digit,,1,,1,20) ; read a digit > > In the log after that statement it says user disconnected. I have an agi > which speaks some text before the read and that agi does not even say > anything, although it does complete. > > Now, if I try to go back to 13.15.0, it does not work at all because it keeps > telling in my log that modules support is not available, so no modules get > loaded. > > Any ideas on thispuzzle would be appreciated. > > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici wb2una > cov...@ccs.covici.com > > > > -- > > Subject: Digest Footer > > ___ > --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 > > -- > > End of asterisk-users Digest, Vol 173, Issue 21 > *** > > > -- > _ > -- 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 > -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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] trying to upgrade asterisk and Debian -- not working
Hi. I am trying to upgrade my asterisk from 13.15 to the latest of asterisk 13 which seems to be 13.24.0-rc1. At the same time I want to go from Debian 8 to DEbian 9 to get a more recent operating system and applications. I ran in to the following problems when trying to do this. When trying to use asterisk 13.24.0-rc1, I ran into some strange problems with some of my custom scripts. It seems the following statement immediately disconnects the user exten => s,n,Read(digit,,1,,1,20) ; read a digit In the log after that statement it says user disconnected. I have an agi which speaks some text before the read and that agi does not even say anything, although it does complete. Now, if I try to go back to 13.15.0, it does not work at all because it keeps telling in my log that modules support is not available, so no modules get loaded. Any ideas on thispuzzle would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- 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] getting invites to rtp ports ??
Hi. So, I applied the patch, works, but I could not figure out a fail2ban regex which will hit that line, have you got one I can use? Thanks. On Thu, 30 Aug 2018 11:03:08 -0400, sean darcy wrote: > > On 08/29/2018 09:33 PM, John Covici wrote: > > OK, Thanks. I have a couple of questions -- the line numbers do not > > match exactly, so can you tell me a couple of lines before and after > > the line in question? Also, when will this be logged, if its only > > during sip debug, I need to change it to log when I can see it more > > readily. > > > > Thanks. > > > > On Wed, 29 Aug 2018 20:31:15 -0400, > > sean darcy wrote: > >> > >> On 08/29/2018 08:07 PM, John Covici wrote: > >>> I wonder if I could have that patch, maybe I could add it to my > >>> fail2ban regexp and if you have the correct regexp, I would apperciate > >>> that as well. > >>> > >>> Thanks. > >>> > >>> On Wed, 29 Aug 2018 19:18:29 -0400, > >>> Telium Support Group wrote: > >>>> > >>>> Depending on log trolling (Asterisk security log) misses a lot, and also > >>>> depends on the SIP/PJSIP folks to not change message structure (which > >>>> has already happened numerous time). If you are comfortable hacking > >>>> chan_sip.c you may prefer to get the same messages from the AMI. It > >>>> still misses a lot but that approach is better than nothing. > >>>> > >>>> Digium warns not to use fail2ban / log trolling as a security system: > >>>> http://forums.asterisk.org/viewtopic.php?p=159984 > >>>> > >>>> > >>>> -Original Message- > >>>> From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On > >>>> Behalf Of sean darcy > >>>> Sent: Wednesday, August 29, 2018 6:33 PM > >>>> To: asterisk-users@lists.digium.com > >>>> Subject: Re: [asterisk-users] getting invites to rtp ports ?? > >>>> > >>>> On 08/29/2018 11:59 AM, Telium Support Group wrote: > >>>>> Block a single IP is the wrong approach (whack-a-mole). You should > >>>>> consider a more comprehensive approach to securing your VoIP > >>>>> environment. Have a look at this wiki: > >>>>> > >>>>> https://www.voip-info.org/asterisk-security/ > >>>>> > >>>>> > >>>>> > >>>>> -Original Message- > >>>>> From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] > >>>>> On Behalf Of sean darcy > >>>>> Sent: Wednesday, August 29, 2018 10:46 AM > >>>>> To: asterisk-users@lists.digium.com > >>>>> Subject: Re: [asterisk-users] getting invites to rtp ports ?? > >>>>> > >>>>> On 08/29/2018 09:42 AM, Carlos Rojas wrote: > >>>>>> Hi > >>>>>> > >>>>>> Probably somebody is trying to hack your system, you should block > >>>>>> that ip on your firewall. > >>>>>> > >>>>>> Regards > >>>>>> > >>>>>> On Wed, Aug 29, 2018 at 9:34 AM, sean darcy >>>>>> <mailto:seandar...@gmail.com>> wrote: > >>>>>> > >>>>>>I'm getting invites to very high ports every 30 seconds from a > >>>>>>particular ip address: > >>>>>> > >>>>>>Retransmitting #10 (NAT) to 5.199.133.128:52734 > >>>>>><http://5.199.133.128:52734>: > >>>>>>SIP/2.0 401 Unauthorized > >>>>>>Via: SIP/2.0/UDP > >>>>>> > >>>>>> 0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734 > >>>>>>From: >>>>>><mailto:sip%3A37120116780191250@67.80.191.250>>;tag=1872048972 > >>>>>>To: >>>>>><mailto:sip%3A3712011972592181418@67.80.191.250>>;tag=as3a52e748 > >>>>>>Call-ID: 1504207870-295758084-609228182 > >>>>>>CSeq: 1 INVITE > >>>>>>... > >>>>>>WARNING[150318]: chan_sip.c:4127 retrans_pkt: Timeout on > >>>>>>15042078
Re: [asterisk-users] Community forum ?
Is Sangoma taking over Digium? Pretty soon there won't be anything open source around in this field at all. On Thu, 30 Aug 2018 11:14:33 -0400, Carlos Rojas wrote: > > [1 ] > [1.1 ] > [1.2 ] > Is the list going to be the same after sangoma take over digium? > > On Thu, Aug 30, 2018 at 11:12 AM, Joshua Colp wrote: > > On Thu, Aug 30, 2018, at 12:05 PM, sean darcy wrote: > > I see a lot of tag lines on posts for the Asterisk Community Forum. Is > > that forum supposed to supersede this mailing list ? > > Both remain available but the community forum seems to be more active, and > it is easier to search and find things. > > -- > 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 -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > 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 > [2 ] > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference 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] getting invites to rtp ports ??
The message currently in the log is not a security message and does not contain the ip address, would it be useful to block ip address from that message or is the challenge message sufficient? On Thu, 30 Aug 2018 09:37:58 -0400, Matthew Jordan wrote: > > [1 ] > [1.1 ] > [1.2 ] > On Thu, Aug 30, 2018 at 6:02 AM John Covici wrote: > > I agree, but is it possible to try over and over with anything other > than the challenge warning in the security log as sean suggested and > put a patch for? > > I don't think I understand your question. > > You shouldn't need a patch if you are using the SECURITY log. The thread > above is suggesting patching the source code to hijack a WARNING message for > the purposes of tracing security information; my point is that you should > have a > specific SECURITY log message that already serves that purpose. > > > > On Wed, 29 Aug 2018 22:52:05 -0400, > Matthew Jordan wrote: > > > > [1 ] > > [1.1 ] > > [1.2 ] > > On Wed, Aug 29, 2018 at 6:20 PM Telium Support Group > wrote: > > > > Depending on log trolling (Asterisk security log) misses a lot, and also > depends on the SIP/PJSIP folks to not change message structure (which has > already happened numerous time). If you are comfortable hacking chan_sip.c > you > may > > prefer to get the same messages from the AMI. It still misses a lot but > that approach is better than nothing. > > > > Digium warns not to use fail2ban / log trolling as a security system: > http://forums.asterisk.org/viewtopic.php?p=159984 > > > > That's some pretty old advice. > > > > The rationale for *not* using general log messages with fail2ban still > stands: the general WARNING/NOTICE/etc. log messages are subject to change > between versions, and no one wants that to impact someone's security. So you > should > not use > > those messages as input into fail2ban. > > > > That rationale did lead to the 'security' event type in log messages. > Security Event Logging - as it is called - got added into Asterisk quite some > time ago. So long ago I'm really not sure which version. At a minimum, > Asterisk 11, > but > > I'm pretty sure it was in 10 as well. > > > > Documentation for it can be found here: > > > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger > > > > And here: > > > > https://wiki.asterisk.org/wiki/display/AST/Logging+Configuration > > > > Note that this also fires off AMI events (and ARI events, IIRC). > > > > If, for whatever reason, you do not get a SECURITY log message or a > corresponding event when something 'bad' happens, that would be worth some > additional discussion. If anything, the events can be a bit chatty... > > > > > > -Original Message- > > From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On > Behalf Of sean darcy > > Sent: Wednesday, August 29, 2018 6:33 PM > > To: asterisk-users@lists.digium.com > > Subject: Re: [asterisk-users] getting invites to rtp ports ?? > > > > On 08/29/2018 11:59 AM, Telium Support Group wrote: > > > Block a single IP is the wrong approach (whack-a-mole). You should > consider a more comprehensive approach to securing your VoIP environment. > Have a look at this wiki: > > > > > > https://www.voip-info.org/asterisk-security/ > > > > > > > > > > > > -Original Message- > > > From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] > > > On Behalf Of sean darcy > > > Sent: Wednesday, August 29, 2018 10:46 AM > > > To: asterisk-users@lists.digium.com > > > Subject: Re: [asterisk-users] getting invites to rtp ports ?? > > > > > > On 08/29/2018 09:42 AM, Carlos Rojas wrote: > > >> Hi > > >> > > >> Probably somebody is trying to hack your system, you should block > > >> that ip on your firewall. > > >> > > >> Regards > > >> > > >> On Wed, Aug 29, 2018 at 9:34 AM, sean darcy > >> <mailto:seandar...@gmail.com>> wrote: > > >> > > >> I'm getting invites to very high ports every 30 seconds from a > > >> particular ip address: > > >> > > >> Retransmitting #10 (NAT) to 5.199.133.128:52734 > > >> <http://5.199.133.128:52734>: > > >> SIP/2.0 401 Unauthorized
Re: [asterisk-users] getting invites to rtp ports ??
sean darcy >> > >>>> <mailto:seandar...@gmail.com>> wrote: > >>>> > >>>> I'm getting invites to very high ports every 30 seconds from > >> a > >>>> particular ip address: > >>>> > >>>> Retransmitting #10 (NAT) to 5.199.133.128:52734 [1] > >>>> <http://5.199.133.128:52734>: > >>>> SIP/2.0 401 Unauthorized > >>>> Via: SIP/2.0/UDP > >>>> > >> > > 0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734 > >>>> From: >>>> > >> <mailto:sip%3A37120116780191250@67.80.191.250>>;tag=1872048972 > >>>> To: >>>> > >> <mailto:sip%3A3712011972592181418@67.80.191.250>>;tag=as3a52e748 > >>>> Call-ID: 1504207870-295758084-609228182 > >>>> CSeq: 1 INVITE > >>>> ... > >>>> WARNING[150318]: chan_sip.c:4127 retrans_pkt: Timeout on > >>>> 1504207870-295758084-609228182... > >>>> > >>>> I thought invites had to go to port 5060 or so. I don't > >> understand > >>>> why somebody (let's assume a bad guy) is trying ports above > >> 5. > >>>> > >>>> sean > >>>> > >>>> > >>> > >>> Ok, so the high port is not the destination port but the source > >> port. > >>> > >>> So I hacked the log warning in chan_sip.c on non-critical invites > >> to show the source ip: > >>> > >>> ast_log(LOG_WARNING, "Timeout on %s non-critic invite trans from > >>> %s.\n", > >>> > >> > > pkt->owner->callid,ast_sockaddr_stringify(sip_real_dst(pkt->owner))); > >>> > >>> With that in the log, I'm now blocking the ip addresses. > >>> > >>> Thanks, > >>> sean > >>> > >>> > >>> -- > >>> > >> > > _ > >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com > >> -- > >>> > >>> Astricon is coming up October 9-11! Signup is available at: > >>> https://www.asterisk.org/community/astricon-user-conference > >>> > >>> Check out the new Asterisk community forum at: > >>> https://community.asterisk.org/ > >>> > >> > >> I agree. That's why I hacked chan_sip.c to get the addresses in the > >> log. > >> > >> I'm surprised they're not in the log by default. I must be the only > >> person who gets these "non-critical invites". > >> > >> sean > >> > >> -- > >> > > _ > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com > >> -- > >> > >> Astricon is coming up October 9-11! Signup is available at: > >> https://www.asterisk.org/community/astricon-user-conference > >> > >> 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 > >> -- > >> > >> Astricon is coming up October 9-11! Signup is available at: > >> https://www.asterisk.org/community/astricon-user-conference > >> > >> 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 > > > > -- > > Matthew Jordan > > Digium, Inc. | CTO > > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > > Check us out at: http://digium.com & http://asterisk.org > > > > Links: > > -- > > [1] http://5.199.133.128:52734 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > 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 > [2 ] > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference 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] getting invites to rtp ports ??
t; pkt->owner->callid,ast_sockaddr_stringify(sip_real_dst(pkt->owner))); > > > > With that in the log, I'm now blocking the ip addresses. > > > > Thanks, > > sean > > > > > > -- > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > Astricon is coming up October 9-11! Signup is available at: > > https://www.asterisk.org/community/astricon-user-conference > > > > Check out the new Asterisk community forum at: > > https://community.asterisk.org/ > > > > I agree. That's why I hacked chan_sip.c to get the addresses in the log. > > I'm surprised they're not in the log by default. I must be the only person > who gets these "non-critical invites". > > sean > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > 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 -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > 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 > > -- > Matthew Jordan > Digium, Inc. | CTO > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com & http://asterisk.org > [2 ] > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference 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] getting invites to rtp ports ??
OK, Thanks. I have a couple of questions -- the line numbers do not match exactly, so can you tell me a couple of lines before and after the line in question? Also, when will this be logged, if its only during sip debug, I need to change it to log when I can see it more readily. Thanks. On Wed, 29 Aug 2018 20:31:15 -0400, sean darcy wrote: > > On 08/29/2018 08:07 PM, John Covici wrote: > > I wonder if I could have that patch, maybe I could add it to my > > fail2ban regexp and if you have the correct regexp, I would apperciate > > that as well. > > > > Thanks. > > > > On Wed, 29 Aug 2018 19:18:29 -0400, > > Telium Support Group wrote: > >> > >> Depending on log trolling (Asterisk security log) misses a lot, and also > >> depends on the SIP/PJSIP folks to not change message structure (which has > >> already happened numerous time). If you are comfortable hacking > >> chan_sip.c you may prefer to get the same messages from the AMI. It still > >> misses a lot but that approach is better than nothing. > >> > >> Digium warns not to use fail2ban / log trolling as a security system: > >> http://forums.asterisk.org/viewtopic.php?p=159984 > >> > >> > >> -Original Message- > >> From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On > >> Behalf Of sean darcy > >> Sent: Wednesday, August 29, 2018 6:33 PM > >> To: asterisk-users@lists.digium.com > >> Subject: Re: [asterisk-users] getting invites to rtp ports ?? > >> > >> On 08/29/2018 11:59 AM, Telium Support Group wrote: > >>> Block a single IP is the wrong approach (whack-a-mole). You should > >>> consider a more comprehensive approach to securing your VoIP environment. > >>> Have a look at this wiki: > >>> > >>> https://www.voip-info.org/asterisk-security/ > >>> > >>> > >>> > >>> -Original Message- > >>> From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] > >>> On Behalf Of sean darcy > >>> Sent: Wednesday, August 29, 2018 10:46 AM > >>> To: asterisk-users@lists.digium.com > >>> Subject: Re: [asterisk-users] getting invites to rtp ports ?? > >>> > >>> On 08/29/2018 09:42 AM, Carlos Rojas wrote: > >>>> Hi > >>>> > >>>> Probably somebody is trying to hack your system, you should block > >>>> that ip on your firewall. > >>>> > >>>> Regards > >>>> > >>>> On Wed, Aug 29, 2018 at 9:34 AM, sean darcy >>>> <mailto:seandar...@gmail.com>> wrote: > >>>> > >>>> I'm getting invites to very high ports every 30 seconds from a > >>>> particular ip address: > >>>> > >>>> Retransmitting #10 (NAT) to 5.199.133.128:52734 > >>>> <http://5.199.133.128:52734>: > >>>> SIP/2.0 401 Unauthorized > >>>> Via: SIP/2.0/UDP > >>>> > >>>> 0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734 > >>>> From: >>>> <mailto:sip%3A37120116780191250@67.80.191.250>>;tag=1872048972 > >>>> To: >>>> <mailto:sip%3A3712011972592181418@67.80.191.250>>;tag=as3a52e748 > >>>> Call-ID: 1504207870-295758084-609228182 > >>>> CSeq: 1 INVITE > >>>> ... > >>>> WARNING[150318]: chan_sip.c:4127 retrans_pkt: Timeout on > >>>> 1504207870-295758084-609228182... > >>>> > >>>> I thought invites had to go to port 5060 or so. I don't understand > >>>> why somebody (let's assume a bad guy) is trying ports above 5. > >>>> > >>>> sean > >>>> > >>>> > >>> > >>> Ok, so the high port is not the destination port but the source port. > >>> > >>> So I hacked the log warning in chan_sip.c on non-critical invites to show > >>> the source ip: > >>> > >>> ast_log(LOG_WARNING, "Timeout on %s non-critic invite trans from > >>> %s.\n", > >>> pkt->owner->callid,ast_sockaddr_stringify(sip_real_dst(pkt->owner))); > >>> > >>> With that in the log, I'm now blocking the ip addresses. > >>> > >>&
Re: [asterisk-users] getting invites to rtp ports ??
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 -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > 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 > -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference 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 UC 4.x Unreachable
I always set it to no, but set the registration time to 60 seconds, and that has always worked for me. On Wed, 23 Aug 2017 17:23:38 -0400, Gary Reuter wrote: > > Hello, > We've had dozens of Polycom 3.x firmware phones deployed and working > great for years. > Now I've finally been charged with the long-overdue task of figuring > out why newer Polycom devices with 4.x firmware register fine but do > not respond to SIP OPTIONS request and therefore always become > UNREACHABLE if the sip qualify setting is set to yes. > > To my dismay, searches for solutions from others who have encountered > this problem have given zero results. > > > 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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] Any way of creating a file to write to from the dialplan, or must I use AGI?
Won't the system command do it? On Fri, 04 Nov 2016 17:26:13 -0400, Jonathan H wrote: > > Seems I can write to an existing file, but is there really no way of > creating a new file to log some data to, without reverting to AGI? > (will be different for each caller ID) > > -- > _ > -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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] Just got defrauded - how do I block calls which contain a dash (RegEx noob question)
Also, make sure you are using fail2ban and that you have good passwords on your extensions. On Fri, 28 Oct 2016 11:55:42 -0400, John Covici wrote: > > How about a \ before the - ? > > On Fri, 28 Oct 2016 11:38:13 -0400, > Markus wrote: > > > > Hi list, > > > > I'm using Asterisk2Billing (v2.0.16) and it appears to have an > > annoying bug. When there are rates for e.g. 44 (UK landline) and > > 44870 (UK premium) and a fraudster manages to somehow dial 44-870 > > instead of 44870 the rate for 44 will match, not the one for > > 44870. > > > > So, I would like to block all calls on a dialplan level that > > contain a dash. -44, 4-4, 44-, 44---, -, ---, just everything > > with a friggin' dash. > > > > My noob-ish try: > > > > exten => _-.,1,NoOp(Blocking dash) > > exten => _-.,n,Hangup > > > > Doesn't work. > > > > On https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching I found: > > > > "The dash (-) character is ignored in extensions and patterns > > except when it is used in a pattern to specify a range in a > > character set. It has no effect in matching or sorting > > extensions." > > > > How do I do it right? > > > > Thank you! > > Markus > > > > -- > > _ > > -- 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 > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > cov...@ccs.covici.com > > -- > _ > -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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] Just got defrauded - how do I block calls which contain a dash (RegEx noob question)
How about a \ before the - ? On Fri, 28 Oct 2016 11:38:13 -0400, Markus wrote: > > Hi list, > > I'm using Asterisk2Billing (v2.0.16) and it appears to have an > annoying bug. When there are rates for e.g. 44 (UK landline) and > 44870 (UK premium) and a fraudster manages to somehow dial 44-870 > instead of 44870 the rate for 44 will match, not the one for > 44870. > > So, I would like to block all calls on a dialplan level that > contain a dash. -44, 4-4, 44-, 44---, -, ---, just everything > with a friggin' dash. > > My noob-ish try: > > exten => _-.,1,NoOp(Blocking dash) > exten => _-.,n,Hangup > > Doesn't work. > > On https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching I found: > > "The dash (-) character is ignored in extensions and patterns > except when it is used in a pattern to specify a range in a > character set. It has no effect in matching or sorting > extensions." > > How do I do it right? > > Thank you! > Markus > > -- > _ > -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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] "Follow me" with Asterisk that detects cellphone voicemail and similar announcements
I know if you use freepbx on top of asterisk, you get a followme which calls one or more cell phones and ask for confirmation, maybe the regular asterisk followme does this as well, but basically this is the way to do it. Robin Kipp <mli...@robin-kipp.net> wrote: > Hi all, > > sorry if the subject is a bit confusing, but I just couldn’t think of a good > way of better describing the situation… > > Basically, I travel a lot and have several SIM cards for my phone from local > carriers. What I’d like to do now is to setup Asterisk, so that people who > want to reach me just have to dial one number which forwards the call to all > my cellphone numbers in turn. I’m still pretty new to Asterisk, so I’m unsure > which method would be most suitable for this scenario. > > Theoretically, I could use the dial function to call one number, then wait a > few seconds and then dial another number. In practice, this won’t work > because as soon as a call is answered by the mobile carrier’s voicemail the > caller would be connected to that, no other numbers would be called. > So here’s my question: how can I possibly avoid this situation? Is there a > way for Asterisk to detect such situations and distinguish them from me > actually trying to answer the call when the correct number is called? > Not sure if this is technically possible, but figured I’d ask just in case > there is any sort of solution. I’m aware that it would be best to simply use > SIP and a SIP client on my phone in order to take the call, but due to most > carriers blocking SIP traffic on their mobile data networks this wouldn’t > work as soon as I’m not connected to any WiFi. > So, in case there’s any solution to this problem I’d greatly appreciate if > you could share that with me! > Many thanks and best wishes, > Robin > -- > _ > -- 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 > -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] my dahdi dont'n start
Richard Mudgett <rmudg...@digium.com> wrote: > On Tue, Apr 26, 2016 at 11:07 AM, Administrator TOOTAI <ad...@tootai.net> > wrote: > > > Le 26/04/2016 17:23, Mamadou NGOM a écrit : > > > >> Hello, > >> > >> > >> Having installed DAHDI to be able to use the meetme() application , when > >> I start the dahdi service it generates me the following error: > >> > >> -bash: /etc/init.d/dahdi: No such file or directory > >> > > > > Clear, the file dahdi is not existing. Did you copy it? > > > > BTW, you shouldn't need dahdi to run meetme. BTW #2, depending on your > > asterisk version, meetme is replaced by ConfBridge > > > > Administrator TOOTAI: You must have DAHDI running when using meetme because > DAHDI does the audio mixing for the conference. > > Meetme is deprecated and replaced by ConfBridge on all currently supported > Asterisk > versions. > Except that confbridge lacks some features that meetme has -- I wish this were not so. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] small pbx for the office [it was: small homebrew pbx]
I have a small pbx and my sp3102 about one out of 10 times does not pick up the line -- I have reduced the voltage to 15 volts, but no joy, every so often it still does not answer. Any ideas on that? Ryan Wagoner rswago...@gmail.com wrote: On Wed, Jun 17, 2015 at 9:07 AM, lu...@sulweb.org wrote: Lukasz Sokol wrote: but have you considered a web-managed config-builder such as FreePBX? Instead of building your dialplan from scratch ? I've never used FreePBX, but, after having looked at its website, I think I have a general understanding of what it can do. What I don't understand is how FreePBX answers my question about the Linksys SPA3102 being good for a mission critical solution or not. I've used the SPA3102 and would recommend it for home use. For business look at the Patton SmartNode 4110 series devices or a Cisco router with FXO card and DSP modules. I have deployed both and haven't had any complaints. They just work once configured. Ryan Alternatives: -- _ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming calls fall into echo test mode
check your logs /var/log/asterisk/full -- make sure your verbosity is set high enough to do you good and you wll probably find the answer. Pat Collins drdialt...@optonline.net wrote: Perhaps assigned as a test number somewhere along the line? Are these ISDN, SIP, IAX calls? There are MANY smart people on this list. Maybe sharing the relevant configs and traces is a good place to start??? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Norman Molhant Sent: Saturday, July 19, 2014 10:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] incoming calls fall into echo test mode Hello all, Weird trouble here: we have 60-some happy subscribers on a FreePBX box, each with its own phone number, with no problem at all, except for one (and only one) subscriber who has this problem: his outgoing calls are ok, but when someone dials his phone number (be it from our network or from any other place in the world), the caller ears the standard message signalling he has entered the echo test mode and must dial # to exit that mode. Most callers don't understand what's going on, then give up and hang up without dialling #. Very few dial # one or more times, then those few get our customer's phone ringing and are then able to reach our customer. I went through all the docs, wikis and discussions I found on the web, without finding any data on how to solve that problem. I tried many things on our FreePBX box and found out the problem seems somehow linked with the customer's extension (or phone number), not his inbound route (changing the latter has no effect on the problem). Creating a new extension with another phone number would solve the problem (I tried it and it works), but this customer wants to keep his current phone number and when I tried deleting his extension then creating a new one with his current phone number, the new extension presented the same problem as the previous one... Anyone knows what could cause such a problem and/or how to solve it ? Thanks, Norman. ad...@csur.ca -- _ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.10.0 Now Available
Asterisk Development Team asteriskt...@digium.com wrote: The Asterisk Development Team has announced the release of Asterisk 11.10.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.10.0 resolves several issues reported by the community and would have not been possible without your participation. ... * ASTERISK-23754 - [patch] Use var/lib directory for log file configured in asterisk.conf (Reported by Igor Goncharovsky) Is this mandatory -- what is wrong with /var/log/asterisk for those files? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.10.0 Now Available
Michael L. Young myo...@acsacc.com wrote: - Original Message - From: cov...@ccs.covici.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 29, 2014 6:42:05 PM Subject: Re: [asterisk-users] Asterisk 11.10.0 Now Available * ASTERISK-23754 - [patch] Use var/lib directory for log file configured in asterisk.conf (Reported by Igor Goncharovsky) Is this mandatory -- what is wrong with /var/log/asterisk for those files? The title on that issue is very misleading. The patch that went in was just for chan_ooh323. The change was to have chan_ooh323 use the log directory configured in asterisk.conf instead of using a hard coded value. OK, thanks, boy that title is sure misleading! -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ControlPlayback can not replay complicated file names
The colon freaks it out, this may be some parsing problem, I bet. Jonathan White j...@uvacity.com wrote: If not sure if I am looking at a bug or expected behaviour as I do not see anything in the documentation. ControlPlayback can not replay complicated file names For example it can replay 1005 but it can not replay 1005-2014-04-08_23:58:17 Playback can replay 1005-2014-04-08_23:58:17 I suspect this relates to how the variables are parsed and parameters set. Does anyone have any further information to suggest what the limitations are around file naming? Thanks --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com Alternatives: -- _ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the possible cause of maximum pbx stack exceeded
OK, thanks. Rusty Newton rnew...@digium.com wrote: I'm not a developer, but from comments in the code, it looks like that warning is generated when Asterisk dialplan processing exceeds a certain depth of includes. Seeing as it is possibly a dialplan related issue, and FreePBX is writing your dialplan, you may have the best odds of getting a relevant answer by asking on the FreePBX forums (and giving them access to a copy of your logs to examine) That's all I got! :) On Wed, Dec 4, 2013 at 3:27 AM, cov...@ccs.covici.com wrote: Hi. I am using asterisk 11 svn r401076M and I am getting this warning at times. I can't find much doing a google search, so anyone with any ideas? I have looked at the logs, but can find no particular pattern to indicate where this is happening and the system appears to be otherwise working, but I am still wondering if something is wrong. I am also using freepbx in case there are known issues there -- because some of these occur during their dialout trunk code. Any suggestions would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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 -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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
[asterisk-users] what is the possible cause of maximum pbx stack exceeded
Hi. I am using asterisk 11 svn r401076M and I am getting this warning at times. I can't find much doing a google search, so anyone with any ideas? I have looked at the logs, but can find no particular pattern to indicate where this is happening and the system appears to be otherwise working, but I am still wondering if something is wrong. I am also using freepbx in case there are known issues there -- because some of these occur during their dialout trunk code. Any suggestions would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP server on windows
Use freeswitch instead, it does run on Windows. Ruddy Gbaguidi plugwo...@micnes.com wrote: This is about an call center application we are building and that need an embedded PBX. We would then like to have that platform run on Windows and Linux. Are there ways to easy ship linux application embedded in virtual machine so they can run on windows ? Le 2013-12-04 08:02, Dan Journo a écrit : FROM: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] ON BEHALF OF Ruddy Gbaguidi SENT: 04 December 2013 09:08 TO: 'Asterisk Users Mailing List - Non-Commercial Discussion' SUBJECT: [asterisk-users] Asterisk SIP server on windows Hi all, I need to build an application that will be an SIP server program that will run on Linux and Windows. The sip server need only some features such as be able to : - Register sip endpoints - Answer a call and play a local file - Make a dial from one channel to another. I know asterisk can be stripped to exactly fit my needs. I would like to know if there is a way to build it on windows after it has been stripped. Or do I have other alternatives out there ? Servers that can run Asterisk are so cheap nowadays, unless you are talking about huge volumes of traffic. I'd recommend getting a server and putting on Centos which is tried and tested. You'll waste less time that way and avoid any unforeseen problems. Or look for a cloud server to do the job for you. -- _ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?
I would thinktwice about Amazon -- and virtual in general is not a good idea for this sort of thing. I have seen messages about bad results with amazon specifically. Todd R. tjrl...@live.com wrote: Just checking one more time to see if anyone has an opinion on this. I am primarily interested in using a cloud type setup such as Amazon AWS for the redundancy, easy backup and recovery options. It's not about price but the idea that it will be very hard for a single piece of hardware to ruin my day. From: tjrl...@live.com To: asterisk-users@lists.digium.com Date: Mon, 18 Nov 2013 18:33:38 -0600 Subject: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system? Took me a while but I have finally embraced cloud computing and all the benefits. The only thing I have yet to feel comfortable about putting in the cloud is real live Asterisk boxes to be used in production. I know it's being done because as far as I know Twilio is using Amazon for their Asterisk boxes. I have read all the fun articles on building hobby type systems and that's all great. What I really need to hear is from those that have deployed Asterisk in Amazon or Digital Ocean and how many simultaneous calls they are pushing through it and what the call quality and reliability has been. Right now I am still using dedicated hardware but I could become much more redundant and scale much faster using Amazon or Digital Ocean. Thanks in advance for any information from those that have already been down this road... -- _ -- 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 Alternatives: -- _ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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
[asterisk-users] problem with dtmf detection in asterisk 11
Hi. I am having problems with asterisk detection dtmf properly in asterisk 11. I am up to rev 390229. Now, when coming in off a did we have with Velocity, the dids work fine, but from extensions often it misses digits -- I can type *4 and it will miss the 4. Often, if I type quite slowly things will work properly. All dtmf modes are set to rfc2833. Strangely enough, I did not notice this with asterisk 8, but I would hate to go back to solve this problem. Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to send dtmf after pause ?
Sean Darcy seandar...@gmail.com wrote: On 06/07/2013 01:48 PM, Asghar Mohammad wrote: hi, you can add more w (ww1234#) for more delay. On Fri, Jun 7, 2013 at 7:17 PM, Yves A. yves...@gmx.de mailto:yves...@gmx.de wrote: This would be possible with an agi... the agi can wait for silence or 10 seconds, as u like and then play the dtmf tones and bridge the call to your extension afterwards. yves Am 07.06.2013 17:51, schrieb Sean Darcy: I'm trying to call a conference service, wait 10 seconds, then send the passcode. I've tried ww: Dial(SIP/18005551212ww12345#@s__ip.com http://sip.com,60,r) The sip channel didn't like that. Added 'p' , still no help. I tried D: Dial(SIP/18005551...@sip.com mailto:18005551...@sip.com,__60,rD(12345#) The dtmf is sent too soon. I tried inserting 'ww' but that was just sent. I tried G: exten = 234.1.Dial(SIP/18005551212@__sip.com mailto:18005551...@sip.com,60,rG(next)) same=n(next),Wait(10) same=n,SendDTMF(12345#) but that didn't work at all, This is a common use case. There must be some simple answer I'm missing. Thanks for any help. sean Thanks for the reply, but any 'w' s in the dial string cause CHAN_UNAVAILABLE. I'm not sure I'm up for learning agi just yet. I was hoping for a dialplan solution. sean Those W's are only available in some dahdi drivers and they only wait at the very beginning, if I remember correctly. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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
[asterisk-users] asterisk-1.8 change in meetme behavior was this on purpose?
Hi. After the latest upgrade of asterisk-1.8, I notice that meetme does not allow the menu, if musiconhold is active which only occurrs if a single user is in the conference. Sometimes I have to unmute someone because for various reasons they cannot do this and they are the first one in the conference. I looked at the code and its a one line change, and I wonder if it was done deliberately or not -- if so I wonder if it were possible to fix this? Thanks in advance for any ideas. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk for small home phone system
Or you could use the followme feature to have asterisk just call your cell phones. Roger Burton West ro...@firedrake.org wrote: On Thu, Oct 25, 2012 at 11:09:01AM -0700, Matthew Hixson wrote: - Is the Linksys SPA3102 a good piece of hardware for this type of setup or is there something cheaper? Perhaps a card that can go right into the Linux box? I'm using an OpenVox A400 (with an FXO module), which Asterisk can drive directly. - Would we configure our SIP clients on our iphones to login directly to Asterisk running on my home Linux box? I have 18MB/2.5MB internet service with a static IP so this wouldn't be a problem. That would be the simplest approach (modulo firewalls). If you already have another SIP provider, you could configure your home asterisk to forward calls to that... R -- _ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failover router recommendation
I am sure Mikrotik routers will do this also, although I have not tried it. Niccolò Belli darkba...@linuxsystems.it wrote: Il 09.10.2012 21:24 Mike Diehl ha scritto: I hope no one considers this off topic... I have a phone customer who wants 2 Internet connections so that if one goes down, he can use the other for phone service. So, I'd like to get a recommendation for a relatively inexpensive router that can perform this function. Also, when the failover occurs, the phone's IP address will obviously change. So, how can/should I configure this to minimize my customer's down-time? http://www.traverse.com.au/geos21-dual-adsl2-x86-router-appliance I achieved fallback in less than 10 seconds flushing routing cache and nat tables with nearly zero false positives (I can do even better but I prefer having less false disconnections). I don't use this router but a Traverse Solos PCI Adsl2+ card and a linux box. Cheers, Niccolò -- _ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting unwanted pager email from Asterisk voicemail
Check voicemail.conf and you will probably find both an Email and pager address in there. Just get rid of the pager address. Danny Nicholas da...@debsinc.com wrote: My guess is that your email provider is forwarding the message since Asterisk should send the same content to both places. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Duncan Turnbull Sent: Thursday, May 31, 2012 6:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Getting unwanted pager email from Asterisk voicemail Hi All I am not sure why but I am getting a pager email as well as a voicemail email when a voicemail is left. I am guessing its a setting somewhere but I can't find it The system is Ubuntu with Asterisk 1.8.12 from source. I am using Freepbx for the configs but freepbx doesn't do much to voicemail The mail system is Postfix My test scenario [general] format=wav49|gsm|wav serveremail=aster...@questterrace.co.nz attach=yes skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 emaildateformat=%A, %B %d, %Y at %r pagerdateformat=%A, %B %d, %Y at %r sendvoicemail=yes ; Allow the user to compose and send a voicemail while inside [default] 121 = 1234,Duncan testing,dun...@e-simple.co.nz,,attach=yes|saycid=no|envelope=no|delete=no I get the voicemail with attachment Subject [PBX]: New message 1 in mailbox 121 Dear Duncan testing: Just wanted to let you know you were just left a 0:08 long message (number 1) in mailbox 121 from 21722545, on Thursday, May 31, 2012 at 08:45:02 PM so you might want to check it when you get a chance. Thanks! --Asterisk And also a pager email Subject:New VM New 0:08 long msg in box 121 from 21722545, on Thursday, May 31, 2012 at 08:45:02 PM Anyone seen something obvious I am missing? Thanks very much Alternatives: -- _ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.8 codec negotiation
Kevin P. Fleming kpflem...@digium.com wrote: On 01/01/2012 04:17 PM, cov...@ccs.covici.com wrote: Hi. I am using asterisk 1.8 and everything was working fine when I was at svn 342661. I then upgraded to vrsion 349339 and discovered the following problem -- one of the end points is a freeswitch box which offers a number of codecs, including PCMU. However, when I tried to make a call I got a 488 response and a message multiple audio streams not supported in the log. multiple audio streams != multiple audio codecs. For some reason Asterisk is receiving an INVITE with an offer for more than one audio stream (m=audio), and that is not supported. OK, but if I have a phone or in my case a server which offers a choice of codecs, why can't asterisk just pick the ones it has rather than reject the call? Is there a way to do this correctly as far as asterisk is concerned? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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
[asterisk-users] asterisk 1.8 codec negotiation
Hi. I am using asterisk 1.8 and everything was working fine when I was at svn 342661. I then upgraded to vrsion 349339 and discovered the following problem -- one of the end points is a freeswitch box which offers a number of codecs, including PCMU. However, when I tried to make a call I got a 488 response and a message multiple audio streams not supported in the log. Is this by design? I found an issue 18859, but that referenced where the end point offered both regular rtp and srtp. But it seems to me if an endpoint offers various codecs, that asterisk could only complain if none of them match one that asterisk likes. If I only offer one codec, it works, but that seems an unnecessary restriction to me. Any assistance on this would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi_dummy required?
Kevin P. Fleming kpflem...@digium.com wrote: On 09/23/2011 02:50 AM, Ishfaq Malik wrote: On Thu, 2011-09-22 at 22:19 -0600, Troy Telford wrote: I'm running Asterisk 1.8.4.4; I'm new to asterisk, and have been reading the Asterisk Definitive Guide, and it mentions that dahdi_dummy should be used to provide an interface for Asterisk to get kernel timing. - espescially if using timing-dependant modules. I have a minor question: is dahdi_dummy necessary or useful anymore - espescially for users who don't have DAHDI hardware? I ask because I just checked out dahdi 2.5 from svn built (against the Linux kernel 3.0) I noticed that dahdi_dummy didn't seem to be built; when I poked around in the changelog, I saw: * README: README: Remove references to dahdi_dummy. Since dahdi_dummy is no longer required remove the references from README. (issue #17959) Reported by: glen201 Origin: http://svnview.digium.com/svn/dahdi?view=revrev=9308 So am I correct in assuming dahdi_dummy isn't needed/useful anymore? Application MeetMe will not work without it. This is completely incorrect. MeetMe never relied on dahdi_dummy specifically, it requires DAHDI to have a working timing source. Yes, at one point dahdi_dummy was available to provide a timing source if there weren't any DAHDI cards in the system... but it is no longer necessary. DAHDI is now able to provide timing and audio mixing using kernel timers using a built-in timer, so there is no need for a separate module. The ChangeLog entry above is correct, as of DAHDI 2.4 and later. So, how do I get this to work -- when I tried to do this, I could get a conference all right, but it would not record the conference till I actually loaded dahdi-dummy -- which seems to be still built. I am using 9729 out of trunk. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi_dummy required?
Shaun Ruffell sruff...@digium.com wrote: On Fri, Sep 23, 2011 at 09:30:59AM -0400, cov...@ccs.covici.com wrote: Kevin P. Fleming kpflem...@digium.com wrote: On 09/23/2011 02:50 AM, Ishfaq Malik wrote: On Thu, 2011-09-22 at 22:19 -0600, Troy Telford wrote: So am I correct in assuming dahdi_dummy isn't needed/useful anymore? Application MeetMe will not work without it. This is completely incorrect. MeetMe never relied on dahdi_dummy specifically, it requires DAHDI to have a working timing source. Yes, at one point dahdi_dummy was available to provide a timing source if there weren't any DAHDI cards in the system... but it is no longer necessary. DAHDI is now able to provide timing and audio mixing using kernel timers using a built-in timer, so there is no need for a separate module. The ChangeLog entry above is correct, as of DAHDI 2.4 and later. So, how do I get this to work -- when I tried to do this, I could get a conference all right, but it would not record the conference till I actually loaded dahdi-dummy -- which seems to be still built. I am using 9729 out of trunk. John, As kpfleming said, dahdi_dummy is no longer built by default. Revision 9729 you referenced was first released in 2.5.0 which definitely does not use dahdi_dummy by default. Perhaps you believe you were able to load dahdi_dummy because dahdi is aliased to dahdi_dummy and before loading it you were using confbridge? Below you can see how only dahdi is needed for timing and conferencing since the timers are processed in the same function that handles the conferencing: You can modprobe dahdi_dummy but only 'dahdi' is loaded and dahdi_test will work fine... # modprobe dahdi_dummy # lsmod | grep dahdi dahdi 196680 0 crc_ccitt 6337 1 dahdi # dahdi_test -v -c 3 Opened pseudo dahdi interface, measuring accuracy... 8192 samples in 8191.592 system clock sample intervals (99.995%) 8192 samples in 8190.720 system clock sample intervals (99.984%) 8192 samples in 8191.288 system clock sample intervals (99.991%) --- Results after 3 passes --- Best: 99.995 -- Worst: 99.984 -- Average: 99.990234, Difference: 99.990233 But you can do the same thing only by loading dahdi and not dahdi_dummy... # modprobe -r dahdi # lsmod | grep dahdi # dahdi_test Unable to open dahdi interface: No such file or directory # modprobe dahdi # dahdi_test -v -c2 Opened pseudo dahdi interface, measuring accuracy... 8192 samples in 8199.624 system clock sample intervals (100.093%) 8192 samples in 8182.688 system clock sample intervals (99.886%) --- Results after 2 passes --- Best: 99.907 -- Worst: 99.886 -- Average: 99.896633, Difference: 99.989697 DAHDI will still use the timing from an installed card if available, but now it is smart enough to detect if there is not a card installed or operating properly and still provide timing without requiring the user to load dahdi_dummy explicitly. You are correct, when meetme didn't work, I did not even load dahdi at all -- that was the confusion. I am surprised that the modprobe of dahdi-dummy even succeeds, but I guess it does not matter. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RESEND: Mixmonitor command parameter problem on Asterisk 1.8.4
Execute a shell script instead -- too bad they have a small limit, but that should work. Ikka - Mitra Kreasindo ikka.vert...@mitrakreasindo.com wrote: Is anyone can help me with this ? I'm really desperate. Thx in ad. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ikka - Mitra Kreasindo Sent: Wednesday, September 14, 2011 5:02 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Mixmonitor command parameter problem on Asterisk 1.8.4 Dear all. I'm using MixMonitor command in my dialplan, and I used the command parameter to execute some thing after recording the file. I used the command parameter to convert the wav file that created earlier to MP3 and than deleted the WAV file. It worked fine with asterisk 1.4.21.2. and 1.6x But than I have a new asterisk server with asterisk 1.8.4. The command parameter doesn't work. It's trimed for about 297 character only. The rest was gone. This is part of the log with Asterisk 1.4.21.2 -- Executing [08129981925@speedy:7] MixMonitor(SIP/10001-b7d71bd0, /var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-1 0001-20110914-163803.wav|bW(2)|/usr/bin/lame /var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-1 0001-20110914-163803.wav /var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-1 0001-20110914-163803.mp3 -b 16 -s 9.6 -m m --bitwidth 8 --lowpass 9.6 --resample 8 --lowpass-width 1 rm -f /var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-1 0001-20110914-163803.wav) in new stack This is part of the log with Asterisk 1.8.4 -- Executing [08129981925@speedy:7] MixMonitor(SIP/10001-001a, /var/spool/asterisk/recording/speedy/2011/09/14/ACCOUNT-08129981925-Admin_I T-10001-20110914-165248.wav,bW(2),/usr/bin/lame /var/spool/asterisk/recording/speedy/2011/09/14/ACCOUNT-08129981925-Admin_I T-10001-20110914-165248.wav /var/spool/asterisk) in new stack As you can see, with 1.8.4 the command paramater is trimed. Is there some changes / bug with MixMonitor in Asterisk 1.8.4 ? Is there a quick workaround for this problem ? Please help Thx Ikka Vertika Jakarta -Indonesia Alternatives: -- _ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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
[asterisk-users] asterisk in some kind of loop
I am using asterisk 1.8 from a few days ago and it goes into some kind of loop after maybe a couple of days of use. I compiled with debugging flags on and no optimize and I then attached gdb to the process. I did a backtrace and one for all threads and put it at http://pastebin.com/AGDsLdr7 . I would appreciate it if someone could take a look and tell me what can be done to fix this problem. Thanks in advance for any ideas. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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
[asterisk-users] dahdi and linux-2.6.38
Under linux-2.6.38 I was able to compile and install dahdi, however when I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have an old 400P card with one FXS and one FXO module. I have dahdi-trunk r9868 and dahdi-tools-trunk 8670. How can I get this to work correctly? Thanks in advance for any ideas. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi and linux-2.6.38
Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cov...@ccs.covici.com Sent: Tuesday, April 05, 2011 1:53 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi and linux-2.6.38 Under linux-2.6.38 I was able to compile and install dahdi, however when I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have an old 400P card with one FXS and one FXO module. I have dahdi-trunk r9868 and dahdi-tools-trunk 8670. How can I get this to work correctly? Thanks in advance for any ideas. You installed libpri ? I don't have any pri's. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi and linux-2.6.38
Shaun Ruffell sruff...@digium.com wrote: On 04/05/2011 01:52 PM, cov...@ccs.covici.com wrote: Under linux-2.6.38 I was able to compile and install dahdi, however when I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have an old 400P card with one FXS and one FXO module. I have dahdi-trunk r9868 and dahdi-tools-trunk 8670. How can I get this to work correctly? Thanks in advance for any ideas. After installing dahdi did you load your wctdm.ko driver? What is the output of 'cat /sys/module/dahdi/version'? What is the output from 'cat /proc/dahdi/1'? I did load the driver, but I am not booted into that system, so I cannot give you the other version info. I did make and make install and I will check to make sure it got to the correct place. And it looks like it did -- the dahdi-version.h has a time stamp about 2 minutes before the timestamp of the modules in the kernel I was using. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi and linux-2.6.38
Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cov...@ccs.covici.com Sent: Tuesday, April 05, 2011 2:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi and linux-2.6.38 Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cov...@ccs.covici.com Sent: Tuesday, April 05, 2011 1:53 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi and linux-2.6.38 Under linux-2.6.38 I was able to compile and install dahdi, however when I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have an old 400P card with one FXS and one FXO module. I have dahdi-trunk r9868 and dahdi-tools-trunk 8670. How can I get this to work correctly? Thanks in advance for any ideas. You installed libpri ? I don't have any pri's. I'd check the dmesg output - AFAIK you need libpri as a backbone for DAHDI (at least on some kernels). No dmesg output at all. Just when the modules were loaded, but not from the dahdi_cfg -vv -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meet me recording
Satish Patel satish...@hotmail.com wrote: Does it create separet file foreach channel? Or single one? -- Sent from my iPhone On Feb 19, 2011, at 12:45 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hi Satish, You can Pass 'r' flag to meetme Application and file will be recorded nothin to load mixmonitor and other Application on Channel, i think 'r' is better than all options Cheers Dhaval On Sat, Feb 19, 2011 at 1:37 AM, satish patel satish...@hotmail.com wrote: Thanks, look like monitor application resolved my issue. From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 18 Feb 2011 09:16:36 -0600 Subject: Re: [asterisk-users] Meet me recording From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of satish patel Sent: Friday, February 18, 2011 9:12 AM To: asterisk-users Subject: [asterisk-users] Meet me recording Hey Users, I am using record application to record MeetMe conf. but look like its creating individual files for every channel. What applucation is best to record MeetMe conf ? ~ # ls -l /var/spool/asterisk/monitor/ total 489220 -rw-r--r-- 1 asterisk asterisk 44 Feb 16 08:42 8881- conf-20110216-084224.wav -rw-r--r-- 1 asterisk asterisk 1858284 Feb 16 13:05 8881- conf-20110216-130321.wav -rw-r--r-- 1 asterisk asterisk 1604204 Feb 16 13:05 8881- conf-20110216-130337.wav -rw-r--r-- 1 asterisk asterisk 241964 Feb 17 08:20 8881- conf-20110217-081957.wav -rw-r--r-- 1 asterisk asterisk 78678124 Feb 17 11:12 8881- conf-20110217-095056.wav -rw-r--r-- 1 asterisk asterisk 612204 Feb 17 09:53 8881- conf-20110217-095310.wav -rw-r--r-- 1 asterisk asterisk 81183084 Feb 17 11:13 8881- conf-20110217-095414.wav -rw-r--r-- 1 asterisk asterisk 69488044 Feb 17 11:12 8881- conf-20110217-100012.wav -rw-r--r-- 1 asterisk asterisk 68917164 Feb 17 11:12 8881- conf-20110217-100052.wav -rw-r--r-- 1 asterisk asterisk 66971884 Feb 17 11:11 8881- conf-20110217-100117.wav -rw-r--r-- 1 asterisk asterisk 66648684 Feb 17 11:12 8881- conf-20110217-100327.wav -rw-r--r-- 1 asterisk asterisk 45056044 Feb 17 11:06 8881- conf-20110217-102007.wav Thanks, S From what I read, mixmonitor. It creates just one file for the conference. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] exceeds the maximum size of ast_fdset error on Asterisk-1.8.0
Benny Amorsen benny+use...@amorsen.dk wrote: Sorry for resurrecting an old thread... Tilghman Lesher writes: Out of curiosity, what platform are you running on? On most platforms that are able to run Asterisk, with the possible exception of Solaris, increasing the maximum file descriptor for use with select(2) is possible. I am not entirely sure yet, but it looks like Asterisk 1.8.x fails to increase the maximum file descriptor when running on Linux, if configure is not run as root. If configure is run as root, everything works as expected. Not so, I always run ./configure as root and I get the message that 32768 exceeds ... -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with chan_dahdi and conferencing
Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, Jan 16, 2011 at 02:30:42AM -0500, cov...@ccs.covici.com wrote: Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sat, Jan 15, 2011 at 11:50:32AM -0500, cov...@ccs.covici.com wrote: Hi. I am using asterisk-1.8 and I am having problems getting conferencing to work properly. I did modprobe on dahdi and did load = chan_dahdi.so in /etc/asterisk/modules.conf. Now I do get conferencing, but meetme says [Jan 15 11:38:56] WARNING[9214] app_meetme.c: No DAHDI channel available for conference, conference recording disabled (is chan_dahdi loaded?) What is the output of: dahdi show channels And the output is? Now chan_dahdi is indeed loaded, but I have an empty chan_dahdi.conf. Not even an empty '[channels]' section? I did put that just now, but I still get the same warning. Have you tried 'dahdi restart' ? I did module reload chan_dahdi.so but no joy. The output of dahdi show channels is just the header line with no channels. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with chan_dahdi and conferencing
Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, Jan 16, 2011 at 07:20:53AM -0500, cov...@ccs.covici.com wrote: Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, Jan 16, 2011 at 02:30:42AM -0500, cov...@ccs.covici.com wrote: Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sat, Jan 15, 2011 at 11:50:32AM -0500, cov...@ccs.covici.com wrote: Hi. I am using asterisk-1.8 and I am having problems getting conferencing to work properly. I did modprobe on dahdi and did load = chan_dahdi.so in /etc/asterisk/modules.conf. Now I do get conferencing, but meetme says [Jan 15 11:38:56] WARNING[9214] app_meetme.c: No DAHDI channel available for conference, conference recording disabled (is chan_dahdi loaded?) What is the output of: dahdi show channels And the output is? Now chan_dahdi is indeed loaded, but I have an empty chan_dahdi.conf. Not even an empty '[channels]' section? I did put that just now, but I still get the same warning. Have you tried 'dahdi restart' ? I did module reload chan_dahdi.so but no joy. 'module reload chan_dahdi.so' is not the same as: module unload chan_dahdi.so module load chan_dahdi.so Please try that one. The output of dahdi show channels is just the header line with no channels. OK. I'm looking for 'pseudo' there. Also: what's the output of the following in the Linux command-line: dahdi_test -c3 Opened pseudo dahdi interface, measuring accuracy... 99.993% 99.617% 99.996% --- Results after 3 passes --- Best: 99.996 -- Worst: 99.617 -- Average: 99.868716, Difference: 100.124381 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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
[asterisk-users] Problem with chan_dahdi and conferencing
Hi. I am using asterisk-1.8 and I am having problems getting conferencing to work properly. I did modprobe on dahdi and did load = chan_dahdi.so in /etc/asterisk/modules.conf. Now I do get conferencing, but meetme says [Jan 15 11:38:56] WARNING[9214] app_meetme.c: No DAHDI channel available for conference, conference recording disabled (is chan_dahdi loaded?) Now chan_dahdi is indeed loaded, but I have an empty chan_dahdi.conf. What do I need to do to get recording to work? Any assistance would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with chan_dahdi and conferencing
Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sat, Jan 15, 2011 at 11:50:32AM -0500, cov...@ccs.covici.com wrote: Hi. I am using asterisk-1.8 and I am having problems getting conferencing to work properly. I did modprobe on dahdi and did load = chan_dahdi.so in /etc/asterisk/modules.conf. Now I do get conferencing, but meetme says [Jan 15 11:38:56] WARNING[9214] app_meetme.c: No DAHDI channel available for conference, conference recording disabled (is chan_dahdi loaded?) What is the output of: dahdi show channels Now chan_dahdi is indeed loaded, but I have an empty chan_dahdi.conf. Not even an empty '[channels]' section? I did put that just now, but I still get the same warning. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] system lockup when going into conference
Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Thu, Jan 06, 2011 at 07:01:00PM -0500, cov...@ccs.covici.com wrote: Hi. I have an asterisk system under Debian Leni using asterisk 1.8 with no Digium hardware -- and when I go into a meetme conference the system either locks up or is 100% cpu utilized or something -- I can't type anything and I have to physically reboot the system. The dahdi module is loaded and the last log entry is the playing of you are the only person in this conference,. How would I even start to debug this one? Any ideas would be appreciated. What version of DAHDI? PProblem solved by Shaun fixing a regression in dahdi-trunk. Thanks to him its now working. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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
[asterisk-users] system lockup when going into conference
Hi. I have an asterisk system under Debian Leni using asterisk 1.8 with no Digium hardware -- and when I go into a meetme conference the system either locks up or is 100% cpu utilized or something -- I can't type anything and I have to physically reboot the system. The dahdi module is loaded and the last log entry is the playing of you are the only person in this conference,. How would I even start to debug this one? Any ideas would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems inserting dahdi modules using Debian Leni
; lsmod | grep dahdi FATAL: Error inserting dahdi (/lib/modules/2.6.26-2-686/dahdi/dahdi.ko): Unknown symbol in module, or unknown parameter (see dmesg) [25991.968325] dahdi: no symbol version for crc_ccitt_table [25991.968330] dahdi: Unknown symbol crc_ccitt_table -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems inserting dahdi modules using Debian Leni
inserting dahdi (/lib/modules/2.6.26-2-686/dahdi/dahdi.ko): Unknown symbol in module, or unknown parameter (see dmesg) [25991.968325] dahdi: no symbol version for crc_ccitt_table [25991.968330] dahdi: Unknown symbol crc_ccitt_table So everything appears right. Have you make clean; make install in your dahdi-linux directory since rebuilding and rebooting into your new kernel? Otherwise, since crc_ccitt_table is in your /proc/kallsyms, you could just 'modprobe --force dahdi' to bypass the symbol version check. I did try that and it worked for a while and locked up the system! But I will do a make clean, etc in dahdi and see if that does anything. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems inserting dahdi modules using Debian Leni
inserting dahdi (/lib/modules/2.6.26-2-686/dahdi/dahdi.ko): Unknown symbol in module, or unknown parameter (see dmesg) [25991.968325] dahdi: no symbol version for crc_ccitt_table [25991.968330] dahdi: Unknown symbol crc_ccitt_table So everything appears right. Have you make clean; make install in your dahdi-linux directory since rebuilding and rebooting into your new kernel? Otherwise, since crc_ccitt_table is in your /proc/kallsyms, you could just 'modprobe --force dahdi' to bypass the symbol version check. Well, that finally did the trick, the make clean, make and make install, so I have no clue as to what it was, but now I will try a conference. Thhanks so much for all your help. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems inserting dahdi modules using Debian Leni
inserting dahdi (/lib/modules/2.6.26-2-686/dahdi/dahdi.ko): Unknown symbol in module, or unknown parameter (see dmesg) [25991.968325] dahdi: no symbol version for crc_ccitt_table [25991.968330] dahdi: Unknown symbol crc_ccitt_table So everything appears right. Have you make clean; make install in your dahdi-linux directory since rebuilding and rebooting into your new kernel? Otherwise, since crc_ccitt_table is in your /proc/kallsyms, you could just 'modprobe --force dahdi' to bypass the symbol version check. Well, the saga goes on -- after all that the system crashes -- not sure whether its a loop or actual crash the minute it says you are the only one in this conference and the play is the last thing in the log file. What would be the best way to debug this? I can see the compiler flags and I can put the debug flags on and do no optomize, but if its a loop, would that be of any use? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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
[asterisk-users] problems inserting dahdi modules using Debian Leni
Hi. I have a Debian Leni system with asterisk-1.8. I was trying to get meetme to work and it depends on dahdi, so I compiled dahdi-trunk and dahdi-tools-trunk, however, when trying to insert dahdi_dummy, it complained about symbol crc_ccitt_table, although the module was actually there in the kernel tree. So, I took the Debian source, and I had the config and I did make Bzimage, make modules and make modules_install, but dahdi_dummy still complains about the same symbol, it says no version for that symbol, so I am confused as to how to resolve this so I can modprobe dahdi_dummy properly. Any ideas would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems inserting dahdi modules using Debian Leni
Shaun Ruffell sruff...@digium.com wrote: On 01/04/2011 05:09 PM, cov...@ccs.covici.com wrote: Hi. I have a Debian Leni system with asterisk-1.8. I was trying to get meetme to work and it depends on dahdi, so I compiled dahdi-trunk and dahdi-tools-trunk, however, when trying to insert dahdi_dummy, it complained about symbol crc_ccitt_table, although the module was actually there in the kernel tree. So, I took the Debian source, and I had the config and I did make Bzimage, make modules and make modules_install, but dahdi_dummy still complains about the same symbol, it says no version for that symbol, so I am confused as to how to resolve this so I can modprobe dahdi_dummy properly. Any ideas would be appreciated. First off, I recommend using dahdi-linux 2.4.0 *without* compiling dahdi_dummy. A dummy span is no longer needed for DAHDI to provide a timing source to asterisk. But you'll still need crc_ccitt module for dahdi to load, so that doesn't fix the problem as you describe here. If you rebuilt your kernel (which probably wasn't necessary...) you need to reboot into the new kernel, then rebuild DAHDI against your running kernel in order to load. Sounds like you have built DAHDI against one version of the kernel and you're running against another one. Also...make sure you're using modprobe and not insmod to load the driver...so that crc_ccitt will automatically be loaded as a dependency. For example you can see it automatically loaded here (and how dahdi_dummy isn't needed for timing). ]# lsmod | grep crc_ccitt ]# dahdi_test -c 1 Unable to open dahdi interface: No such file or directory ]# modprobe dahdi ]# lsmod | grep crc_ccitt crc_ccitt 10240 1 dahdi ]# dahdi_test -c 5 Opened pseudo dahdi interface, measuring accuracy... 99.998% 99.981% 99.990% 99.990% 99.991% --- Results after 5 passes --- Best: 99.998 -- Worst: 99.981 -- Average: 99.990100, Difference: 99.990101 ]# I did rebuild the kernel, it has the same version and the same config as the old one and it did build a crc_ccitt module, and I even rebooted the system with the new modules, but no joy at all. Igot the same results whether I rebuilt the kernel or not, so this is what is confusing to me. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems inserting dahdi modules using Debian Leni
Shaun Ruffell sruff...@digium.com wrote: On 1/4/11 9:26 PM, cov...@ccs.covici.com wrote: Shaun Ruffellsruff...@digium.com wrote: On 01/04/2011 05:09 PM, cov...@ccs.covici.com wrote: Hi. I have a Debian Leni system with asterisk-1.8. I was trying to get meetme to work and it depends on dahdi, so I compiled dahdi-trunk and dahdi-tools-trunk, however, when trying to insert dahdi_dummy, it complained about symbol crc_ccitt_table, although the module was actually there in the kernel tree. So, I took the Debian source, and I had the config and I did make Bzimage, make modules and make modules_install, but dahdi_dummy still complains about the same symbol, it says no version for that symbol, so I am confused as to how to resolve this so I can modprobe dahdi_dummy properly. Any ideas would be appreciated. First off, I recommend using dahdi-linux 2.4.0 *without* compiling dahdi_dummy. A dummy span is no longer needed for DAHDI to provide a timing source to asterisk. But you'll still need crc_ccitt module for dahdi to load, so that doesn't fix the problem as you describe here. If you rebuilt your kernel (which probably wasn't necessary...) you need to reboot into the new kernel, then rebuild DAHDI against your running kernel in order to load. Sounds like you have built DAHDI against one version of the kernel and you're running against another one. Also...make sure you're using modprobe and not insmod to load the driver...so that crc_ccitt will automatically be loaded as a dependency. For example you can see it automatically loaded here (and how dahdi_dummy isn't needed for timing). ]# lsmod | grep crc_ccitt ]# dahdi_test -c 1 Unable to open dahdi interface: No such file or directory ]# modprobe dahdi ]# lsmod | grep crc_ccitt crc_ccitt 10240 1 dahdi ]# dahdi_test -c 5 Opened pseudo dahdi interface, measuring accuracy... 99.998% 99.981% 99.990% 99.990% 99.991% --- Results after 5 passes --- Best: 99.998 -- Worst: 99.981 -- Average: 99.990100, Difference: 99.990101 ]# I did rebuild the kernel, it has the same version and the same config as the old one and it did build a crc_ccitt module, and I even rebooted the system with the new modules, but no joy at all. Igot the same results whether I rebuilt the kernel or not, so this is what is confusing to me. What you get from the following commands: ]# lsmod | grep crc_ccitt I had to modprobe it, but I got: crc_ccitt 2080 0 ]# modinfo crc_ccitt filename: /lib/modules/2.6.26-2-686/kernel/lib/crc-ccitt.ko license:GPL description:CRC-CCITT calculations depends: vermagic: 2.6.26-2-686 SMP mod_unload modversions 686 ]# uname -a Linux eirasterisk 2.6.26-2-686 #3 SMP Tue Jan 4 15:29:02 EST 2011 i686 GNU/Linux ]# cat /proc/kallsyms | grep crc_ccitt a crc-ccitt.c [crc_ccitt] f8c6d284 ? __mod_license69 [crc_ccitt] f8c6d290 ? __mod_description68 [crc_ccitt] f8c72250 r __ksymtab_crc_ccitt [crc_ccitt] f8c72268 r __kstrtab_crc_ccitt [crc_ccitt] f8c72260 r __kcrctab_crc_ccitt [crc_ccitt] f8c72258 r __ksymtab_crc_ccitt_table[crc_ccitt] f8c72272 r __kstrtab_crc_ccitt_table[crc_ccitt] f8c72264 r __kcrctab_crc_ccitt_table[crc_ccitt] a crc-ccitt.mod.c [crc_ccitt] f8c6d2b4 ? __module_depends [crc_ccitt] f8c6d32c ? versions [crc_ccitt] f8c6d2c0 ? __mod_vermagic5 [crc_ccitt] f8c725e0 d __this_module[crc_ccitt] 3771b461 a __crc_crc_ccitt [crc_ccitt] f8c72000 T crc_ccitt[crc_ccitt] 75811312 a __crc_crc_ccitt_table[crc_ccitt] f8c72050 R crc_ccitt_table [crc_ccitt] ]# modinfo dahdi filename: /lib/modules/2.6.26-2-686/dahdi/dahdi.ko version:SVN-trunk-r9614 alias: dahdi_dummy license:GPL v2 description:DAHDI Telephony Interface author: Mark Spencer marks...@digium.com srcversion: A63E42F5ADDDE39777BCC24 depends: vermagic: 2.6.26-2-686 SMP mod_unload modversions 686 parm: debug:Sets debugging verbosity as a bitfield, to see general debugging set this to 1. To see RBS debugging set this to 32 (int) parm: deftaps:int parm: max_pseudo_channels:Maximum number of pseudo channels. (int) -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URGENT Help needed
Do your asterisk logs say anything -- /var/log/asterisk/messages or full? Also, what happens if you do asterisk -c this may help you figure things out. Michael voip.quest...@gmail.com wrote: Hello, We tried to upgrade our Asterisk from 1.6.2.13 to 1.6.2.14, after trying to install iksemel (jabber support) and spandsp, but now Asterisk doesn't work anymore and we can't get it to run, althorugh we tried to remove it completely and reinstall 1.6.2.13. when trying to start it via /etc/init.d/asterisk start we get the following error: Asterisk died with code 1. Automatically restarting Asterisk. Asterisk ended with exit status 1 When just trying to run it as asterisk from the command line, we don't see the process being active and we get this message when running asterisk -r, although the file is present: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) Any help would be highly appreciated. Thank you in advance, Michael -- _ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Unix Connection
Do you have freepbx anywhere it always tries to connect -- via a socket I think and it usually uses the manager, so if you disable the manager it will break things. Also take the port stanza off of the tcpdump and you will soon see what is connecting. You will get other stuff, but this will tell you. Dan Journo d...@keshercommunications.com wrote: Some service is definitely connecting to your asterisk using AMI. Such services use username/password described in manager.conf. Usually its is some monitoring service. Although the message says 'remote UNIX connection' but it can be very well something from localhost. I would suggest to use tcpdump to find out the IP of this service. AMI uses TCP port 5038. I ran the following command and waited for the cli to show the remote unix connection message a few times. [r...@sip2 ~]# tcpdump port 5038 -w tcpdump.log -s0 tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes The result was 0 packets captured 0 packets received by filter 0 packets dropped by kernel Therefore, it seems like nothing is connecting to the AMI? Also, in manager.conf enabled=no Any other ideas? Is this a bug? Thanks Dan Alternatives: -- _ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
Dan Journo d...@keshercommunications.com wrote: Based on this, your call is probably getting to the provider as ulaw (the alaw is thrown out since it isn't in both selections; if you are in U.S. you don't need the alaw). Try the call with higher debug (at least 5) and verify which one is being selected. debug 5 doesnt give me any info regarding the codec. By the way, i'm using asterisk 1.4.36 if that makes any difference. I would suggest log dtmf in your logger.conf and put rtp debug on and see if its sending dtmf. Also call the provider and see if they hear the tones. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rtp problem with 1.8.0-rdc1
Leif Madsen leif.mad...@asteriskdocs.org wrote: On 10-09-23 05:40 PM, cov...@ccs.covici.com wrote: Hi. I am having a very strange problem --aren't they all -- with the release candidate. I have softphone which talks to asterisk from behind nat -- the asterisk is on a public ip -- and when I hit mute on the softphone, all rtp traffic ceases! Now, a version which does work is r281875, this does not happen in that vrsion, but right after that this strange thing starts and is not fixed in the current one. Any assistance here would be appreciated. We're probably going to need some sort of debugging information such as a console trace and SIP (I assume chan_sip) debug. More information here: doc/HOWTO_collect_debug_information.txt Leif. I certainly can do a sip set debug, is that what you need? I did do an rtp set debug and this is how I found out that when I hit the mute button on the soft phone all rtp traffic ceased between the phone and the asterisk box. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rtp problem with 1.8.0-rdc1
Benny Amorsen benny+use...@amorsen.dk wrote: cov...@ccs.covici.com writes: Hi. I am having a very strange problem --aren't they all -- with the release candidate. I have softphone which talks to asterisk from behind nat -- the asterisk is on a public ip -- and when I hit mute on the softphone, all rtp traffic ceases! Now, a version which does work is r281875, this does not happen in that vrsion, but right after that this strange thing starts and is not fixed in the current one. Why is it a problem? It sounds like Asterisk does silence suppression. But it surpresses in both directions! I still want to hear the other end. For a test is there a way to turn off that feature to see if that is the cause? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rtp problem with 1.8.0-rdc1
Lyle Giese l...@lcrcomputer.net wrote: Benny Amorsen wrote: cov...@ccs.covici.com writes: Hi. I am having a very strange problem --aren't they all -- with the release candidate. I have softphone which talks to asterisk from behind nat -- the asterisk is on a public ip -- and when I hit mute on the softphone, all rtp traffic ceases! Now, a version which does work is r281875, this does not happen in that vrsion, but right after that this strange thing starts and is not fixed in the current one. Why is it a problem? It sounds like Asterisk does silence suppression. /Benny 1) With no rtp traffic, the nat device will drop the connection in it's nat table and thus disconnecting the softphone from Asterisk. (after the router's timeout period of course) 2) The other issue is you are connected to a conference call and you want to mute your transmitter while listening to the conference. This is my issue, I am on a conference and mute myself, but I still want to hear the other end and asterisk is cutting off both ends audio. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rtp problem with 1.8.0-rdc1
Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hi! Why is it a problem? It sounds like Asterisk does silence suppression. 1) With no rtp traffic, the nat device will drop the connection in it's nat table and thus disconnecting the softphone from Asterisk. (after the router's timeout period of course) 2) The other issue is you are connected to a conference call and you want to mute your transmitter while listening to the conference. Set internaltiming to yes in asterisk.conf and see if that helps. In addition you might also be able to change the mute behaviour of your SIP clients so that it keeps on sending silent RTP packets. I cannot change the soft phone, so this is why I need asterisk to behave properly or at least have an option to behave differently -- and it did work up to a point and then they fixed something. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rtp problem with 1.8.0-rdc1
All I have to do to make it work is to use 1.8.0 revision 281875 -- after that something is broke. I was hoping someone could look and see what changed just after that rev and see if it makes sense. Benny Amorsen benny+use...@amorsen.dk wrote: cov...@ccs.covici.com writes: But it surpresses in both directions! I still want to hear the other end. For a test is there a way to turn off that feature to see if that is the cause? Ah, so it isn't Asterisk doing silence suppression, it's Asterisk being unable to handle that other devices do. If you switch to 1.6.2.x and enable internal-timing, you should have a shot at getting it working. /Benny -- _ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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
[asterisk-users] rtp problem with 1.8.0-rdc1
Hi. I am having a very strange problem --aren't they all -- with the release candidate. I have softphone which talks to asterisk from behind nat -- the asterisk is on a public ip -- and when I hit mute on the softphone, all rtp traffic ceases! Now, a version which does work is r281875, this does not happen in that vrsion, but right after that this strange thing starts and is not fixed in the current one. Any assistance here would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-1.8 problem with one-way audio with no nat
Tilghman Lesher tles...@digium.com wrote: On Thursday 02 September 2010 01:13:35 cov...@ccs.covici.com wrote: Matt Riddell li...@venturevoip.com wrote: On 25/08/10 8:54 PM, cov...@ccs.covici.com wrote: Hi. I have a soft phone -- expresstalk-- on a computer in my network and I use the internal ip address of the asterisk box to register the phone. But using asterisk-1.8 between revisions 281912 and 281982 it breaks -- after a few seconds of the call, I lose audio from the asterisk box to my soft phone, but not the other way around. This looks like one commit, but obviously I would like to know what's going on here? What's in the commit? Its the 282911 commit seems to break audio to the soft phone, but not to my ata -- very strange. That doesn't make any sense. Revision 282911 is a merge to a team branch, nothing related to the 1.8 branch. Maybe 282891 (same change, but to the 1.8 branch)? Or did you fat finger the revision? That was the one next in the logs, maybe I will try latest and see if it goes away. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-1.8 problem with one-way audio with no nat
Tilghman Lesher tles...@digium.com wrote: On Thursday 02 September 2010 01:13:35 cov...@ccs.covici.com wrote: Matt Riddell li...@venturevoip.com wrote: On 25/08/10 8:54 PM, cov...@ccs.covici.com wrote: Hi. I have a soft phone -- expresstalk-- on a computer in my network and I use the internal ip address of the asterisk box to register the phone. But using asterisk-1.8 between revisions 281912 and 281982 it breaks -- after a few seconds of the call, I lose audio from the asterisk box to my soft phone, but not the other way around. This looks like one commit, but obviously I would like to know what's going on here? What's in the commit? Its the 282911 commit seems to break audio to the soft phone, but not to my ata -- very strange. That doesn't make any sense. Revision 282911 is a merge to a team branch, nothing related to the 1.8 branch. Maybe 282891 (same change, but to the 1.8 branch)? Or did you fat finger the revision? Or to put it another way the last good install for me is 281875 so it right after that where from express talk to an outside line through asterisk is failing with one way audio after the first several seconds. I did try latest update and it is still failing. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-1.8 problem with one-way audio with no nat
Matt Riddell li...@venturevoip.com wrote: On 25/08/10 8:54 PM, cov...@ccs.covici.com wrote: Hi. I have a soft phone -- expresstalk-- on a computer in my network and I use the internal ip address of the asterisk box to register the phone. But using asterisk-1.8 between revisions 281912 and 281982 it breaks -- after a few seconds of the call, I lose audio from the asterisk box to my soft phone, but not the other way around. This looks like one commit, but obviously I would like to know what's going on here? What's in the commit? Its the 282911 commit seems to break audio to the soft phone, but not to my ata -- very strange. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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
[asterisk-users] asterisk-1.8 problem with one-way audio with no nat
Hi. I have a soft phone -- expresstalk-- on a computer in my network and I use the internal ip address of the asterisk box to register the phone. But using asterisk-1.8 between revisions 281912 and 281982 it breaks -- after a few seconds of the call, I lose audio from the asterisk box to my soft phone, but not the other way around. This looks like one commit, but obviously I would like to know what's going on here? Thanks in advance for any ideas. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with meetme in 1.4.26
Danny Nicholas da...@debsinc.com wrote: Hi list, I was going through my dialplan today and found these 2 oddities with meetme using DAHDI to join the conference. 1. Although music on hold is indicated, I don't get any sound until I press *. Then the conference menu plays and all is well except - 2. According to the instructions, pressing 8 should leave the conference. No dice. 8 exits the menu, not the conference. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8.0 beta2: courtesy tone being played to callee
Leif Madsen leif.mad...@asteriskdocs.org wrote: On 7/29/2010 8:30 PM, cov...@ccs.covici.com wrote: Hi. I am using *1 in features to initiate a mix monitor recording. However, when I hit *1, the callee hears the courtesy tone which I have, so I know when the recording is started or stopped. This is a problem, particularly in automated system where the beep is mistaken for a tone or other problems. Should I file a bug, or is this going to be fixed? This doesn't really sound like a bug to me, but it's hard to tell without any debugging information. Please provide the configuration you're using along with the console output of the dialplan showing what is happening during a call. Likely because you're executing (a macro?) on the other channel, that whatever tone you're executing is being played to the other channel because you're executing the entire feature on the other channel. I'm just speculating at this point though. Well, this did not happen in 1.6.2, so I figured it was a regression. Here are the lines from the log once the call was answered and after the dtmf for the *1 [Jul 29 20:31:03] VERBOSE[22300] file.c: -- SIP/202-005b Playing 'beep.gsm' (language 'en') [Jul 29 20:31:04] VERBOSE[22300] file.c: -- SIP/flowroute-005c Playing 'beep.gsm' (language 'en') Why the second line? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8.0 beta2: courtesy tone being played to callee
Leif Madsen leif.mad...@asteriskdocs.org wrote: On 7/31/2010 11:56 AM, cov...@ccs.covici.com wrote: Leif Madsenleif.mad...@asteriskdocs.org wrote: On 7/29/2010 8:30 PM, cov...@ccs.covici.com wrote: Hi. I am using *1 in features to initiate a mix monitor recording. However, when I hit *1, the callee hears the courtesy tone which I have, so I know when the recording is started or stopped. This is a problem, particularly in automated system where the beep is mistaken for a tone or other problems. Should I file a bug, or is this going to be fixed? This doesn't really sound like a bug to me, but it's hard to tell without any debugging information. Please provide the configuration you're using along with the console output of the dialplan showing what is happening during a call. Likely because you're executing (a macro?) on the other channel, that whatever tone you're executing is being played to the other channel because you're executing the entire feature on the other channel. I'm just speculating at this point though. Well, this did not happen in 1.6.2, so I figured it was a regression. Here are the lines from the log once the call was answered and after the dtmf for the *1 [Jul 29 20:31:03] VERBOSE[22300] file.c: --SIP/202-005b Playing 'beep.gsm' (language 'en') [Jul 29 20:31:04] VERBOSE[22300] file.c: --SIP/flowroute-005c Playing 'beep.gsm' (language 'en') Why the second line? Hard to say without the information previously asked for. Which particular configuration files do you need? I am using freepbx so I certainly cannot give youall the configs. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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
[asterisk-users] 1.8.0 beta2: courtesy tone being played to callee
Hi. I am using *1 in features to initiate a mix monitor recording. However, when I hit *1, the callee hears the courtesy tone which I have, so I know when the recording is started or stopped. This is a problem, particularly in automated system where the beep is mistaken for a tone or other problems. Should I file a bug, or is this going to be fixed? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] undocumented change in expression handling in 1.8 beta
Paul Belanger paul.belan...@polybeacon.com wrote: On Sun, Jul 25, 2010 at 12:52 AM, cov...@ccs.covici.com wrote: Hi. I hava a variable and in 1.6 I set the string variable to and it got the null string. In 1.8, it gets the quotes, I have to set it to nothing at all to make it get the null value. Post an example for working (1.6) and not working (1.8) OK, the line actually is: exten = s,n(auth),Set(password=) which sets to a null value in 1.6.2 but does not in 1.8. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] getting some segmentation faults with 1.8
Paul Belanger paul.belan...@polybeacon.com wrote: On Sun, Jul 25, 2010 at 9:34 AM, unsero...@aol.com wrote: I am also getting segmentation fault when doing a reload from CLI. I believe this is your issue : https://issues.asterisk.org/view.php?id=17704 If not, create a new issue on the tracker with an unoptimized backtrace. That fixed my module reload seg fault as well. Thanks. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] undocumented change in expression handling in 1.8 beta
Paul Belanger paul.belan...@polybeacon.com wrote:
On Sun, Jul 25, 2010 at 11:39 AM, cov...@ccs.covici.com wrote:
OK, the line actually is:
exten = s,n(auth),Set(password=) which sets to a null value in 1.6.2
but does not in 1.8.
It's possible something changed, how are you checking if the
${password} is null? Post an example dialplan that works in 1.6.2 and
does not in 1.8.
Well, when I executed the line
exten = s,n,SET(password=${password}${digit});add a digit
and when
I wanted to test the
variable
exten =
s,n(pswd_done),GotoIf($[x${password}=x14036]?custom-conf8200,s,1)
I got the following error in 1.8:
[Jul 25 00:36:14] WARNING[7242] ast_expr2.fl: ast_yyerror(): syntax
error: syntax error, unexpected 'token', expecting $end; Input:
x14036=x14036
^
After the was gone in the set, it works in 1.8, have not tried in
1.6.
--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?
John Covici
cov...@ccs.covici.com
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Re: [asterisk-users] undocumented change in expression handling in 1.8 beta
Tilghman Lesher tles...@digium.com wrote: On Saturday 24 July 2010 23:52:39 cov...@ccs.covici.com wrote: Hi. I hava a variable and in 1.6 I set the string variable to and it got the null string. In 1.8, it gets the quotes, I have to set it to nothing at all to make it get the null value. Please read the 6th item in UPGRADE.txt. The sixth item I have is: * The default behavior for Set, AGI, and pbx_realtime has been changed to implement 1.6 behavior by default, if there is no [compat] section in asterisk.conf. In prior versions, the behavior defaulted to 1.4 behavior, to assist in upgrades. Specifically, that means that pbx_realtime and res_agi expect you to use commas to separate arguments in applications, and Set only takes a single pair of a variable name/value. The old 1.4 behavior may still be obtained by setting app_set, pbx_realtime, and res_agi each to 1.4 in the [compat] section of asterisk.conf. I guess that referrs to the following in the 1.6 upgrade file: You now only need to quote strings in configuration files if you literally want quotation marks within a string. Thanks much for that clarification. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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
