[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] A reason TO run Asterisk as root
on Wednesday 07/22/2009 Gordon Henderson(gordon+aster...@drogon.net) wrote On Wed, 22 Jul 2009, Jonathan Moore wrote: On Wed, Jul 22, 2009 at 10:31 AM, Steve Edwardsasterisk@sedwards.com wrote: I finally found a reason TO run Asterisk as root. By default, ext[23] file systems reserve 5% of the filesystem for root. Thus, you may get some warning when everything non-root starts failing and give you a chance to free up some space before Asterisk is affected. Couldn't you get the same effect using quotas? Also, using separate partitions for various parts of the filesystem is a nice addition. Having your /var/log somewhere besides the same partition as / helps keep runaway logs at bay, just as an example. This is real sysadmin territory And it's a dying art, I fear. Too many people just creating one big partition, doing stupid (IMO) tricks like tune2fs -m 0 ... and so on. It's something you can't/won't ever learn from just doing a modern Linux install, or (worse, I reckon), installing something like pbxinaflash, etc. although to their credit, most of these pre-canned installs do seem to work well. Until they break. Then you need a sysadmin... I do agree, but I do change the reserved blocks to 0, otherwise even as root the DF numbers are wrong and I have a number of partitions, even one for /tmp, so I figure its not so bad. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Step-by-Step Asterisk and MeetMe Help
on Monday 05/18/2009 Jimmy Ezell(jez...@hmhca.com) wrote Time to spotlight my ignorance here. I am up to post 9 on my Cisco Gateway to Asterisk Step by Step posts. http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html The next post is about MeetMe but I am running into trouble. I am pretty sure the ztdummy thing is the problem but I have no idea why. From my /var/log/boot.log May 18 14:46:59 localhost modprobe: FATAL: Module zaptel not found. May 18 14:46:59 localhost zaptel: Loading zaptel framework: failed On boot I also see Waiting for zap to come online...Error: missing /dev/zap! When I do a make and make install in the Zaptel directory (zaptel-1.4.12.1), I get Zaptel installed successfully. I saw a similar post in archives saying they would work offline and post the answer, but I did not see an answer posted. http://lists.digium.com/pipermail/asterisk-users/2006-March/142876.html What do you suggest? I would modprobe ztdummy and see what happens and look at the log and lsmod after you do that. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help
This is your problem -- check the compile of the zaptel package and see what happened. It might not compile it if you have a hardware zaptel crd, otherwise it should have done so. on Monday 05/18/2009 Jimmy Ezell(jez...@hmhca.com) wrote I would modprobe ztdummy and see what happens and look at the log and lsmod after you do that. -- 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 John, modprobe comes back right away with ztdummy not found [r...@localhost zaptel-1.4.12.1]# modprobe ztdummy FATAL: Module ztdummy not found. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help
You should not have removed all kernel modules, at least zaptel, but I keep them anyway. on Monday 05/18/2009 Jimmy Ezell(jez...@hmhca.com) wrote From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]on Behalf Of John covici Sent: Monday, May 18, 2009 05:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help This is your problem -- check the compile of the zaptel package and see what happened. It might not compile it if you have a hardware zaptel crd, otherwise it should have done so. I stopped asterisk. I have no card. From the zaptel directory I ran # make distclean # make clean # ./configure # make menuselect In menuselect I removed all Kernel Modules except for ztdummy # make --output of make command -- [r...@localhost zaptel-1.4.12.1]# make make[1]: Entering directory `/usr/src/zaptel-1.4.12.1' make -C /usr/src/linux-2.4 ARCH=i386 SUBDIRS=/usr/src/zaptel-1.4.12.1/kernel HOTPLUG_FIRMWARE=yes KBUILD_OBJ_M=zaptel.o ztdummy. o modules make[2]: Entering directory `/usr/src/kernels/2.6.9-78.0.13.EL-smp-i686' CC [M] /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.o /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c: In function `__zt_receive_chunk': /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:6703: warning: 'res' might be used uninitialized in this function LD [M] /usr/src/zaptel-1.4.12.1/kernel/zaptel.o CC [M] /usr/src/zaptel-1.4.12.1/kernel/ztdummy.o Building modules, stage 2. MODPOST CC /usr/src/zaptel-1.4.12.1/kernel/zaptel.mod.o LD [M] /usr/src/zaptel-1.4.12.1/kernel/zaptel.ko CC /usr/src/zaptel-1.4.12.1/kernel/ztdummy.mod.o LD [M] /usr/src/zaptel-1.4.12.1/kernel/ztdummy.ko make[2]: Leaving directory `/usr/src/kernels/2.6.9-78.0.13.EL-smp-i686' gcc -c -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o zonedata.lo zonedata.c gcc -c -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o tonezone.lo tonezone.c gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -shared -Wl,-sonam e,libtonezone.so.1.0 -o libtonezone.so zonedata.lo tonezone.lo -lm gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o zonedata.o zonedata.c gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o tonezone.o tonezone.c ar rcs libtonezone.a zonedata.o tonezone.o ranlib libtonezone.a gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o fxotune.o -c fx otune.c gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o fxotune fxotune .o -lm gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o ztcfg.o -c ztcf g.c gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o ztcfg ztcfg.o l ibtonezone.a -lm gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o ztmonitor.o -c ztmonitor.c gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o ztmonitor ztmon itor.o gcc -o ztspeed.o -c ztspeed.c gcc -o ztspeed ztspeed.o gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o zttest.o -c ztt est.c gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o zttest zttest.o gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o ztscan.o -c zts
Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help
On reading your mail more carefully, it looks like you do have ztdummy, do depmod -a and then try to modprobe it. on Monday 05/18/2009 Jimmy Ezell(jez...@hmhca.com) wrote From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]on Behalf Of John covici Sent: Monday, May 18, 2009 05:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help This is your problem -- check the compile of the zaptel package and see what happened. It might not compile it if you have a hardware zaptel crd, otherwise it should have done so. I stopped asterisk. I have no card. From the zaptel directory I ran # make distclean # make clean # ./configure # make menuselect In menuselect I removed all Kernel Modules except for ztdummy # make --output of make command -- [r...@localhost zaptel-1.4.12.1]# make make[1]: Entering directory `/usr/src/zaptel-1.4.12.1' make -C /usr/src/linux-2.4 ARCH=i386 SUBDIRS=/usr/src/zaptel-1.4.12.1/kernel HOTPLUG_FIRMWARE=yes KBUILD_OBJ_M=zaptel.o ztdummy. o modules make[2]: Entering directory `/usr/src/kernels/2.6.9-78.0.13.EL-smp-i686' CC [M] /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.o /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c: In function `__zt_receive_chunk': /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:6703: warning: 'res' might be used uninitialized in this function LD [M] /usr/src/zaptel-1.4.12.1/kernel/zaptel.o CC [M] /usr/src/zaptel-1.4.12.1/kernel/ztdummy.o Building modules, stage 2. MODPOST CC /usr/src/zaptel-1.4.12.1/kernel/zaptel.mod.o LD [M] /usr/src/zaptel-1.4.12.1/kernel/zaptel.ko CC /usr/src/zaptel-1.4.12.1/kernel/ztdummy.mod.o LD [M] /usr/src/zaptel-1.4.12.1/kernel/ztdummy.ko make[2]: Leaving directory `/usr/src/kernels/2.6.9-78.0.13.EL-smp-i686' gcc -c -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o zonedata.lo zonedata.c gcc -c -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o tonezone.lo tonezone.c gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -shared -Wl,-sonam e,libtonezone.so.1.0 -o libtonezone.so zonedata.lo tonezone.lo -lm gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o zonedata.o zonedata.c gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o tonezone.o tonezone.c ar rcs libtonezone.a zonedata.o tonezone.o ranlib libtonezone.a gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o fxotune.o -c fx otune.c gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o fxotune fxotune .o -lm gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o ztcfg.o -c ztcf g.c gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o ztcfg ztcfg.o l ibtonezone.a -lm gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o ztmonitor.o -c ztmonitor.c gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o ztmonitor ztmon itor.o gcc -o ztspeed.o -c ztspeed.c gcc -o ztspeed ztspeed.o gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o zttest.o -c ztt est.c gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o zttest zttest.o gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o ztscan.o -c zts
Re: [asterisk-users] duration of rfc2833 generated dtmf
Well, this solution seemed not to work for me, maybe because I did not set the minimum and also if I am using a sip phone or ATA, the solution would not apply -- correct me if I am wrong on either of these. on Wednesday 04/15/2009 Mark G. Thomas(m...@misty.com) wrote Hi, On Mon, Apr 13, 2009 at 05:32:45PM -0400, John covici wrote: Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. I had a similar problem. Adding dtmf to the console = line in logger.conf is tremendously helpful in diagnosing the dtmf behavior. My successful work-around was to recompile asterisk with the following adjustments. Asterisk then extends the duration of the short tones. I'm puzzled why these aren't a run-time configuration settings, since I'd think this would be a common problem. [r...@sylvester asterisk-1.4.24]# diff main/channel.c_orig main/channel.c 91c91 #define AST_DEFAULT_EMULATE_DTMF_DURATION 100 --- #define AST_DEFAULT_EMULATE_DTMF_DURATION 150 94c94 #define AST_MIN_DTMF_DURATION 80 --- #define AST_MIN_DTMF_DURATION 150 I also later got my provider (Vitelity) to provision my service on a different server of theirs, which then also seemed to improve both their RFC2833 DTMF reliability and duration. -Mark -- Mark G. Thomas (m...@misty.com) voice: 215-591-3695 http://mail-cleaner.com/ -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] duration of rfc2833 generated dtmf
OK, thanks. If I could convince them to use info, would that be better as far as the duration is concerned? on Monday 04/13/2009 Brent Davidson(br...@texascountrytitle.com) wrote John covici wrote: Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. If your provider insists on rfc2833, then their servers will be responsible for setting the tone duration sent to PSTN lines. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] duration of rfc2833 generated dtmf
on Tuesday 04/14/2009 Kristian Kielhofner(kristian.kielhof...@gmail.com) wrote On Mon, Apr 13, 2009 at 5:32 PM, John covici cov...@ccs.covici.com wrote: Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. 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 John, Assuming this is Asterisk 1.4 or later... The duration used by Asterisk is the same duration sent from the phone. The duration of those DTMF key presses should match the time the user is holding down the key. What type(s) of phones are these? You should also look into using Asterisk 1.4.24.1 or later (if you aren't already). There have been many improvements to the RTP code to better handle quirks with the equipment (especially Sonus) used by various providers. Assuming your provider is to spec (and so is your phone) your provider should not be complaining that the duration of your DTMF key presses are too short... With that being said AFAIK there is no way to specify a minimum duration for an RFC 2833 DTMF in Asterisk on a bridged channel. OK, thanks for that info -- but it seems to me no matter how long I press the keys on the phone, (connected to a Digium board) the other end gets the same duration. Now, the problems I run into are not dialing the phone number, but dtmf on the call such as an IVR. Some of them don't like what seems to be a too short key press, whereas if I call the same number from the cell phone, there is no problem. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] duration of rfc2833 generated dtmf
Thanks -- can not find sip dtmf mode or sip dtmfmode in asterisk-1.4. Is this new in 1.6? on Tuesday 04/14/2009 Brent Davidson(br...@texascountrytitle.com) wrote To the best of my knowledge, the only way for you to control the duration sent to the PSTN lines is for you to be directly connected to the lines so you can set the tone duration in zapata.conf / dahdi.conf or to use inband signalling. One thing you might try is researching the SipDtmfMode command. It allows you to change the DTMF mode on an active channel. A suggestion might be to set up the dial command with the M() option that point to a Macro that changes the DTMF to INBAND once you are connected to the problem number. At least in theory, if your provider is expecting RFC2833 and they get inband, they should just ignore the inband signaling and pass it on as part of the audio stream. The only problem is that this may only work if you use uLaw or aLaw for your codec and I don't know exactly how to set the tone duration without having a zapata.conf or dahdi.conf entry. Even with one of those files, I don't know how Asterisk chooses to do the rfc2833 to inband translation or where it pulls the toneduration setting from if no PSTN interface is involved in the call. -Brent John covici wrote: OK, thanks. If I could convince them to use info, would that be better as far as the duration is concerned? on Monday 04/13/2009 Brent Davidson(br...@texascountrytitle.com) wrote John covici wrote: Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. If your provider insists on rfc2833, then their servers will be responsible for setting the tone duration sent to PSTN lines. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] duration of rfc2833 generated dtmf
Its not there and the link you gave me says its for sip originating rather than calls to a sip channel. on Tuesday 04/14/2009 Brent Davidson(br...@texascountrytitle.com) wrote It's been around awhile. I've used it in 1.4 Check out this link for basic info: http://www.voip-info.org/wiki/view/Asterisk+cmd+SIPdtmfmode John covici wrote: Thanks -- can not find sip dtmf mode or sip dtmfmode in asterisk-1.4. Is this new in 1.6? on Tuesday 04/14/2009 Brent Davidson(br...@texascountrytitle.com) wrote To the best of my knowledge, the only way for you to control the duration sent to the PSTN lines is for you to be directly connected to the lines so you can set the tone duration in zapata.conf / dahdi.conf or to use inband signalling. One thing you might try is researching the SipDtmfMode command. It allows you to change the DTMF mode on an active channel. A suggestion might be to set up the dial command with the M() option that point to a Macro that changes the DTMF to INBAND once you are connected to the problem number. At least in theory, if your provider is expecting RFC2833 and they get inband, they should just ignore the inband signaling and pass it on as part of the audio stream. The only problem is that this may only work if you use uLaw or aLaw for your codec and I don't know exactly how to set the tone duration without having a zapata.conf or dahdi.conf entry. Even with one of those files, I don't know how Asterisk chooses to do the rfc2833 to inband translation or where it pulls the toneduration setting from if no PSTN interface is involved in the call. -Brent John covici wrote: OK, thanks. If I could convince them to use info, would that be better as far as the duration is concerned? on Monday 04/13/2009 Brent Davidson(br...@texascountrytitle.com) wrote John covici wrote: Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. If your provider insists on rfc2833, then their servers will be responsible for setting the tone duration sent to PSTN lines. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] duration of rfc2833 generated dtmf
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] agi no longer working with 1.4 svn 186229
If I revert back to my old version of asterisk, it works just fine. on Thursday 04/09/2009 Matt Riddell(li...@venturevoip.com) wrote On 4/04/2009 2:22 a.m., John covici wrote: The minute asterisk tries to execute an agi, it gets utils.c write error broken pipe and so hangs up the call. Anyone know what is going on? I am using kernel 2.6.27 with dahdi trunk if that makes a difference. thanks in advance for any ideas. Can you run the AGI from the Linux command line? Just press enter lots to get into it. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] agi no longer working with 1.4 svn 186229
The minute asterisk tries to execute an agi, it gets utils.c write error broken pipe and so hangs up the call. Anyone know what is going on? I am using kernel 2.6.27 with dahdi trunk if that makes a difference. 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Verizon Wireless
I have a possible suggestion -- don't consider the call answered unless someone types a 1 or something -- makes the dial plan more complex, but it should work pretty well. on Monday 03/16/2009 drew einhorn(drew.einh...@gmail.com) wrote On Mon, Mar 16, 2009 at 6:47 PM, C F shma...@gmail.com wrote: Good luck having Verizon change that. In the meantime why don't you try implementing a call screen feature so that the call is not considered answered until a key is pressed by the one answering? That way the caller will still hear ringing until the one answering presses that key. Maybe I don't understand this suggestion. I think your suggestion applys to my sip phones/atas, but they are not the problem. The problem is that when Verizon's network notices the the cell phone is currently not on their network, they pick up the call and answer with a voice error message (sometimes after only one ring), before anybody has a chance to answer on a sip device. Or, am I misunderstanding you suggestion. -- Drew Einhorn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] evaluate SIP response codes in dialplan
That is very nice, but where are the HANGUPCAUSE values documented? Thanks. on Thursday 01/15/2009 Johansson Olle E(o...@edvina.net) wrote 14 jan 2009 kl. 14.02 skrev Klaus Darilion: Hi! Is it somehow possible to evaluate the SIP response code inside the dialplan? I have an Asterisk server which forwards requests to various PSTN gateways with SIP. If the Dial() attempt is not successful I want to differ at least these 3 options: - called destination is busy (486): e.g. activate auto-redial - called destination does not exist, unassigned number (404) - gateway is broken, error, circuit busy (e.g. 503) 486 is mapped to DIALSTATUS=BUSY but both 503 and 404 is mapped to DIALSTATUS=CONGESTION As when Asterisk forwards the response with SIP to the caller the same response code is used, I suspect this information must be stored somewhere inside the channel variable. So, are there any means to access it? Check the HANGUPCAUSE, it's much more detailed than DIALSTATUS. We do map the SIP (and all other protocol errors in various channel drivers) codes to ISDN hangup causes, which gives you much more information about why a call failed. The conversion we're using follows the RFC, and where that doesn't cover it, Cisco's documentation. /Olle ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to escape DTMF?
Or just type slowly enough -- I think the timeout is half a second or so. on Friday 12/05/2008 Matthew J. Roth([EMAIL PROTECTED]) wrote Carsten Maass wrote: we are in the need to reach an external Conference-System, whos numbering system is *2*. Unfortunately *2 is the featurecode for attended transfer in our local asterisk, so the call doesn't come through. Is there a way to somehow escape the featurecode, so we can reach the external Conference? Carsten, Feature codes are configured in /etc/asterisk/features.conf. In your case, you'll want to set: [featuremap] atxfer = *2 ; Attended transfer to some other value. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call parking
OK, I can park the calls OK, but I don't get the announcement -- I am using freepbx if that makes any difference. on Wednesday 12/03/2008 Doug Lytle([EMAIL PROTECTED]) wrote Mike wrote: Can`t the parked call just go park itself (and hang up my leg of the call), and ideally call me back if not picked up within x seconds? Look at the parkcall option under the features.conf parkcall = ## ; Park call (one step parking) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call parking
Yep, those are fine and as I say, it does actually park the call because I can hang up and type 701 and get the call back, but my only problem is it hangs up immediately instead of playing the announcement. on Wednesday 12/03/2008 Eric \ManxPower\ Wieling([EMAIL PROTECTED]) wrote By legacy phone I assume you have an analog card connected to your Asterisk server. I've not used analog phones with Asterisk in many years, but IIRC you need transfer=yes and threewaycalling=yes in zapata.conf/chan_dhadi.conf. You would then do a 2nd flash to complete the transfer. On Polyom phones you do Transfer button/dial number/hear parking slot/Transfer button again/Hang up. John covici wrote: I do the following from the legacy phone: hit theflash and get a dialtone from the call, dial 70, the call is parked, but hangs up from me immediately -- isn't this an attended transfer? on Wednesday 12/03/2008 Eric \ManxPower\ Wieling([EMAIL PROTECTED]) wrote John covici wrote: OK, I can park the calls OK, but I don't get the announcement -- I am using freepbx if that makes any difference. If you park a call and do not hear the announcement then you are doing a BLIND transfer, not an ATTENDED transfer. You should be doing attended transfers for parking. If you park a call and hear the announcement and then the hold music then you did not COMPLETE the attended transfer. -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and ztdummy
Try just modprobing the module and see what happens. This worked for me when it was zaptel. on Tuesday 12/02/2008 Mike([EMAIL PROTECTED]) wrote I have no cards (nothing dahdi related). Why is my other server, built with default settings, working then? Still what do I do ? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David fire Sent: Tuesday, December 02, 2008 19:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi and ztdummy ok dont pay attention to that file for now... do you have any card on that machine? any digium card or any other brand? or not? if not the problem is that you dont need to load any module (just the dummy one) if you have any card you have a problem in the config. David 2008/12/2 Mike [EMAIL PROTECTED] Thanks Joseph. I went and read thos pages, nothing helps me. As mentionned in my other post, I don`t have a /dev/dadhi fileI don`t know why it wasn`t created or where to go from here. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph L. Casale Sent: Tuesday, December 02, 2008 18:24 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Dahdi and ztdummy I did build dahdi before building asterisk, but that`s it. No problem. But what steps did you use? Did you edit *any* dahdi related configs? See the voip-info url below. I find it hard to find any documentation referring to dadhi instead of zaptel. :) Yeah, it's not the most documented aspect of Asterisk, but there is enough for your need... I have no Digium hardware, but I still need the ztdummy timer (or whatever it`s called now). How do I get myself going? Well you need to check the README, for your application it has all you need to know: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/README Installation Note: If using `sudo` to build/install, you may need to add /sbin to your PATH. make make install Note that you'll need the utilities provided in the package dahdi-tools to configure DAHDI devices on your system. At the bottom of that file, it points you to a source for making the transition when reading older docs: http://voip-info.org/wiki/view/DAHDI I suggest you pull in dahdi-linux-complate, run #make, #make install, #make config, then #chkconfig dahdi on (or your distro equiv) and the bare configs that get installed will allow all modules to load, see that there is no hardware and fall back to dahdi_dummy. Do an lsmod and look for something like so: [EMAIL PROTECTED] ~]# lsmod | grep dahdi dahdi_dummy38984 0 dahdi 231760 9 dahdi_dummy,xpp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp crc_ccitt 35265 1 dahdi Also, [EMAIL PROTECTED] ~]# cat /proc/dahdi/1 Span 1: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: RTC) 1 (MASTER) Note that UPGRADE.txt suggests: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/UPGRADE.txt * This package no longer includes the 'menuselect' utility for choosing which modules to build; all modules that can be built are built automatically. HTH, jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi trunk does not compile with kernel 2.6.27
Hi. I am using gentoo kernel 2.6.27-r2 and dahdi trunk svn 5211 and it will not compile with this kernel whereas it does compile with 2.6.25. Here is the relevant portion of the build: VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_1_30 VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_2_30 VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_3_30 VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_4_30 HOSTCC /usr/src/dahdi-trunk/drivers/dahdi/xpp/print_fxo_modes.o HOSTLD /usr/src/dahdi-trunk/drivers/dahdi/xpp/print_fxo_modes GEN /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_fxo_modes CHECK /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_2_30 Use of uninitialized value in concatenation (.) or string at /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_2_30 line 74. make[3]: *** [/usr/src/dahdi-trunk/drivers/dahdi/xpp/init_fxo_modes.verified] Error 1 make[2]: *** [/usr/src/dahdi-trunk/drivers/dahdi/xpp] Error 2 make[1]: *** [_module_/usr/src/dahdi-trunk/drivers/dahdi] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.27-gentoo-r2' make: *** [modules] Error 2 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi trunk does not compile with kernel 2.6.27
OK, thanks.
on Tuesday 11/04/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote
On Tue, Nov 04, 2008 at 03:17:30PM -0500, John covici wrote:
Hi. I am using gentoo kernel 2.6.27-r2 and dahdi trunk svn 5211 and
it will not compile with this kernel whereas it does compile with
2.6.25. Here is the relevant portion of the build:
VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_1_30
VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_2_30
VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_3_30
VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_4_30
HOSTCC /usr/src/dahdi-trunk/drivers/dahdi/xpp/print_fxo_modes.o
HOSTLD /usr/src/dahdi-trunk/drivers/dahdi/xpp/print_fxo_modes
GEN /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_fxo_modes
CHECK /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_2_30
Use of uninitialized value in concatenation (.) or string at
/usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_2_30 line 74.
make[3]: ***
[/usr/src/dahdi-trunk/drivers/dahdi/xpp/init_fxo_modes.verified]
Error 1
make[2]: *** [/usr/src/dahdi-trunk/drivers/dahdi/xpp] Error 2
make[1]: *** [_module_/usr/src/dahdi-trunk/drivers/dahdi] Error 2
make[1]: Leaving directory `/usr/src/linux-2.6.27-gentoo-r2'
make: *** [modules] Error 2
Any assistance on this would be appreciated.
A known issue: http://bugs.digium.com/13832
'Use of uninitialized value $ENV{XBUS_NAME}' if stderr from make
redirected to a file
I hope to get it fixed. In the mean time: patch out that test from
drivers/dahdi/xpp/Kbuild
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Extension registration
=citecontext=internbr/blockquote blockquote type=citecanreinvite=yesbr/blockquote blockquote type=citedtmfmode=rfc2833br/blockquote blockquote type=citenbsp;br/blockquote blockquote type=citebr/blockquote/div/blockquotebr/div/div divbr/div div divFred Posner/div divba href=mailto:[EMAIL PROTECTED] target=_blank[EMAIL PROTECTED]/a/b/div div divbr/div divUsing VoIP? nbsp;/div div style=WORD-WRAP: break-wordSIP:span style=WHITE-SPACE: pre /spana href=mailto:[EMAIL PROTECTED] target=_blank[EMAIL PROTECTED]/a/div/div/div/divbr___br -- Bandwidth and Colocation Provided by a href=http://www.api-digital.com/; target=_blankhttp://www.api-digital.com/a --brbrAstriCon 2008 - September 22 - 25 Phoenix, ArizonabrRegister Now: a href=http://www.astricon.net/; target=_blankhttp://www.astricon.net/abr brasterisk-users mailing listbrTo UNSUBSCRIBE or update options visit:brnbsp; a href=http://lists.digium.com/mailman/listinfo/asterisk-users; target=_blankhttp://lists.digium.com/mailman/listinfo/asterisk-users/abr /blockquote/divbr/div ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] echo cancellation problem with dahdi
;channel = 16 ;signalling=em_w ; ; All those in group 0 I'll use for outgoing calls ; ; Strip most significant digit (9) before sending ; ;stripmsd=1 ;callerid=asreceived ;group=0 ;signalling=fxs_ls ;channel = 45 ;signalling=fxo_ls ;group=1 ;callerid=Joe Schmoe (256) 428-6131 ;channel = 25 ;callerid=Megan May (256) 428-6132 ;channel = 26 ;callerid=Suzy Queue (256) 428-6233 ;channel = 27 ;callerid=Larry Moe (256) 428-6234 ;channel = 28 ; ; Sample PRI (CPE) config: Specify the switchtype, the signalling as ; either pri_cpe or pri_net for CPE or Network termination, and generally ; you will want to create a single group for all channels of the PRI. ; ; switchtype = national ; signalling = pri_cpe ; group = 2 ; channel = 1-23 ; ; Used for distintive ring support for x100p. ; You can see the dringX patterns is to set any one of the dringXcontext fields ; and they will be printed on the console when an inbound call comes in. ; ;dring1=95,0,0 ;dring1context=internal1 ;dring2=325,95,0 ;dring2context=internal2 ; If no pattern is matched here is where we go. ;context=default ;channel = 1 ;signalling=fxs_ks ;channel=1 ;of course I place my incoming callers in a different context from local users ;context=home ;and reverse the signalling ;signalling=fxo_ks ;channel=2 [channels] language=en rxwink=300 ; Atlas seems to use long (250ms) winks ; XTDM20B Port #2 plugged into PSTN 1 to phone ; #include zapata_additional.conf context=from-internal signalling=fxo_ks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes immediate=no echocancel=256 echocancelwhenbridged=yes echotraining=400 group=0 relaxdtmf=yes cidsignalling=bell mailbox=200 channel=1 context=from-pstn signalling=fxs_ks faxdetect=incoming rxgain=4.0 callwaiting=yes channel=4 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi vs zap with latest version of asterisk -- having some problems
Hi. I am using asterisk 1.4 branch svn from yesterday and I am having some problems -- I am still using the zaptel drivers temporarily. Now there are two problems -- the first minor in that my asterisk would not work at all when I first tried it -- or at least not on my X400p card chan_dahdi would not load. there was still a chan_zap.so in my source directory so I copied it over and at least got things going. However conferences do not work at all, unable to open pseudo device. So, how can I get chan_dahdi to recognize my zaptel drivers? I already have dahdichanname = no in my options section of asterisk.conf. 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi vs zap with latest version of asterisk -- having some problems
I was still using the zaptel kernel drivers -- this is what I would like to do for now. on Thursday 09/11/2008 Matt Gibson([EMAIL PROTECTED]) wrote Did you setup the new /etc/dadhi/system.conf as well as unloading your old zaptel modules and re-inserting the new dahdi modules? * The primary kernel modules have changed names; the new names are: zaptel.ko -dahdi.ko ztd-eth.ko - dahdi_dynamic_eth.ko ztd-loc.ko - dahdi_dynamic_loc.ko ztdummy.ko - dahdi_dummy.ko ztdynamic.ko - dahdi_dynamic.ko zttranscode.ko - dahdi_transcode.ko Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: Thursday, September 11, 2008 11:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi vs zap with latest version of asterisk -- having some problems Hi. I am using asterisk 1.4 branch svn from yesterday and I am having some problems -- I am still using the zaptel drivers temporarily. Now there are two problems -- the first minor in that my asterisk would not work at all when I first tried it -- or at least not on my X400p card chan_dahdi would not load. there was still a chan_zap.so in my source directory so I copied it over and at least got things going. However conferences do not work at all, unable to open pseudo device. So, how can I get chan_dahdi to recognize my zaptel drivers? I already have dahdichanname = no in my options section of asterisk.conf. 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] error while trying to compile dahdi-tools-trunk
Hi. I am getting the following error while trying to compile the dahdi-tools-trunk from svn this morning. gcc -g -O2 -I. -O2 -g -fPIC -Wall -DBUILDING_TONEZONE -MD -MT sethdlc.o -MF .sethdlc.o.d -MP -c -o sethdlc.o sethdlc.c sethdlc.c: In function 'set_iface': sethdlc.c:205: error: 'union anonymous' has no member named 'ifru_settings' I previously installed dahdi-trunk before trying to compile this package, but no joy. 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error while trying to compile dahdi-tools-trunk
on Sunday 09/07/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote On Sun, Sep 07, 2008 at 09:22:57AM -0400, John covici wrote: Hi. I am getting the following error while trying to compile the dahdi-tools-trunk from svn this morning. gcc -g -O2 -I. -O2 -g -fPIC -Wall -DBUILDING_TONEZONE -MD -MT sethdlc.o -MF .sethdlc.o.d -MP -c -o sethdlc.o sethdlc.c sethdlc.c: In function 'set_iface': sethdlc.c:205: error: 'union anonymous' has no member named 'ifru_settings' Not reproduced here. struct ifreq is defined in /usr/include/linux/if.h Maybe they moved something -- here is my if.h from linux-headers-2.6.26. /* * INET An implementation of the TCP/IP protocol suite for the LINUX * operating system. INET is implemented using the BSD Socket * interface as the means of communication with the user level. * * Global definitions for the INET interface module. * * Version: @(#)if.h1.0.2 04/18/93 * * Authors: Original taken from Berkeley UNIX 4.3, (c) UCB 1982-1988 * Ross Biro * Fred N. van Kempen, [EMAIL PROTECTED] * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License * as published by the Free Software Foundation; either version * 2 of the License, or (at your option) any later version. */ #ifndef _LINUX_IF_H #define _LINUX_IF_H #include net/if.h #include linux/types.h/* for __kernel_caddr_t et al */ #include linux/socket.h /* for struct sockaddr et al */ #define IFNAMSIZ16 #include linux/hdlc/ioctl.h /* Standard interface flags (netdevice-flags). */ #define IFF_UP 0x1 /* interface is up */ #define IFF_BROADCAST 0x2 /* broadcast address valid */ #define IFF_DEBUG 0x4 /* turn on debugging*/ #define IFF_LOOPBACK0x8 /* is a loopback net*/ #define IFF_POINTOPOINT 0x10/* interface is has p-p link*/ #define IFF_NOTRAILERS 0x20/* avoid use of trailers*/ #define IFF_RUNNING 0x40/* interface RFC2863 OPER_UP*/ #define IFF_NOARP 0x80/* no ARP protocol */ #define IFF_PROMISC 0x100 /* receive all packets */ #define IFF_ALLMULTI0x200 /* receive all multicast packets*/ #define IFF_MASTER 0x400 /* master of a load balancer*/ #define IFF_SLAVE 0x800 /* slave of a load balancer */ #define IFF_MULTICAST 0x1000 /* Supports multicast */ #define IFF_PORTSEL 0x2000 /* can set media type */ #define IFF_AUTOMEDIA 0x4000 /* auto media select active */ #define IFF_DYNAMIC 0x8000 /* dialup device with changing addresses*/ #define IFF_LOWER_UP0x1 /* driver signals L1 up */ #define IFF_DORMANT 0x2 /* driver signals dormant */ #define IFF_ECHO0x4 /* echo sent packets*/ #define IFF_VOLATILE(IFF_LOOPBACK|IFF_POINTOPOINT|IFF_BROADCAST|IFF_ECHO|\ IFF_MASTER|IFF_SLAVE|IFF_RUNNING|IFF_LOWER_UP|IFF_DORMANT) /* Private (from user) interface flags (netdevice-priv_flags). */ #define IFF_802_1Q_VLAN 0x1 /* 802.1Q VLAN device. */ #define IFF_EBRIDGE 0x2 /* Ethernet bridging device.*/ #define IFF_SLAVE_INACTIVE 0x4 /* bonding slave not the curr. active */ #define IFF_MASTER_8023AD 0x8 /* bonding master, 802.3ad. */ #define IFF_MASTER_ALB 0x10/* bonding master, balance-alb. */ #define IFF_BONDING 0x20/* bonding master or slave */ #define IFF_SLAVE_NEEDARP 0x40 /* need ARPs for validation */ #define IFF_ISATAP 0x80/* ISATAP interface (RFC4214) */ #define IF_GET_IFACE0x0001 /* for querying only */ #define IF_GET_PROTO0x0002 /* For definitions see hdlc.h */ #define IF_IFACE_V350x1000 /* V.35 serial interface*/ #define IF_IFACE_V240x1001 /* V.24 serial interface*/ #define IF_IFACE_X210x1002 /* X.21 serial interface*/ #define IF_IFACE_T1 0x1003 /* T1 telco serial interface*/ #define IF_IFACE_E1 0x1004 /* E1 telco serial interface*/ #define IF_IFACE_SYNC_SERIAL 0x1005 /* can't be set by software */ #define IF_IFACE_X21D 0x1006 /* X.21 Dual Clocking (FarSite) */ /* For definitions see hdlc.h */ #define IF_PROTO_HDLC 0x2000 /* raw HDLC protocol*/ #define IF_PROTO_PPP0x2001 /* PPP protocol */ #define IF_PROTO_CISCO 0x2002 /* Cisco HDLC protocol */ #define IF_PROTO_FR 0x2003
[asterisk-users] svn branches for dhadi and its tools
Hi. I want to use the new asterisk 1.4 with dahdi, but I would like to know the svn branches for the dahdi, so I can use them that way -- much easier to keep up with bug fixes, etc. Thanks. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] svn branches for dhadi and its tools
OK, thanks. on Friday 09/05/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote On Fri, Sep 05, 2008 at 10:32:30AM -0400, John covici wrote: Hi. I want to use the new asterisk 1.4 with dahdi, but I would like to know the svn branches for the dahdi, so I can use them that way -- much easier to keep up with bug fixes, etc. trunk, in both cases. http://svn.digium.com/svn/dahdi/linux/trunk http://svn.digium.com/svn/dahdi/tools/trunk -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme replacement with native 729 support
OK, I guess I need to show my ignorance -- what is the difference between ulaw and signed linear? on Tuesday 07/15/2008 Tilghman Lesher([EMAIL PROTECTED]) wrote On Tuesday 15 July 2008 13:32:12 Artie Gold wrote: Does anyone know of a replacement for meetme that provides native G729 support? The transcoding back and forth from/to 711 is eating too much processor for what we're doing... Buy a hardware transcoder board. There is simply no way to mix compressed audio like that without decompressing first. And by the way, it's decompressing to signed linear 16-bit audio, not ulaw. Even mixing of ulaw requires a decompress to signed linear. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] strange iax authentication behavior
Hi. I have a number of sections in my iax configurationwhich was generated by freepbx like this: [covici] username=covici type=friend secret=password host=dynamic disallow=all context=from-internal allow=ulaw [Carol 0326] host=iax.binfone.com username=3234740326 secret=another password type=friend context=from-pstn Now with this configuration, if I receive a call over the trunk covici, it complains that the host sending the call is trying to authenticate as Carol 0326 . If I change the second trunk to type=peer it works, or if I put the covici trunk at the end it also works -- anyone know why this is happening because of course the config gets regenerated all the time and this should not work this way. Thanks in advance for any assistance. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SayNumber while reading DTMF?
Well, how about using an app like cepstral to record it as a wav and using background or waitexten to play the wav -- the time lag should never be noticed. on Tuesday 06/10/2008 Russell Bryant([EMAIL PROTECTED]) wrote Douglas Garstang wrote: I'm using the SayNumber() app to read out a users balance for an IVR. Is there a way I can do that while waiting for DTMF input? Obviously, read() and Background() don't correctly say a number in number format. I do not know of a way to do that. It would be an extremely useful new feature to have, but as fair as I know, is not currently available. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Limits
12 people is nothing -- I do 20 regularly -- however you may want to have them come in as muted or tell them to mute themselves, because the latency can cause very severe echoes if they are on a speaker phone or cell phone. on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote Actually I think they will all be calling in using regular pstn phones and cell phones. Sam Al Baker wrote: The 2 big questions are: -Are all participants using QoS end to end ? -Are all of them using the SAME CODEC. As the amount of Transcoding goes up, the work on the * box goes up and can be a problem. Sam wrote: I am thinking about using my asterisk server to host a conference with about 12 other people from around the USA. Bandwidth issues aside, will this work or will all the different latencies cause issues? Yea I know, I could just try it and find out but it is going to take alot of time to get everyones schedule to line up, I don't want to go through the trouble if I will just be disappointed. Thanks, Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Card: Power Connector, from SATA to NORMAL
How can you tell if a server will have enough power for such a card? Most servers have power to spare. on Sunday 05/04/2008 Jay R. Ashworth([EMAIL PROTECTED]) wrote On Sat, May 03, 2008 at 01:42:22PM +1000, Rob Hillis wrote: Actually I can see exactly where this may be useful. I can think of at least one customer off the top of my head who has insisted on having a TDM2400 installed in a Dell server that we know can't provide enough power to the card where it's being used as a collection of FXS ports. Of course we've advised them of the limitation, but they completely ignored us. :) Something like this would have allowed us to provide a solution to a problem that is likely to come back and bite us in the future. Advise the customer in writing; make them sign it. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor not merging calls
Newer version of sox don't seem to have soxmix anymore, but you can use sox -m and I think asterisk should be changed to use that instead. on Monday 04/21/2008 Jared Smith([EMAIL PROTECTED]) wrote On Mon, 2008-04-21 at 21:11 +0530, Sanjay Rajdev wrote: One of the box that have Asterisk 1.4.18 is properly merging calls and the other box that has Asterisk 1.4.15 is recording the calls but not merging them, I have made sure that SOX is installed on the box. It might be worth giving the MixMonitor() application a try instead. :-) -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor not merging calls
I changed the codeof the monitor app to use sox -m instead of soxmix
which I no longer have. Mixmonitor would work as well, but the
one-touch recording was set to the other, so I am using that.
on Monday 04/21/2008 Sanjay Rajdev([EMAIL PROTECTED]) wrote
John,
Is their something that I can change on my side to get this working ?
Jared,
I thought MixMonitor() was for Queue, Can you let me know how to use it?
Thanking you for replying.
Regards,
Sanjay Rajdev
- Original Message -
From: John covici [EMAIL PROTECTED]
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Sent: Monday, April 21, 2008 9:39:39 PM GMT +05:30 Chennai, Kolkata, Mumbai,
New Delhi
Subject: Re: [asterisk-users] Monitor not merging calls
Newer version of sox don't seem to have soxmix anymore, but you can
use sox -m and I think asterisk should be changed to use that instead.
on Monday 04/21/2008 Jared Smith([EMAIL PROTECTED]) wrote
On Mon, 2008-04-21 at 21:11 +0530, Sanjay Rajdev wrote:
One of the box that have Asterisk 1.4.18 is properly merging calls and
the other box that has Asterisk 1.4.15 is recording the calls but not
merging them, I have made sure that SOX is installed on the box.
It might be worth giving the MixMonitor() application a try instead. :-)
--
Jared Smith
Community Relations Manager
Digium, Inc.
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--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?
John Covici
[EMAIL PROTECTED]
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htmlheadstyle type='text/css'p { margin: 0; }/stylestyle
type='text/css'body { font-family: 'Times New Roman'; font-size: 12pt;
color: #00}/style/headbodyJohn,brIs their something that I can
change on my side to get this working?brbrJared,brI thought
MixMonitor() was for Queue, Can you let me know how to use
it?brbrThanking you for replying.brbrRegards,brSanjay
Rajdevbrbr- Original Message -brFrom: John covici lt;[EMAIL
PROTECTED]gt;brTo: [EMAIL PROTECTED], Asterisk Users Mailing List -
Non-Commercial Discussion lt;asterisk-users@lists.digium.comgt;brSent:
Monday, April 21, 2008 9:39:39 PM GMT +05:30 Chennai, Kolkata, Mumbai, New
DelhibrSubject: Re: [asterisk-users] Monitor not merging
callsbrbrNewer version of sox don't seem to have soxmix anymore, but you
canbruse sox -m and I think asterisk should be changed to use that
instead.brbron Monday 04/21/2008 Jared Smith([EMAIL PROTECTED])
wrotebrnbsp;gt; On Mon, 2008-04-21 at 21:11 +0530, Sanjay Rajdev
wrote:brnbsp;gt; gt; One of the box that have Asterisk 1.4.18 is
properly merging calls andbrnbsp;gt; gt; the other box that has
Asterisk 1.4.15 is recording the calls but notbrnbsp;gt; gt; merging
them, I have made sure that SOX is installed on the box. brnbsp;gt;
brnbsp;gt; It might be worth giving the MixMonitor() application a try
instead. :-)brnbsp;gt; brnbsp;gt; brnbsp;gt; -- brnbsp;gt;
Jared Smithbrnbsp;gt; Community Relations Managerbrnbsp;gt; Digium,
Inc.brnbsp;gt; brnbsp;gt; brnbsp;gt;
___brnbsp;gt; -- Bandwidth
and Colocation Provided by http://www.api-digital.com --brnbsp;gt;
brnbsp;gt; asterisk-users mailing listbrnbsp;gt; To UNSUBSCRIBE or
update options visit:brnbsp;gt; nbsp;
nbsp;http://lists.digium.com/mailman/listinfo/asterisk-usersbrbr--
brYour life is like a penny. nbsp;You're going to lose it. nbsp;The
question is:brHow dobryou spend it?brbrnbsp;nbsp; nbsp; nbsp;
nbsp; John Covicibrnbsp;nbsp; nbsp; nbsp; nbsp; [EMAIL
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[asterisk-users] Asterisk and the Mitel SX 200 integration
Hi. One of my clients has an old Mitel SX 200 with a separate computer doing the voicemail and auto attendant and integrated via a COV card which is in his case an ISA card! We would all like to migrate to asterisk, but as a first step, can asterisk integrate into the Mitel, so it can serve as auto attendant and the voicemail for the extensions? If this is successful we could gradually migrate extensions, particularly if we could get the Mitel to talk to asterisk via one of its t1 cards. Any assistance or experience along these lines 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and the Mitel SX 200 integration
Yep, you guessed it, an activvoice system. Anyway to make Asterisk act like that for a while? Thanks. on Friday 04/11/2008 Doug([EMAIL PROTECTED]) wrote At 17:32 4/11/2008, John covici wrote: Hi. One of my clients has an old Mitel SX 200 with a separate computer doing the voicemail and auto attendant and integrated via a COV card which is in his case an ISA card! Is it an ActiveVoice system? We would all like to migrate to asterisk, but as a first step, can asterisk integrate into the Mitel, so it can serve as auto attendant and the voicemail for the extensions? We've got a client with the exact same setup. They have suffered long enough with this dinosaur. They are in the process of going with an all-Asterisk system. You would probably make more money trying an intermediate step using the SX-200 and Asterisk, but it would be obviously more costly for them as well as prolong their misery. It's your call, but I would recommend getting away from 30 year old technology as fast as you can run. The ActiveVoice system is a cantankerous 20 year old system in itself. You have just received 2 cents worth of advice for FREE! If this is successful we could gradually migrate extensions, particularly if we could get the Mitel to talk to asterisk via one of its t1 cards. Any assistance or experience along these lines 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and the Mitel SX 200 integration
OK, this is exactly what I would like to do, can you either write me on or off list for further details. This would be the first baby step toward the 20th Century!! on Friday 04/11/2008 Alexander Lopez([EMAIL PROTECTED]) wrote Jorge is correct you will not get the information need via FXO/FXS unless you program the Mitel to do DTMF inband. It is possible but a cludge of a fix at best. We have successfully integrated several Mitel SX200 and SX2000 switches via the PRI (preferred) or T1 using EM_Wink (works but you have delays while waiting for the winks. (wink, wink :-) ). The Mitel is rock-solid and depending on the size of the install a fork-lift replacement may not be desirable. I would start by replacing the VM (ActiveVoice) with and Asterisk box, you can give them unified messaging as well as a stable and current platform ( I have seen the Octel COV card catch on fire!!) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jorge Mendoza Sent: Friday, April 11, 2008 8:32 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Cc: Doug Subject: Re: [asterisk-users] Asterisk and the Mitel SX 200 integration I second Doug advice. Migrate to Asterisk asap. We have several Asterisk auto attendant integrated with Mitel, even billing using the Mitel's smdr. But voicemail is different. The COV card emulate a SS4 phone and receive information needed for a voice mail system. With FXO/FXS ports is not possible receive such information. Jorge Mendoza John covici wrote: Yep, you guessed it, an activvoice system. Anyway to make Asterisk act like that for a while? Thanks. on Friday 04/11/2008 Doug([EMAIL PROTECTED]) wrote At 17:32 4/11/2008, John covici wrote: Hi. One of my clients has an old Mitel SX 200 with a separate computer doing the voicemail and auto attendant and integrated via a COV card which is in his case an ISA card! Is it an ActiveVoice system? We would all like to migrate to asterisk, but as a first step, can asterisk integrate into the Mitel, so it can serve as auto attendant and the voicemail for the extensions? We've got a client with the exact same setup. They have suffered long enough with this dinosaur. They are in the process of going with an all-Asterisk system. You would probably make more money trying an intermediate step using the SX-200 and Asterisk, but it would be obviously more costly for them as well as prolong their misery. It's your call, but I would recommend getting away from 30 year old technology as fast as you can run. The ActiveVoice system is a cantankerous 20 year old system in itself. You have just received 2 cents worth of advice for FREE! If this is successful we could gradually migrate extensions, particularly if we could get the Mitel to talk to asterisk via one of its t1 cards. Any assistance or experience along these lines 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] testing please ignore
If I see this, then messages are getting through. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Telemarketer Torture....
I have simply gotten rid of most of the telemarketer calls by simply making them press 1 or wait to talk to me and they give up almost always. on Sunday 03/16/2008 Horwich IT Services (Godwin Stewart)([EMAIL PROTECTED]) wrote On Sun, 16 Mar 2008 08:50:50 -0500, Lyle Giese [EMAIL PROTECTED] wrote: I just forward them to one of those two extensions. If callerid worked more reliably I would automate it. But I get a lot of caller id failures on my incoming POTS lines, esp when calling in from my cell phone. The way I worked around this problem was to give a passcode to people I want to hear from even if they conceal CLI. If an inbound call comes in without CLI (or with CLI but the number is in my blocklist for that matter), I forward it to a recorded message saying Caller ID screening is in operation. Please press 1 if you are an authorized caller. When the user complies, they're prompted for the passcode. If it's correct, then the call is forwarded to my extension. Those I do want to hear from are not just blown off, they have a chance to get through to me regardless of the screening. Teleslime doesn't, and they've paid for the call anyway. -- Godwin Stewart - Horwich IT services ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P dialout problem
Hi. Since I am stuck with kernel 2.6.24, is there any way to compile zaptel 1.4.7.1 under kernel 2.6.24? I tried using make KBUILD_NOPEDANTIC=1 -- however this does not compile. Any other suggestions for this and can I still use the latest version of asterisk if I do this successfully? Thanks. on Monday 02/25/2008 sean darcy([EMAIL PROTECTED]) wrote On Mon, Feb 25, 2008 at 3:42 AM, Anthony Messina [EMAIL PROTECTED] wrote: Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: -- Starting simple switch on 'Zap/1-1' -- Executing [EMAIL PROTECTED]:1] Dial(Zap/1-1, Zap/3/8801234) in new stack [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing '8801234' [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:2030 zt_call: Deferring dialing... -- Called 3/8801234 [Feb 25 02:37:00] WARNING[7194]: chan_zap.c:3835 zt_handle_event: Detected alarm on channel 3: No Alarm -- Hungup 'Zap/3-1' == Everyone is busy/congested at this time (1:0/0/1) [Feb 25 02:37:00] NOTICE[7082]: chan_zap.c:6678 handle_init_event: Alarm cleared on channel 3 So the call fails and if I weren't using a test extension: exten = 2111,1,Dial(Zap/3/8801234) it would proceed in the dialplan. asterisk]# cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 (MASTER) 1 WCTDM/0/0 FXOKS (In use) 2 WCTDM/0/1 3 WCTDM/0/2 FXSKS (In use) 4 WCTDM/0/3 Where do I go with this? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Look at http://bugs.digium.com/view.php?id=11855. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P dialout problem
I am getting this strange error: make[1]: Entering directory `/usr/src/zaptel-1.4.7.1' make -C /lib/modules/2.6.24-gentoo-r2/build SUBDIRS=/usr/src/zaptel-1.4.7.1 HOTPLUG_FIRMWARE=yes modules make[2]: Entering directory `/usr/src/linux-2.6.24-gentoo-r2' CC [M] /usr/src/zaptel-1.4.7.1/wcfxo.o /usr/src/zaptel-1.4.7.1/wcfxo.c:38:27: error: zaptel/zaptel.h: No such file or directory But the file is there. At least its in the source directory. I did the ./configure and the make menuselect to eliminate some unnecessary modules. on Tuesday 02/26/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote On Tue, Feb 26, 2008 at 06:45:15AM -0500, John covici wrote: Hi. Since I am stuck with kernel 2.6.24, is there any way to compile zaptel 1.4.7.1 under kernel 2.6.24? I tried using make KBUILD_NOPEDANTIC=1 -- however this does not compile. Any other suggestions for this and can I still use the latest version of asterisk if I do this successfully? What error(s) do you get? Later on I fixed a number of build problems with ztd-eth.c . But you can probably skip that module altogether if you don't need TDM over Ethernet. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P dialout problem
I do have the newer one installed -- is this the problem? And why this weird error anyway? on Tuesday 02/26/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote On Tue, Feb 26, 2008 at 07:50:09AM -0500, John covici wrote: I am getting this strange error: make[1]: Entering directory `/usr/src/zaptel-1.4.7.1' make -C /lib/modules/2.6.24-gentoo-r2/build SUBDIRS=/usr/src/zaptel-1.4.7.1 HOTPLUG_FIRMWARE=yes modules make[2]: Entering directory `/usr/src/linux-2.6.24-gentoo-r2' CC [M] /usr/src/zaptel-1.4.7.1/wcfxo.o /usr/src/zaptel-1.4.7.1/wcfxo.c:38:27: error: zaptel/zaptel.h: No such file or directory But the file is there. At least its in the source directory. I did the ./configure and the make menuselect to eliminate some unnecessary modules. -DSTANDALONE_ZAPATA not getting through to the EXTRA_CFLAGS? Note: luckily you didn't have zaptel installed and this it didn't use a different version of zaptel.h from /usr/include/zaptel/zaptel.h . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P dialout problem
OK, here is the problem -- how do I compile 1.4.7.1 using kernel 2.6.24 -- when I try the make KBUILD_NOPEDANTIC=1 I get the no such file or directory that I already mentioned -- when I take out that KBUILD option, I get what I got before -- the error from the kernel module build about the CFLAGS being changed in the Makefile. So how can I compile this thing till there is a fix for the bug or the regression is removed? on Tuesday 02/26/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote Slightly edited your message: On Tue, Feb 26, 2008 at 08:48:13AM -0500, John covici wrote: on Tuesday 02/26/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote On Tue, Feb 26, 2008 at 07:50:09AM -0500, John covici wrote: I am getting this strange error: make[1]: Entering directory `/usr/src/zaptel-1.4.7.1' make -C /lib/modules/2.6.24-gentoo-r2/build SUBDIRS=/usr/src/zaptel-1.4.7.1 HOTPLUG_FIRMWARE=yes modules make[2]: Entering directory `/usr/src/linux-2.6.24-gentoo-r2' CC [M] /usr/src/zaptel-1.4.7.1/wcfxo.o /usr/src/zaptel-1.4.7.1/wcfxo.c:38:27: error: zaptel/zaptel.h: No such file or directory But the file is there. At least its in the source directory. I did the ./configure and the make menuselect to eliminate some unnecessary modules. -DSTANDALONE_ZAPATA not getting through to the EXTRA_CFLAGS? And why this weird error anyway? Because STANDALONE_ZAPATA was not #define-d . Note: luckily you didn't have zaptel installed and this it didn't use a different version of zaptel.h from /usr/include/zaptel/zaptel.h . I do have the newer one installed -- is this the problem? Hmmm I guess /ysr/include is not in the include path for kernel modules building. A good thing in this case. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Had it with Dell Garbage
I had a server built for me by J and N Computer Services http://www.jncs.com which is using a Super Micro c2sbe MB which I think has what you need plus 4 PCI-32 slots! Its a nice MB and I have an e8400 cpu in it. on Tuesday 02/26/2008 Matt([EMAIL PROTECTED]) wrote I've had it with Dell server garbage.They seem to change RAID controllers as much as I change socks, and then the controllers don't work with Linux, unless you load a new driver.They sell servers with a PCI-e slot in them, but then you get it and find out the RAID controller is using the PCI-e slot! Their sales folks are dumber than rocks, and they change them more often than I change underwear. [end rant]. Can anyone recommend an IBM or Gateway server that you have used with Asterisk and are happy with, and which will support RAID-1 or RAID-5 and has room for one or two PCI-express interface cards? I#39;ve had it with Dell server garbage.nbsp;nbsp;nbsp; They seem to change RAID controllers as much as I change socks, and then the controllers don#39;t work with Linux, unless you load a new driver.nbsp;nbsp;nbsp; They sell servers with a PCI-e slot in them, but then you get it and find out the RAID controller is using the PCI-e slot!nbsp;nbsp; Their sales folks are dumber than rocks, and they change them more often than I change underwear.br [end rant].brbrCan anyone recommend an IBM or Gateway server that you have used with Asterisk and are happy with, and which will support RAID-1 or RAID-5 and has room for one or two PCI-express interface cards?br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cannot dial out with latest zaptel and kernel 2.6.24
I updated zaptel and I can dial out, but when someone calls in it won't hangup unless my extension hangs up, which was not true before. This is a better state than before, thanks much for fixing so far. on Tuesday 02/26/2008 Shaun Ruffell([EMAIL PROTECTED]) wrote John Covici wrote: Hi. I am using asterisk 1.4 (latest as of today) and zaptel 1.4 (latest as of today) and I cannot dial out using my 400P card with one fxs module and one fxo module. I am using kernel 2.6.24 and get the following log entries: [Feb 25 17:28:13] VERBOSE[25071] logger.c: -- Executing [EMAIL PROTECTED]:23] Dial(Zap/1-1, ZAP/4/www411|300|wW) in new stack [Feb 25 17:28:13] DEBUG[25071] chan_zap.c: Dialing 'www411' [Feb 25 17:28:13] DEBUG[25071] chan_zap.c: Deferring dialing... [Feb 25 17:28:13] VERBOSE[25071] logger.c: -- Called 4/www411 [Feb 25 17:28:14] WARNING[25071] chan_zap.c: Detected alarm on channel 4: No Alarm [Feb 25 17:28:14] VERBOSE[25071] logger.c: -- Hungup 'Zap/4-1' [Feb 25 17:28:14] VERBOSE[25071] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Any assistance on this would be appreciated. It looks like this might have been a combination of zaptel generating battery alarms which asterisk 1.4 didn't recognize. Could you try updating just zaptel and see if you still see the alarm? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cannot dial out with latest zaptel and kernel 2.6.24
Hi. I am using asterisk 1.4 (latest as of today) and zaptel 1.4 (latest as of today) and I cannot dial out using my 400P card with one fxs module and one fxo module. I am using kernel 2.6.24 and get the following log entries: [Feb 25 17:28:13] VERBOSE[25071] logger.c: -- Executing [EMAIL PROTECTED]:23] Dial(Zap/1-1, ZAP/4/www411|300|wW) in new stack [Feb 25 17:28:13] DEBUG[25071] chan_zap.c: Dialing 'www411' [Feb 25 17:28:13] DEBUG[25071] chan_zap.c: Deferring dialing... [Feb 25 17:28:13] VERBOSE[25071] logger.c: -- Called 4/www411 [Feb 25 17:28:14] WARNING[25071] chan_zap.c: Detected alarm on channel 4: No Alarm [Feb 25 17:28:14] VERBOSE[25071] logger.c: -- Hungup 'Zap/4-1' [Feb 25 17:28:14] VERBOSE[25071] logger.c: == Everyone is busy/congested at this time (1:0/0/1) 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cannot dial out with latest zaptel and kernel 2.6.24
How would I use that with kernel 2.6.24? on Monday 02/25/2008 Shaun Ruffell([EMAIL PROTECTED]) wrote Hi John, John Covici wrote: Hi. I am using asterisk 1.4 (latest as of today) and zaptel 1.4 (latest as of today) and I cannot dial out using my 400P card with one fxs module and one fxo module. It looks like you might be hitting a regression with DTMF tone generation in the latest zaptel releases. The running commentary on this issue is at: http://bugs.digium.com/view.php?id=11855 Could you try using zaptel version 1.4.7.1 and see if that helps? Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SPA3000 -- PSTN to VoIP
What I did was to change the dial plan to ring exttension at the asterisk ip address -- which under freepbx simulates an incoming call -- this is the only way I have ever gotten that function to work properly. What does not work for me is voip-to-pstn -- I get a 403 response from the spa3102. on Friday 01/25/2008 Rudolf Ladyzhenskii([EMAIL PROTECTED]) wrote Hi, all I am trying to figure out how to forward incoming PSTN call on SPA3000 to VoIP extension(s). Basically, I have converted my home to VoIP. I have normal phone (connected to SPA3000) and couple of IP phones. All call coming from VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I need to do same thing for incoming PSTN calls. I have enabled gateway function in SPA3000 and configured PSTN as a VoIP extension in asterisk, but on incoming PSTN call, I do not see anything on asterisk console. Can someone point me into the right direction? Thanks, Rudolf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme recording
Just set the variable ${MEETME_RECORDINGFORMAT} to the desired format
and voila its done.
Have fun.
on Tuesday 01/15/2008 Lees, James (UK)([EMAIL PROTECTED]) wrote
Hello,
Is there a way to change the format from the default?
'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format
${MEETME_RECORDINGFORMAT}). Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. -
requires chan_zap.so
Many thanks
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How do
you spend it?
John Covici
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[asterisk-users] is Power fail transfer possible with asterisk?
Hi. I have a client who wants some way that his analog phones can call out even after the power is out and the UPS has died -- some way that a phone can connect directly to an fxo or some such when power is gone. Any hardware around which can do this? I have heard of some ATA's which do this, do any of the channel banks have this capability? Thanks. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is Power fail transfer possible with asterisk?
OK, to clarify a bit, he wants to fix things so that all we are depending on are the pots lines -- I know if they go out you are gone. So what can we do in that case? on Wednesday 01/02/2008 Tilghman Lesher([EMAIL PROTECTED]) wrote On Wednesday 02 January 2008 17:10:05 John covici wrote: Hi. I have a client who wants some way that his analog phones can call out even after the power is out and the UPS has died -- some way that a phone can connect directly to an fxo or some such when power is gone. Any hardware around which can do this? I have heard of some ATA's which do this, do any of the channel banks have this capability? 1) If his phones are this critical, he needs a triple redundant generator. 2) Ask him what he would like to do after 36 hours of power outage, when even the telco stops being able to provide battery on their POTS lines. If your provider is out, there's very little you can do. Perhaps a ham radio attached to a car battery? Speed costs; how fast would you like to go? -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No cdr_csv after upgrade from 1.2.x to 1.4.x
I believe you will find your new Master.csv in cdr-custom if memory serves. on Thursday 12/27/2007 Godson Gera([EMAIL PROTECTED]) wrote On Dec 26, 2007 11:36 PM, Don Pobanz [EMAIL PROTECTED] wrote: After upgrading from 1.2.x to 1.4.x call detail records are not being written to /var/log/asterisk/cdr-csv/Master.csv In cdr_manager.conf I have [general] Enabled = yes Apparently there is something else that needs to be configured for call detail records in 1.4.x. Can someone point me in the right direction? cdr_manager.conf is for sending CDR events over AMI (Asterisk Manager Interface). The file you need to edit is cdr.conf make sure you have enable = yes in its [general] section default is yes and you can define and custom setting for csv engine in [csv] section of same cdr.conf file and also you can do some settings in cdr_custom.conf file. Look at sample files that come with asterisk for examples. -- Godson Gera, http://godson.in brbrdiv class=gmail_quoteOn Dec 26, 2007 11:36 PM, Don Pobanz lt;a href=mailto:[EMAIL PROTECTED][EMAIL PROTECTED]/agt; wrote:brblockquote class=gmail_quote style=border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex; After upgrading from 1.2.x to 1.4.x call detail records are not beingbrwritten to /var/log/asterisk/cdr-csv/Master.csvbrbrIn cdr_manager.conf I havebr[general]brEnabled = yesbrbrApparently there is something else that needs to be configured for call brdetail records in 1.4.x. Can someone point me in the right direction?/blockquotedivnbsp;/div/divcdr_manager.confnbsp; is for sending CDR events over AMI (Asterisk Manager Interface). The file you need to edit is cdr.conf nbsp; make sure you have enable = yes in its [general] section default is yes and you can define and custom setting for csv engine in [csv] section of same cdr.conf file and also you can do some settings in cdr_custom.conf file. Look at sample files that come with asterisk for examples. br clear=allbr-- brGodson Gera,bra href=http://godson.in;http://godson.in/a ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pstn call waiting and zap
I have the extension connected to the fxs on the x400p (2 modules) and I use *0 which is actually built into the code to flash the fxo line. Hope this helps. on Tuesday 12/04/2007 C F([EMAIL PROTECTED]) wrote application map in features.conf On 12/4/07, Patricio Valarezo Lozano [EMAIL PROTECTED] wrote: Hi, I hope someone could help me, i have a x100p interface for testing purpose and on each incomming call I redirect the call to handytone 388 atas, the problem comes when i'm during a call and another call comes in, i hear the call waiting beep (comming from the zap channel), but I can't catch the call as usually using flash+2 (my pstn call wait sequence), because when i flash the sip channel i get the tone for transfering. How should i get the call ? i was trying to flash the zap channel using zapflash but it did not work. thanks a lot for your time, i hope have exposed the problem crearly. PV -- patoVala Linux User#280504 Hablando en http://www.elprimoalcahuete.com SlayR i just bought MS Office 2000 for only $20!!! Knghtbrd you got ripped off ; SlayR i know ;) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI commands missing...
Well, this happened to me one time when I forgot to compile the pri library before the asterisk! Could you have done that? on Wednesday 10/31/2007 Tzafrir Cohen([EMAIL PROTECTED]) wrote On Wed, Oct 31, 2007 at 12:06:25AM -0600, Carlos Chavez wrote: I have an Asterisk server running Elastix but patched to use Unicall. Everything seems to be working fine and the TE220 card is up and running with port 1 configured as PRI and port 2 as MFC/R2. We can already send and receive calls on port two but we cannot on port one. That is when we noticed that there are no PRI commands available on the Asterisk CLI. We cannot use PRI DEBUG SPAN to determine why port 1 is not receiving or sending calls. Why would this commands be missing? I wonder how those two should interact. The first thing chan_zap tries to do is to open all of its spans. Maybe it has failed there? Try playing with [trunkgroups] to explicitly tell it to only touch the Zaptel spans that are PRI. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Affordable SIP Trunk for Home PBX ?
And my experience with the unlimited plans is after a certain point -- which is sometimes quite obscure -- they start charging -- sometimes at a rather high rate, so be careful with those. Unlimited means whatever I want it to mean! on Saturday 10/13/2007 Lee Jenkins([EMAIL PROTECTED]) wrote Steve Edwards wrote: On Sat, 13 Oct 2007, Lee Jenkins wrote: I have been using axVoice.com for some about 9 month to a year now and their service is pretty damn good. For home users they have unlimited plan for around 22.00-24.00 U.S. per month. I think the pay as you go plans make more sense for most people -- why do you think the vendors push the flat rate plans? At $25.00 per month, you'd have to be on the phone for about an hour a day for it to be cheaper than a $0.015 per minute plan. True, but I work from home, have a wife and 4 kids with friends and family all over the U.S. so it makes more sense for me. Good point though, Steve. --- Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hiding extensions from app_directory
If you are using freepbx, I think freepbx actually simulates the app_directory, so you may have to do something in the gui to fix it, or they may not have such an option. Hope this helps. on Friday 10/12/2007 Jesse Scott([EMAIL PROTECTED]) wrote Hi Everyone, Sorry in advance if this not the correct place to ask this question, feel free to point me somewhere more appropriate to ask. We have an Asterisk 1.2.7.1 server (about a year old version of Asterisk @ Home with FreePBX) running the phone system for our small office (roughly 15 extensions). I'm trying to hide a couple of extensions from the app_directory generated company directory. I found some information about adding the hidefromdir=yes option to the user's entry in voicemail.conf, but that doesn't seem to have any effect. I'm a little unclear as to whether or not that option is something native to Asterisk or if it comes from one of the external applications. If it is built in, is my version too old to have this feature? Am I totally on the wrong track and is there another way to accomplish this? Thanks, -Jesse ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme conference room duplex issue
I have not noticed this here at all -- although too much of talking over each other makes a mess, but in both 1.2 and 1.4 I have not noticed any such behavior. What are you using for a carrier? on Wednesday 10/10/2007 jamespev([EMAIL PROTECTED]) wrote Hello. We are very successfully using asterisk (in the form of trixbox 2.2/asterisk 1.2). We run a few conference lines for customer teleconferences which mostly work well but they seem to operate at half duplex. If a person starts talking they will cut off others on the call. Is this normal behavior? Are there any options I can change to change this? Thanks! Jamesbr / nbsp;nbsp; Hello.nbsp; We are very successfully using asterisk (in the form of trixbox 2.2/asterisk 1.2).nbsp; We run a few conference lines for customer teleconferences which mostly work well but they seem to operate at half duplex.nbsp; If a person starts talking they will cut off others on the call.nbsp; Is this normal behavior?nbsp; Are there any options I can change to change this?br / br / nbsp;nbsp; Thanks!br / br / Jamesbr / br / ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial-Chain interrupted by Operator Called Party not reachable Messages
Well, I would put in logic so the person has to confirm by hitting 1 or something and otherwise no matter what the phone company does, you will know. We have the same problem here and this is the only thing I can think of to fix it. on Friday 10/05/2007 Christoph Adomeit([EMAIL PROTECTED]) wrote Hi Eric, thanks for your hint. Unfortunately it doesn't work, I just tested it. It seems that at least in german T-D1 Mobile Network a mobile call is answered even if that mobile is switched off. On Thu, Oct 04, 2007 at 05:30:11PM -0500, Eric ManxPower Wieling wrote: Christoph Adomeit wrote: Hi, I have the following problem: I want asterisk to dial a chain of n-numbers until somebody picks up the line. I am using Digium E1 Hardware (zaptel) for dialing out. Dialing a Chain is basically no problem, I use somwthing like: dial(no1,50) dial(no2,50) dial(no3,50) However, If no1 is not reachable, for example it is a mobile and switched off, then some automatic Operator-Voice from the Mobile-Telco says forever: This number is currently not reachable and this means asterisk thinks the call was succesfull and does not continue with the the other numbers. Does somebody has an idea how I can distinguish those Operator Voices from real calls ? This is one of the VERY few times the r option to Dial will be helpful. Dial(no1,50,r) etc. As long as the call is not answered (and the telco does not answer when they play that message) the r option will hide the audio the telco is sending. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Christoph Adomeit GATWORKS GmbH Reststrauch 191 41199 Moenchengladbach Sitz: Moenchengladbach Amtsgericht Moenchengladbach, HRB 6303 Geschaftsfuehrer: Christoph Adomeit, Hans Wilhelm Terstappen [EMAIL PROTECTED] Internetloesungen vom Feinsten Fon. +49 2166 9149-32 Fax. +49 2166 9149-10 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Rhino RCB8FXX
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Re: [asterisk-users] Rhino RCB8FXX
I am actually using 1.1.0, I might take a look at .1, but this should
work with 1.4.
on Tuesday 10/02/2007 Jeremy Mann([EMAIL PROTECTED]) wrote
Latest being 1.1.1 ?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici
Sent: Tuesday, October 02, 2007 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Rhino RCB8FXX
We are using it successfully with zaptel 1.4 -- just be sure and get
the latest drivers which are now independent of the zaptel sources.
on Tuesday 10/02/2007 Jeremy Mann([EMAIL PROTECTED]) wrote
Anyone know if Rhino is planning on supporting zaptel 1.4 anytime soon?
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copying, printing, or use of this information is strictly prohibited and
possibly a violation of federal or state law and regulations. If you have
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Re: [asterisk-users] ChanSpy issue
I am not an expert on chanspy, but it seems to me spying on the trunk would not work very well, would not you hear multiple conversations mixed if more than one extension were calling? Seems best to me to spy on an extension. YOu also can do a show channels to see who is talking to whom. on Wednesday 09/26/2007 Wai Wu([EMAIL PROTECTED]) wrote The parameter to Chanspy should be the whole or part of the channel name. I do not understand what you mean by sip trunk. It make perfect sense that you can hear both streams of voice when you use the phone's extension as Asterisk usually uses SIP/extension+xxx as the channel name of the call. -Original Message- From: [EMAIL PROTECTED] on behalf of Ed Nuñez Sent: Wed 9/26/2007 4:48 PM To: [EMAIL PROTECTED] Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ChanSpy issue Hello list I am having an issue with Chanspy/SIP that I'm hoping someone has come across and resolved in the past. I am sending calls that come in TDM through T1 ZAP channels and go out to a SIP trunk. If I spy on the SIP channel, I can hear the person on the SIP side of the call just fine, but the person on the ZAP channel fades in and out. If I spy on the ZAP channel, and can hear both sides just fine, but I don't know who I am spying on since I have other calls coming in on the same T1. If I spy on a SIP extension instead of a SIP trunk, I hear both sides just fine. I am using a recent version of Asterisk 1.2 and I am using g729 licenses. This is the command I am using to spy. exten = 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4)) !DOCTYPE HTML PUBLIC -//W3C//DTD HTML 3.2//EN HTML HEAD META HTTP-EQUIV=Content-Type CONTENT=text/html; charset=iso-8859-1 META NAME=Generator CONTENT=MS Exchange Server version 6.5.7638.1 TITLERE: [asterisk-users] ChanSpy issue/TITLE /HEAD BODY !-- Converted from text/plain format -- PFONT SIZE=2The parameter to Chanspy should be the whole or part of the channel name. I do not understand what you mean by quot;sip trunkquot;. It make perfect sense that you can hear both streams of voice when you use the phone's extension as Asterisk usually uses quot;SIP/extension+xxxquot; as the channel name of the call.BR BR BR -Original Message-BR From: [EMAIL PROTECTED] on behalf of Ed NuñezBR Sent: Wed 9/26/2007 4:48 PMBR To: [EMAIL PROTECTED]BR Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'BR Subject: Re: [asterisk-users] ChanSpy issueBR BR BR BR Hello listBR BR BR BR I am having an issue with Chanspy/SIP that I'm hoping someone has comeBR across and resolved in the past.BR BR BR BR I am sending calls that come in TDM through T1 ZAP channels and go out to aBR SIP trunk.BR BR BR BR If I spy on the SIP channel, I can hear the person on the SIP side of theBR call just fine, but the person on the ZAP channel fades in and out.BR BR If I spy on the ZAP channel, and can hear both sides just fine, but I don'tBR know who I am spying on since I have other calls coming in on the same T1.BR BR BR BR If I spy on a SIP extension instead of a SIP trunk, I hear both sides justBR fine.BR BR BR BR I am using a recent version of Asterisk 1.2 and I am using g729 licenses.BR BR BR BR This is the command I am using to spy.BR BR BR BR exten =gt; 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4))BR BR BR BR BR BR BR BR BR BR BR /FONT /P /BODY /HTML___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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 [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
