Re: [asterisk-users] asterisk-users Digest, Vol 179, Issue 1
We are not allowed to insert anything into the call path. So somehow we have get S included into call without adding anything into the call path. That’s why I thought a SIP JOIN would work (where device C would handle the multiparty call) – but it sounds like Asterisk doesn’t support that. From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Israel Gottlieb Sent: Monday, July 1, 2019 1:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk-users Digest, Vol 179, Issue 1 how about sticking in a pbx between [c] and [h] so when [h] hangsup you send to [s] if that is 3rd party else i dont see how you could redirect [c] at all else maybe ask them to have [h] redirect [c] to [s] then [h] will also be out of the call On Mon, Jul 1, 2019, 20:03 mailto:asterisk-users-requ...@lists.digium.com> wrote: Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com <mailto:asterisk-users@lists.digium.com> To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-requ...@lists.digium.com <mailto:asterisk-users-requ...@lists.digium.com> You can reach the person managing the list at asterisk-users-ow...@lists.digium.com <mailto:asterisk-users-ow...@lists.digium.com> When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-users digest..." Today's Topics: 1. Re: Second Asterisk server SIP JOIN a call to conduct a post-call survey (Joshua C. Colp) 2. Re: Second Asterisk server SIP JOIN a call to conduct a post-call survey (Jason N) 3. Re: Second Asterisk server SIP JOIN a call to conduct a post-call survey (Joshua C. Colp) -- Message: 1 Date: Mon, 01 Jul 2019 11:15:01 -0300 From: "Joshua C. Colp" mailto:jc...@digium.com> > To: asterisk-users@lists.digium.com <mailto:asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call to conduct a post-call survey Message-ID: mailto:be3a1911-7870-4039-9a35-39f7b5be8...@www.fastmail.com> > Content-Type: text/plain;charset=utf-8 On Sun, Jun 30, 2019, at 11:09 AM, Jason N wrote: > I am designing a solution for a hotel booking call center with the > following (mandatory) design: After the call from the customer with the > booking agent is complete (and the Hotel PBX disconnects from the > call), a second PBX takes over to conduct a survey of how the call > went. Both PBX’s are Asterisk based. > > > So customer phone [C] connects to hotel PBX [H]. Once [H] disconnects, > the survey PBX [S] grabs the call and conducts the survey. [H] must > completely disconnect from the call before [S] can start the survey. > [H] cannot transfer/forward the call to [S]. > > > At a high level the solution seems to be: On [C] connection to [H], [H] > sends call information to [S]. [S] issues a SIP JOIN to [C] and joins > the call. [S] somehow detects that [H] has disconnected and then begins > the survey. > > > Would the above work conceptually? If so, how do I tell Asterisk [S] to > contact [C] and join the call already in progress? (I can get call info > from [H] to [S]). It would be easiest for H to just Dial S after the first call leg is done. This can be done using the 'g' option to Dial[1] which continues dialplan application after the outgoing call leg hangs up. You could even send information as SIP headers if need be so S sees the info. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Dial -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com <http://www.digium.com> & www.asterisk.org <http://www.asterisk.org> -- Message: 2 Date: Mon, 1 Jul 2019 14:53:47 + From: "Jason N" mailto:supp...@telium.io> > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" mailto:asterisk-users@lists.digium.com> > Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call to conduct a post-call survey Message-ID: <0100016bae071017-8cd5329f-5e33-493c-a339-c997586e4708-000...@email.amazonses.com <mailto:0100016bae071017-8cd5329f-5e33-493c-a339-c997586e4708-000...@email.amazonses.com> > Content-Type: text/plain; charset="utf-8" Unfortunately I am not allowed any changes to H's PBX / dialplan.The restriction I have is that upon H's total disconnection from C, that S continue
Re: [asterisk-users] Second Asterisk server SIP JOIN a call to conduct a post-call survey
Unfortunately I am not allowed any changes to H's PBX / dialplan.The restriction I have is that upon H's total disconnection from C, that S continues the call with C. That's why I thought that if I could get S to SIP JOIN the call from C, that once H disconnects S can continue. I can extract the SIP call info on H and pass that to S (so it can join the call). I'm just not sure if this concept is possible/practical. -Original Message- From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua C. Colp Sent: Monday, July 1, 2019 10:15 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call to conduct a post-call survey On Sun, Jun 30, 2019, at 11:09 AM, Jason N wrote: > I am designing a solution for a hotel booking call center with the > following (mandatory) design: After the call from the customer with > the booking agent is complete (and the Hotel PBX disconnects from the > call), a second PBX takes over to conduct a survey of how the call > went. Both PBX’s are Asterisk based. > > > So customer phone [C] connects to hotel PBX [H]. Once [H] disconnects, > the survey PBX [S] grabs the call and conducts the survey. [H] must > completely disconnect from the call before [S] can start the survey. > [H] cannot transfer/forward the call to [S]. > > > At a high level the solution seems to be: On [C] connection to [H], > [H] sends call information to [S]. [S] issues a SIP JOIN to [C] and > joins the call. [S] somehow detects that [H] has disconnected and then > begins the survey. > > > Would the above work conceptually? If so, how do I tell Asterisk [S] > to contact [C] and join the call already in progress? (I can get call > info from [H] to [S]). It would be easiest for H to just Dial S after the first call leg is done. This can be done using the 'g' option to Dial[1] which continues dialplan application after the outgoing call leg hangs up. You could even send information as SIP headers if need be so S sees the info. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Dial -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Second Asterisk server SIP JOIN a call to conduct a post-call survey
And, how would [S] know that [H] has disconnected? (Is there an Asterisk event that indicates one party has disconnected from a multi-party call) From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason N Sent: Sunday, June 30, 2019 10:08 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Second Asterisk server SIP JOIN a call to conduct a post-call survey I am designing a solution for a hotel booking call center with the following (mandatory) design: After the call from the customer with the booking agent is complete (and the Hotel PBX disconnects from the call), a second PBX takes over to conduct a survey of how the call went. Both PBX's are Asterisk based. So customer phone [C] connects to hotel PBX [H]. Once [H] disconnects, the survey PBX [S] grabs the call and conducts the survey. [H] must completely disconnect from the call before [S] can start the survey. [H] cannot transfer/forward the call to [S]. At a high level the solution seems to be: On [C] connection to [H], [H] sends call information to [S]. [S] issues a SIP JOIN to [C] and joins the call. [S] somehow detects that [H] has disconnected and then begins the survey. Would the above work conceptually? If so, how do I tell Asterisk [S] to contact [C] and join the call already in progress? (I can get call info from [H] to [S]). Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Second Asterisk server SIP JOIN a call to conduct a post-call survey
I am designing a solution for a hotel booking call center with the following (mandatory) design: After the call from the customer with the booking agent is complete (and the Hotel PBX disconnects from the call), a second PBX takes over to conduct a survey of how the call went. Both PBX's are Asterisk based. So customer phone [C] connects to hotel PBX [H]. Once [H] disconnects, the survey PBX [S] grabs the call and conducts the survey. [H] must completely disconnect from the call before [S] can start the survey. [H] cannot transfer/forward the call to [S]. At a high level the solution seems to be: On [C] connection to [H], [H] sends call information to [S]. [S] issues a SIP JOIN to [C] and joins the call. [S] somehow detects that [H] has disconnected and then begins the survey. Would the above work conceptually? If so, how do I tell Asterisk [S] to contact [C] and join the call already in progress? (I can get call info from [H] to [S]). 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
Re: [asterisk-users] AMI not responding correctly
I have a C app that communicates with the AMI over a socket. The app works fine (has for years), and it dumps a debug log with all tx/rx traffic. So what I posted is exactly what the AMI is responding with. A telnet session would product the same. I don't have access to the CLI, but I did ask the customer to try that command on the CLI. Your output is exactly what I expect (and what I see on other systems) The real mystery here is why is the AMI on this system responding strangely?! Permissions? Corruption? Some asterisk config file setting I should look at? Jason -Original Message- From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Antony Stone Sent: Wednesday, May 29, 2019 4:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AMI not responding correctly On Wednesday 29 May 2019 at 22:01:11, Jason wrote: > I am communicating How? > with Asterisk 13.18.3 over the AMI and issue the command: > > ActionID: 11 > Action: command > Command: core show calls > > And the response I get is: > > Response: Follows > Privilege: Command > ActionID: 11 > --END COMMAND- What happens if (at pretty much the same time) you run the command "core show calls" in the Asterisk command console? > But where is the call data? On my system (Asterisk 13.14.1) I get: Action: Command Command: core show calls Response: Follows Privilege: Command 0 active calls 0 calls processed --END COMMAND-- > What is going wrong on this system?I confirmed the AMI connection has > full read/write permissions. Why is the call data missing from the > response? How are you connecting and what are you using to parse the response? Try a simple telnet and see if the result is the same: $ telnet localhost 5038 Trying 127.0.0.1... Connected to localhost.localdomain. Escape character is '^]'. Asterisk Call Manager/2.9.0 Action: login Username: myusername Secret: secretpassword Response: Success Message: Authentication accepted Event: FullyBooted Privilege: system,all Status: Fully Booted Action: Command Command: core show calls Response: Follows Privilege: Command 0 active calls 0 calls processed --END COMMAND-- Antony. Response: Goodbye Message: Thanks for all the fish. -- 3 logicians walk into a bar. The bartender asks "Do you all want a drink?" The first logician says "I don't know." The second logician says "I don't know." The third logician says "Yes!" Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMI not responding correctly
I am communicating with Asterisk 13.18.3 over the AMI and issue the command: ActionID: 11 Action: command Command: core show calls And the response I get is: Response: Follows Privilege: Command ActionID: 11 --END COMMAND- But where is the call data? What is going wrong on this system?I confirmed the AMI connection has full read/write permissions. Why is the call data missing from the response? Jason -- _ -- 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] Dante and asterisk
Hi Jerry, I'm Dante Level 3 certified so I can help on the Dante side tho I've never done AES67 on Linux. So I think you can do this although I know you can't do it purely in Dante (which I think you guessed based on the AES67 reference) because you need Dante Virtual Soundcard and it is only available on Windows/Mac. You need to get Dante to create an AES67 stream that your Asterisk box can pickup with an appropriate driver. I'm assuming you have a Dante network up and running and you have the controller software connected to the Dante net and you at least know the basics. The hitch is that your source Dante device must support AES67 streaming. Audinate added support for that to their Brooklyn II cards in 2015 but that doesn't mean your device is running a firmware rev that has support for it. Additionally, I believe OEMs can decide if they want to enable it or not. So before doing anything I would make sure your source supports AES67 streaming. You can open the device view for your source device in the Dante controller and if there is an AES67 Config tab you are good to go. If not, you will have to bounce the audio through another device that does support it (like a mixer) or get one of the Audinate AVIO output dongles and just plug it into a analog sound card in your Asterisk box. Assuming your source support AES67 streaming, in the device view on the Dante controller you'll see an AES67 Config tab. You will probably have to enable it since I don't think any OEMs turn it on by default. That will likely require the device to be rebooted when you make that change. Once you have AES67 enabled, you go through the steps to create a multicast flow just like you would for a normal source sending to more than 4 targets except there should (now) be a check box to make the multicast AES67. Then, add the channel(s) to the flow just like normal. Save it to the controller and you are good. On the receiving side you need something that can receive the AES67 stream and expose that audio to Asterisk. This is where I can't help you anymore but it looks like there are some audio drivers available that should work. All the AES67 receivers need to "find" the stream which could be automatic or manual depending on your network config. Remember that the audio stream is multicast for AES67 and not point-to-point like a regular Dante stream is so if you have router(s) between your source and destination it requires different handling. Good luck! Jason On Mon, Mar 18, 2019 at 9:20 AM Jerry Geis wrote: > I was trying to find if Asterisk supports Dante ? > > Dante -- https://www.audinate.com/ > AES67 -- http://www.aes.org/publications/standards/search.cfm?docID=96 > > Thanks, > > Jerry > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13 attended transfer alcatel
Hi, I've put the sip output here : https://pastebin.com/W7M4zxHA Thanks -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Joshua Colp Envoyé : vendredi 9 juin 2017 11:39 À : asterisk-users@lists.digium.com Objet : Re: [asterisk-users] Asterisk 13 attended transfer alcatel On Fri, Jun 9, 2017, at 04:59 AM, Jason TOMLINSON wrote: > Hello, > > Since upgrading from asterisk 11 to asterisk 13 (I have tested up to > the latest 13.16.0 release), we have a problem with attended transfers > to an alcatel pbx in which the call being transferred still has music > on hold even after the transfer has completed. > Is this a known issue? Any new flags that need setting, etc? There's no filed issues about it that come to mind and no new flags that need setting. I'd suggest providing console output and SIP traffic. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 13 attended transfer alcatel
Hello, Since upgrading from asterisk 11 to asterisk 13 (I have tested up to the latest 13.16.0 release), we have a problem with attended transfers to an alcatel pbx in which the call being transferred still has music on hold even after the transfer has completed. Is this a known issue? Any new flags that need setting, etc? Thanks Jason -- _ -- 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] Running configure from subdirectory of source tree
That's not something that is likely to be supported. Any configure script in the tree will be run via the top-level build process, as needed. Is there some reason you think you need to run the other configure scripts yourself? On 03/05/2014 08:54 AM, Gianluca Merlo wrote: Hello everyone, I would like to seek your advice regarding a build (or rather configure) problem I am running into. For reference, tests are all relative to a build from a 1.8.26.0 tarball, on Debian Wheezy. I would like to understand if it is possible, and if any of you have tried, to build Asterisk from a subdirectory of the source tree, i.e., from a clean source tree # mkdir my-build-directory # cd my-build-directory # ../configure # make I lack a proper amount of knowledge on the matter, but I think that this should be legit with a common autotools build toolchain. Tests suggest that (at least in my case) this is not working with configure: error: cannot find install-sh, install.sh, or shtool in `pwd` ../`pwd` Looking in the configure process in detail, the failure seem to follow the checks (/bin/sh -x output) + for ac_dir in '`pwd`' '$srcdir/`pwd`' + test -f /home/gian/src/asterisk-1.8.26.0/my-build-directory/install-sh + test -f /home/gian/src/asterisk-1.8.26.0/my-build-directory/install.sh + test -f /home/gian/src/asterisk-1.8.26.0/my-build-directory/shtool + for ac_dir in '`pwd`' '$srcdir/`pwd`' + test -f ..//home/gian/src/asterisk-1.8.26.0/my-build-directory/install-sh + test -f ..//home/gian/src/asterisk-1.8.26.0/my-build-directory/install.sh + test -f ..//home/gian/src/asterisk-1.8.26.0/my-build-directory/shtool It looks to me that despite checking `pwd` leads to a correct behaviour, checking ../`pwd` is not correct. I seem to understand that this behaviour was introduced in configure.ac http://configure.ac at r259848, by adding AC_CONFIG_AUX_DIR(`pwd`) The log for the commit reports r259848 | qwell | 2010-04-28 22:32:14 +0200 (Wed, 28 Apr 2010) | 9 lines Merged revisions 259847 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr 2010) | 1 line Add AC_CONFIG_AUX_DIR to configure script, so systems without install can use install-sh from our source dir. Isn't the default behaviour for autoconf enough (http://www.gnu.org/software/automake/manual/html_node/Optional.html)? Can this be considered as a bug in Asterisk's the build system, preventing an otherwise working build scenario (i.e. configuring and building in a subdirectory of the source tree)? Thank you in advance for your help Gianluca -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Broadcasting DTMF to confbridge users or DTMF passthrough
Hi, Trying to properly broadcast / relay DTMF digits to other confbridge users, but does not appear to work. Goal is to have a conference user be able to receive the DTMF, so it has the effect of being 'broadcasted.' I have the following set up in 'confbridge.conf': dtmf_passthrough=yes From logger.conf, I can see the DTMF tones via setting console = dtmf. When I dial into the conference bridge with a SIP UA and dial 9, for example, this is what I see: sip1*CLI [Dec 19 01:29:50] DTMF[22561][C-05ba]: channel.c:4164 __ast_read: DTMF begin '9' received on SIP/3002-003d [Dec 19 01:29:50] DTMF[22561][C-05ba]: channel.c:4175 __ast_read: DTMF begin passthrough '9' on SIP/3002-003d [Dec 19 01:29:50] DTMF[22561][C-05ba]: channel.c:4078 __ast_read: DTMF end '9' received on SIP/3002-003d, duration 110 ms [Dec 19 01:29:50] DTMF[22561][C-05ba]: channel.c:4119 __ast_read: DTMF end accepted with begin '9' on SIP/3002-003d [Dec 19 01:29:50] DTMF[22561][C-05ba]: channel.c:4134 __ast_read: DTMF end '9' detected to have actual duration 59 on the wire, emulation will be triggered on SIP/3002-003d [Dec 19 01:29:50] DTMF[22561][C-05ba]: channel.c:4141 __ast_read: DTMF end '9' has duration 59 but want minimum 80, emulating on SIP/3002-003d [Dec 19 01:29:50] DTMF[22561][C-05ba]: channel.c:4198 __ast_read: DTMF end emulation of '9' queued on SIP/3002-003d sip1*CLI So what is missing here or how to identify / troubleshoot? Is there an application that needs to pass the DTMF from the SIP user in sip.conf to the conference application? Thanks in advance, Jason -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RHEL6 packages - SRTP support?
The packages currently do not support SRTP. On 06/03/2013 10:56 AM, Daniel Pocock wrote: I tried installing the Asterisk 11 RHEL6 packages from packages.asterisk.org I followed this guide: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages The SRTP support appears to be missing though. I notice libsrtp was not automatically installed as a dependency, and no srtp module exists under /usr/lib64/asterisk/modules Is it still necessary to do a source build every time SRTP is needed? Or is the srtp module distributed in some other rpm? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RHEL6 packages - SRTP support?
On 06/03/2013 12:03 PM, Daniel Pocock wrote: I tried building manually from the source RPM Before running rpmbuild, I installed libsrtp-devel and I notice that res_srtp.so is generated during the build However, the rpmbuild fails for other reasons (see the other email I sent to the list about mISDNutils-devel and other spec file errors) Can you confirm the exact procedure you recommend for rpmbuild on a CentOS6 system rpmbuild --rebuild --without tds --without misdn somepackage.src.rpm -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Log rotate not working
On 05/21/2013 10:19 AM, Ahmed Munir wrote:
Hi,
Last year, I installed Asterisk 10.4.2 and enabled logrotate on daily
basis which was working perfect. Now in couple of months back, the
logrotate feature is not working at all but simply appending the logs
in 'messages' file. Listing down down the configuration for logrotate
below;
/var/log/asterisk/messages {
missingok
rotate 5
daily
postrotate
/usr/sbin/asterisk -rx 'logger reload' /dev/null 2 /dev/null
endscript
}
As asterisk is running by user: root so no need set asterisk
permissions 'create 0640 asterisk asterisk' in above configuration.
Please advise so I can resolve this issue.
I believe you want to execute logger rotate, rather than logger reload.
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Re: [asterisk-users] What is bootstrap.sh for ? Possible bug in 11.3.0 ?
On 05/07/2013 05:13 AM, Olivier wrote: 2013/5/7 Matthew Jordan mjor...@digium.com mailto:mjor...@digium.com 2. It appears as if you're running a modified version of Asterisk, in which case all bets are off. This works fine on the Linux build agents, which is what we use to build the tarballs on downloads.asterisk.org http://downloads.asterisk.org. So, no, I don't think there's a bug in the shell script. I can reproduce this behaviour at will on a fresh new untouched asterisk 11.3.0 install on a debian squeeze (see ASTERISK-21760 https://issues.asterisk.org/jira/browse/ASTERISK-21760) Would you say that for a given asterisk version, included configure script should match the one generated by bootstrap.sh ? The bootstrap.sh script is run by developers after making changes that require regenerating the configure script. It isn't needed on an unpatched installation. Having said that - the generated configure script will very rarely match the one provided in the source, since it will change (sometimes significantly) with differing versions of autoconf, et al. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving User Agent To Remote Location
On 01/03/2013 02:23 PM, Markus Weiler wrote: Am 03.01.2013 21:21, schrieb Nick Khamis: Oh that's so smart!!! So, if I did not misunderstand you, for this one call, have: rtpstart=10004 rtpend=1008 do you mean 1_000_8 ? Markus I think he means 10007. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Termination Provider Madness
On 10/03/2012 10:46 AM, Eric Wieling wrote: A port is not a door if there is nothing listening on the port. Open ports are not a security issue. Stuff running on open ports are. Do you have some external software listening on those ports when there isn't an active call? Asterisk isn't listening on them. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do you convert your prompts to an asterisk-friendly format?
On 08/28/2012 10:04 AM, Andrew Latham wrote: On Tue, Aug 28, 2012 at 11:00 AM, Johan Wilfer li...@jttech.se wrote: 2012-08-28 16:44, Andrew Latham skrev: Try this to test with http://www.digium.com/en/products/ivr/audio-converter.php and compare your output first... Interesting. Didn't know about this. It's good for testing, but I would like to automate it. Is the source-code open or available? Yep, check out repotools for that http://svn.asterisk.org/svn/repotools/sound_tools/scripts/ I don't know whether those scripts are what is actually used on the digium.com website, but they are what we use to create the various Asterisk sounds packages on http://downloads.asterisk.org/. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do you convert your prompts to an asterisk-friendly format?
On 08/28/2012 10:32 AM, Danny Nicholas wrote: Does the .c program compile stand-alone or as an add-on? g++ check_sounds.c check_sounds.c: In function âint main(int, char**)â: check_sounds.c:152: error: invalid conversion from âvoid*â to âdirent**â check_sounds.c:154: error: invalid conversion from âvoid*â to âdirent*â There may be issues building it with g++. I just added a basic Makefile, so you should be able to `svn update` and `make`. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems installing DPMA
On 06/12/2012 02:56 PM, Danny Dias wrote: Hi, I'm just trying to install the DPMA on my Asterisk. I already made the upgrade from Asterisk 1.8.5 to Asterisk 1.8.11-cert2. This is what i did: /mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules-185 / *compiling Asterisk-Cert2 1.8.11* /./configure make make install make config / Afther that i register the DPMA license, and finally copied the *res_digium_phone.so* to //usr/lib/asterisk/modules / When i try to load the module on asterisk console this is what i get /*CLI module load res_digium_phone.so Unable to load module res_digium_phone.so Command 'module load res_digium_phone.so' failed. / With /tail -f /var/log/asterisk/message / /[Jun 11 18:53:26] WARNING[2554] loader.c: Error loading module 'res_digium_phone.so': libavahi-client.so.3: cannot open shared object file: No such file or directory [Jun 11 18:53:26] WARNING[2554] loader.c: Module 'res_digium_phone.so' could not be loaded. / Hope you can help Questions like this should usually be directed to Digium support. Your issue can be fixed by installing the package containing libavahi-client. On CentOS: yum install avahi on Debian/Ubuntu: apt-get install libavahi-client3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cannot get Digium Phones back into service after changing sip device name.
On 06/05/2012 10:23 AM, Chet W. Stevens wrote: During testing with the Digium phones I have run into a problem where I try to make a change to the sip device name. I make the device name change in sip.conf then make the matching change to the lines in res_digium_phone.conf. I then do 'sip reload' and 'module reload res_digium_phone.so'. I then end up with phones that I cannot bring into service no matter what I have tried. They act normally as far as seeing the configuration server, allowing me to enter the key, select the user, and then I receive the Error fetching config from proxy. message. I occasionally receive the following error in the CLI: NOTICE[5941]: phone_users.c:1626 token_set_last_info: Phone at '' reconfigureed. Another phone located at 'sip:10.72.65.114:5060' took over the config. I have tried Reset to Factory Defaults on the phones, I have tried clearing the keys from the Asterisk database, I have tried 'sip unregister', I have tried restarting Asterisk and rebooting. I cannot seem to get these test phones back into service. I am only in testing now where the phones are literally at arms reach and I am really nervous what can happen when we go into production. I really hope that I did something wrong in the process and that the phones are not really this fragile. My test system is currently running these versions: Asterisk 1.8.11-cert2 x86_32 DPMA Module: 1.8.11_1.0.1-x86_32 Digium Phone Firmware: 1_0_5_46476 Your help on this is really appreciated. Thank you. The first step would be to contact Digium technical support. They would be happy to assist you with any issues you're having. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Deleting OLD Voicemails
On 05/22/2012 04:54 PM, Danny Dias wrote: There are 4 files for each voicemail: msg.gsm msg.txt msg.wav msg.WAV That is perfectly normal. The .txt file is metadata that contains things like caller ID and duration. Asterisk will also save voicemails into every format you have specified in voicemail.conf. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flashphoner
On 04/27/2012 01:39 PM, Don Kelly wrote: What flavor are flashphoner minties? --Don Dailing flavored. What else? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compiling asterisk with mysql support
On 03/06/2012 12:31 PM, Ron Bergin wrote: Mathew, Each of those odbc modules are unavailable i.e., marked with XXX I even deleted the asterisk build directory and started over, but had the same results. What prereqs do I need besides these: mysql.i386 5.0.95-1.el5_7.1installed mysql-connector-odbc.i386 3.51.26r1127-1.el5 installed mysql-devel.i3865.0.95-1.el5_7.1installed mysql-server.i386 5.0.95-1.el5_7.1installed unixODBC.i386 2.2.11-7.1 installed unixODBC-devel.i386 2.2.11-7.1 installed libtool-ltdl-devel should be a dependency for unixODBC-devel in CentOS, but it is not. You'll need to install that and re-run ./configure. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Group write permissions /etc/asterisk/.
On 03/06/2012 03:44 PM, Karl Fife wrote: It's not a question of whether the default directory permissions are appropriate. I agree with those. What we're talking about here is what happens during updates to an existing directory. I can't see any rationale for changing the group permissions. If the group permissions differ from the installation defaults, it is because the sysadmin needed them to be different in order to implement one or more methods of extensibility / interoperability that make Asterisk so powerful. Absolutely, it would make sense for the installer to check to be sure it has SUFFICIENT permissions to operate properly, but it is a huge leap of faith to assume that it's appropriate to simply delete certain group permissions. Users only in the owner's group if they belong there, no?? The upshot is that ever since upgrading to 1.8 we have to re-re-re-reset the group directory permissions to make things work, and that just seems insane to me if that is a design choice, not a regression. -Karl It should only set them if the directory does not exist. If it's changing them, something is very seriously broken. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Group write permissions /etc/asterisk/.
On 03/06/2012 04:24 PM, Patrick Lists wrote: On 06-03-12 23:07, Karl Fife wrote: Yep. That's what's happening. I'll file a bug. AFAICT it's not a bug but the way RPM works. Regards, Patrick He didn't suggest that he was talking about RPMs. If that's the case, then I take back everything I said. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available
On 03/05/2012 06:34 AM, Eric Germann wrote: Does anyone have an idea on when 1.8.9.3 might show up in the RPM repositories? Thanks! EKG They should be available now. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available
On 03/05/2012 01:49 PM, Eric Germann wrote: Will a 1.8.10.0 build be imminent or should we go ahead and push this in to production with testing? Thanks! EKG ~20 minutes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available
On 03/05/2012 06:00 PM, Lefteris Zafiris wrote: Some packages seem to lag behind, eg asterisk18-addons-mysql is compiled against 1.8.7: asterisk18-addons-core-1.8.7.0-2_centos5 asterisk18-addons-mysql-1.8.7.0-2_centos5 Is this a problem with the repo? Are these packages obsolete/unmaintained or have been replaced by others? They've been replaced. The latest packages are in new repositories and are now more appropriately named. See http://packages.asterisk.org/centos/5/asterisk-1.8/ as an example. Also, as of Asterisk 1.8, the -addons RPMs are now built from the same SRPM as the rest of the asterisk RPMs. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Group write permissions /etc/asterisk/.
On 03/05/2012 06:22 PM, Karl Fife wrote: I notice that the installation of Asterisk 1.8.8 thru 1.8.10 (probably earlier versions too) remove the group write permissions from /etc/asterisk/. which is different than 1.4. And 1.6. Is this expected behavior? If so, what's the rationale? If not, I'll submit a bug report if someone hasn't beaten me to it. -K The difference comes from using `install` rather than `mkdir`. mkdir defaults to a+rwx (777) - umask (likely 002 on your system), whereas install defaults to the much more sane u+rwx,g+rx,o+rx (755). I don't know if I would call it a bug since the switch to install was intentional, but I wouldn't say it's necessarily expected either. I don't really have a strong opinion either way though. If anything, I might be inclined to argue that 750 (or 770) would be more appropriate. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk 10 available in Digium Repository? I doesn't show up
On 02/26/2012 06:22 PM, Patrick Lists wrote: On 25-02-12 19:47, Jason Parker wrote: yum and rpm do not support downgrades. Incorrect. There is yum downgrade. See man yum. yum downgrade is extremely broken. It fails, often, potentially leaving a system in an unrecoverable state. That is not to mention how poorly conceived the concept is. Consider what would happen if a package upgraded some resource to a non-backwards-compatible version. It is completely unsupported on the Digium repositories. Please don't try it - I will not help fix it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk 10 available in Digium Repository? I doesn't show up
yum and rpm do not support downgrades. You can try using `yum shell` to uninstall one version and install another version in one transaction, but you'll have to go it alone. On 02/25/2012 11:49 AM, Ast Coder wrote: Thanks Jason. One more question: Is there anyway to go back on an Asterisk version when using the repository? For example, Asterisk 1.8.9.2 is available now. But I want to use 1.8.9.1. Can I downgrade somehow? I want to test NAT bug issue. Thanks On Thu, Feb 23, 2012 at 11:15 AM, Jason Parker jpar...@digium.com mailto:jpar...@digium.com wrote: On 02/23/2012 10:09 AM, Ast Coder wrote: Hi, I have followed instruction on https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-Prerequisites to add Digium Asterisk repositories but doing a, yum search asterisk only shows me Asterisk 1.4, 1.6, and 1.8. There is no Asterisk 10 and yum install asterisk10 fails. Am I missing something? or Asterisk 10 is just no available in binary? Thanks, There are now repositories for each major version of Asterisk, which have to be explicitly enabled to use them. `yum update` to get to the latest of everything, then do `yum update --enablerepo=asterisk-10`. Asterisk 10 will be installed, and that repository will be enabled permanently. I'll add that information to the wiki shortly. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk 10 available in Digium Repository? I doesn't show up
On 02/23/2012 10:09 AM, Ast Coder wrote: Hi, I have followed instruction on https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-Prerequisites to add Digium Asterisk repositories but doing a, yum search asterisk only shows me Asterisk 1.4, 1.6, and 1.8. There is no Asterisk 10 and yum install asterisk10 fails. Am I missing something? or Asterisk 10 is just no available in binary? Thanks, There are now repositories for each major version of Asterisk, which have to be explicitly enabled to use them. `yum update` to get to the latest of everything, then do `yum update --enablerepo=asterisk-10`. Asterisk 10 will be installed, and that repository will be enabled permanently. I'll add that information to the wiki shortly. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Praking lot issues.
On 02/21/2012 02:55 PM, Bryant Zimmerman wrote: Ok I now have the basics for dynamic parking working but for some reason when a caller calls in and is parked with a transfer the return call dials the sip peer of the caller and not hte peer of the last party that parked the call. Anyone have any ideas on this? Also when a call is transfered to a parking space. the caller hears the space number. How can I stop that as well? Thanks Bryant See https://issues.asterisk.org/jira/browse/ASTERISK-19322 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Streaming musiconhold via mpg123
On 02/21/2012 05:34 PM, Stephen Brown wrote: application=/usr/bin/mpg123 -q -s -f 8192 --mono -r 8000 /var/lib/asterisk/sounds/music/Rolling In The Deep.mp3 Probably unrelated to your issue, but you're going to want to quote that filename. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SER Still recommended for large installs?
I'm reading some information that recommends using SER / OpenSER for large installation to offload SIP traffic from the Asterisk server. http://www.voip-info.org/wiki/view/Asterisk+at+large However, it looks like the information might be dated. I'm looking at a potential 750 SIP phone and 150 Analog installation, all internal network, PRI trunks, and am trying to nail down an architecture. Opinions? You think I skip the SER box if I'm using 1.8? Thanks! -- I get enough exercise just pushing my luck. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP hardware phones
Thanks for the info. As we move forward, we'll be testing and making a phone selections. No doubt we'll run into this. Are you saying if the phone is stated to be a 10/100 phone, it still may not work at 10? On 2/13/2012 1:32 AM, Benny Amorsen wrote: Jason W. Parksjason.w.pa...@gmail.com writes: I can move my voice infrastructure to an IP-based one running 10Mbps, utilize existing wiring infrastructure, with the only cost outlay being low cost PoE managed switches (48 ports for about a grand), and it ends up a lot cheaper than upgrading the data network to support the phones. ...and I can still stay within standard. You can, but not all phones will link up at 10Mbps. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP hardware phones
The existing infrastructure I'm speaking of is the existing voice infrastructure. It currently supports a digital PBX. No IP whatsoever, but the wiring is rated for 10BaseT. As we look to replace the digital PBX with VoIP, my options are to abandon that wiring and start using our data network, or upgrade our existing voice infrastructure to support VoIP. The numbers are showing It would be cost prohibitive to upgrade our existing data network to support VoIP. ...and I think you've just supported one of my reasons for continuing to keep voice and data networks separate. Since the voice network will be completely and physically separate from any non-voice data, and all devices on that network are phones, it just became a lot less complicated. ...and I'm only talking 10Mb between the phone and the switch. All switches would be interconnected either with 100 or 1000. Thanks for the response. Jason Cheer up, the worst is yet to come. On 2/13/2012 2:48 AM, Hans Witvliet wrote: On Mon, 2012-02-13 at 09:32 +0100, Benny Amorsen wrote: Jason W. Parksjason.w.pa...@gmail.com writes: I can move my voice infrastructure to an IP-based one running 10Mbps, utilize existing wiring infrastructure, with the only cost outlay being low cost PoE managed switches (48 ports for about a grand), and it ends up a lot cheaper than upgrading the data network to support the phones. ...and I can still stay within standard. You can, but not all phones will link up at 10Mbps. /Benny -- _ Are you realy shure you want to do that? I mean _existing_ infra (with probably a number of other (non-voip) machines connected to it? Even on a 100Mbps network, if one of the machines on the same network is doing a rsync-job (no saturation), I notice a drop in voip-quality. Adding voip to existing infra might work, if your network is good enough, like Gb with enough unused bandwith and low latency. Or if you can tell complaining users, that it is a temporary solution. hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP hardware phones
Gotcha! That was my plan. I ran into that exact issue when I was randomly speed testing a couple of the lines. The computer under test immediately negotiated to 100Mb and ran just fine, but I know I'm asking for trouble to keep it that way. I will be forcing all ports down to 10. ...and thanks for the example. That's good information. On 2/13/12, Bryant Zimmerman brya...@zktech.com wrote: Jason A standard SIP VOIP phone will use less than 100k per voice call. For example I have several bussiness customers that have a dedicated DSL line and they do up to 6 lines very well on that 1.5x384 (we do g729 which is 37k per call). If your networks drops can test solid at 10mb you should be in good shape if they do not run solid at 100mb you should force the switch port to negoitate to 10mb not 100mb. Make sure the POE switches you are looking at allow you to force the port speed this may save you in the long run. Also make sure that the POE switch can handle the load and run lengths you are looking to put on it. Bryant BrFrom: Jason W. Parks jason.w.pa...@gmail.com Sent: Monday, February 13, 2012 8:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP hardware phones Thanks for the info. As we move forward, we'll be testing and making a phone selections. No doubt we'll run into this. Are you saying if the phone is stated to be a 10/100 phone, it still may not work at 10? On 2/13/2012 1:32 AM, Benny Amorsen wrote: Jason W. Parksjason.w.pa...@gmail.com writes: I can move my voice infrastructure to an IP-based one running 10Mbps, utilize existing wiring infrastructure, with the only cost outlay being low cost PoE managed switches (48 ports for about a grand), and it ends up a lot cheaper than upgrading the data network to support the phones. ...and I can still stay within standard. You can, but not all phones will link up at 10Mbps. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP hardware phones
I'm in a similar situation. However, most of my buildings were re-wired around 1994 to provide Cat5 or 5E to the desktop for data, and 2-pair Cat3 for voice, all in a star topology. I can move my voice infrastructure to an IP-based one running 10Mbps, utilize existing wiring infrastructure, with the only cost outlay being low cost PoE managed switches (48 ports for about a grand), and it ends up a lot cheaper than upgrading the data network to support the phones. ...and I can still stay within standard. Is this an option for you or are you still living with the remnants of an old key system or something like that? The journey of a thousand miles begins with a broken fan belt and a flat tire. On 2/8/2012 10:46 AM, Vieri wrote: Let me answer that, Carlos. A big hospital. These big infrastructures can be quite outdated and messy. Getting someone to cable old parts of the buildings can be very expensive. However, replacing just the backbone switches is something they can afford. And they don't need PoE, really. What kind of applications benefit from gigabit speed? Well, plenty, such as MDs having to view a whole bunch of x-ray images of several patients, as fast as possible. Believe me, doctors aren't patient and Gbps makes a big difference. So basically, that's your answer: these sites don't need PoE, just Gbps and can't afford cabling a huge old building. Now, they don't care for PoE on the hardphones either. So in these cases, I think it's clearly justifiable to have a low-budget Digium D40 or Grandstream GXP280 with a 2-NIC Gbps switch. Not a big deal anyway, because they can always add a mini 5 or 8-port gigiabit switch for around 20$ between the wall socket and the hardphone+PC, but that just adds another appliance to the doctor's office... --- On *Wed, 2/8/12, Carlos Alvarez /car...@televolve.com/* wrote: From: Carlos Alvarez car...@televolve.com Subject: Re: [asterisk-users] SIP hardware phones To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, February 8, 2012, 9:26 AM If the customer is so cheap that they won't properly build out the network, why would they have gigabit switches to the desktop which have a limited set of applications that actually benefit from it? Then there's PoE, which is expensive to start and very expensive with gigabit. So this mythical customer is too cheap to cable, but will buy a gigabit switch of dubious value, will they buy a PoE gigabit switch? If not, why not buy a value-priced PoE 100m switch which has a clear benefit instead of a low-end GB switch of dubious value? I just don't see the fit, and I'm guessing the vendors don't either. What is the exact network topology (brands/models) and applications that justify GB to the desktop, don't justify additional cabling, and how do you account for PoE in this environment? On Wed, Feb 8, 2012 at 7:13 AM, Vieri rentor...@yahoo.com /mc/compose?to=rentor...@yahoo.com wrote: --- On Wed, 2/8/12, Jason W. Parks jason.w.pa...@gmail.com /mc/compose?to=jason.w.pa...@gmail.com wrote: From everything I've researched to date, my understanding is most locations have chosen to double their port density and continue to service the phone and computer on separate ports than to share a single line for both computer and phone. Reason primarily mentioned being troubleshooting concerns. If this is the case, the second port is not required, and become nothing but another gimmick to sell to you. Is this everyone else's experience as well? Well, at some locations, for technical and mostly political reasons, doubling port density so that the computer connects to a separate port is too costly, way over what a 60$ hardphone can cost (eg. Grandstream GXP285). I'd be glad to pay just a tad more for hundreds of basic hardphones, just as long as they can do gigabit. Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.9.0 Now Available
On 01/30/2012 11:06 AM, Eric Germann wrote: We mirror off http://packages.asterisk.org to a staging server, then update from there. When will this show up on packages.asterisk.org? Thanks! EKG The RPMs should be there in a few minutes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cordless SIP phone
In my neck of the woods... A Cordless Phone refers to a cordless handset with a wired base. The phone communicates with the base and can't work without it. It's usually proprietary in nature as well. A Wireless Phone usually refers to any phone communicating via 802.11. No base required. A cell phone is also sometimes referred to as a wireless phone. On 1/23/2012 8:19 AM, Carlos Alvarez wrote: On Mon, Jan 23, 2012 at 7:35 AM, eherr email.eherr9...@gmail.com mailto:email.eherr9...@gmail.com wrote: I have an asterisk box which has Polycom Soundpoints IP335 and IP650s registering to it both locally and remote. I want to be able to incorporate a cordless phone at the remote location; not a wireless phone. I want it to also be able to register to the same asterisk box so it can take calls and transfers. What is the difference between a cordless phone and a wireless one? We use and recommend the Panasonic KX-TGP500 and 550. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6 and 1.8
On 12/28/2011 03:10 PM, Danny Nicholas wrote: Can somebody point me to an explanation from Kevin or Tzafir or someone else up the food chain explaining the differences/benefits of 1.6/1.8 vs 1.4/10.0? Every branch (1.0, 1.2, 1.4, 1.6.0, 1.6.1, 1.6.2, 1.8, 10) of Asterisk contains new features that previous branches did not have. Many of these changes are documented in http://svnview.digium.com/svn/asterisk/branches/10/CHANGES Each branch of Asterisk has a lifecycle, which is documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions. As you can see, 1.8 and 10 are the currently supported branches. 1.4 and 1.6.2 are in security maintenance mode, which means that the only issues that will be fixed are security issues. They will both be EOL in April 2012, and will no longer receive any updates. Short version: If you aren't already using Asterisk 1.8 or higher, you really should be - and soon. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What version to upgrade to...?
On 12/12/2011 09:26 AM, Danny Nicholas wrote: I'm wondering if the bind 161 as root statement is a mis-statement or if not, maybe somebody like Tzafir can explain why since none of the other Asterisk binds require root access (this message is still in 10.0-rc3). This is accurate. Non-root users cannot bind ports =1024. There are ways around it, however. See setcap/CAP_NET_BIND_SERVICE at http://www.kernel.org/doc/man-pages/online/pages/man7/capabilities.7.html I haven't looked at the Asterisk code, but there may be changes necessary to disable that check, if this is enabled. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Standard UIDs, especially for asterisk?
On 11/15/2011 09:58 AM, Tony Mountifield wrote: I see on my CentOS systems that certain users for particular subsystems have standardised UIDs and GIDs. For example mysql=27, ntp=38, sshd=74. My two questions are: 1. Is there a list of these standard assignments somewhere? Googling did not turn up anything for me. 2. Are there standard values of UID and GID reserved for the asterisk user, if used for running Asterisk as non-root.? Cheers Tony There are no standard UID/GIDs for things. They are just system users that have no login shell. They are given lower IDs than normal user accounts (on redhat systems, see -r option to useradd) so that they can be easily distinguished. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Standard UIDs, especially for asterisk?
On 11/15/2011 10:42 AM, Tony Mountifield wrote: Yes, I was hoping to use such a system user and group for asterisk, which would not conflict with any other system package I might install in the future, by virtue of being reserved for asterisk. There shouldn't be any conflict either way. (Properly written) packages don't specify a UID to use - they just get created sequentially, so the next available ID is used. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] shared_lastcall for 1.4.42
Does anyone have a patch for 1.4.42 to enable shared_lastcall? I've seen patches for 1.4.19 and 1.4.24.1 (http://goo.gl/WL6Fx). Thanks, Jason -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon: GPG Key signing event
On 10/20/2011 05:16 PM, Paul Belanger wrote: Greetings, If you are planning on attending Astricon, please take the time to attend the GPG key signing event. More information can be found on the wiki page[1]. [1] https://wiki.asterisk.org/wiki/display/~pabelanger/Astricon+2011+Key+signing+event I fail at wikis and don't know how to add comments (perhaps a permissions thing, with it being in your private space, Paul?). I just wanted to note that if you already have a keypair, you will need to have access to a copy of your *private* key in order to be able sign somebody else's key at the event. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use menuselect.makeopts?
On 10/18/2011 09:52 PM, Luke Hamburg wrote: I think this might actually be a bug. https://issues.asterisk.org/jira/browse/ASTERISK-18137 It is indeed a bug, but it's not the bug you referenced. This issue only exists in 1.8.8.0-rc1. It has been fixed for 1.8.8.0-rc2 which will be released this morning. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Centos RPM packages question
On 10/17/2011 02:22 PM, Ioan Indreias wrote: Hello, Trying to upgrade (from asterisk18-1.8.6.0-1) to the latest RPM version from Asterisk repo I found that asterisknow-version is needed by package asterisk18-core-1.8.7.0-2 How could this be explained? Best regards, Ioan The asterisknow-version package contains the repository files (see /etc/yum.repos.d/) for the repositories on packages.asterisk.org and packages.digium.com. Installing this should have been in the setup instructions. The repository layout has changed significantly, and people that didn't install this package would have been stuck on old versions of Asterisk. We opted to add this package as a dependency (as it should have always been one) to resolve that issue. Short version: This is intentional. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Core show translation 4000ms
On 09/30/2011 09:53 AM, Tony Mountifield wrote: In article 4e85d19f.4090...@digium.com, Kevin P. Fleming kpflem...@digium.com wrote: This is why the output was changed to microseconds from milliseconds; in the older version, the lowest number that should be shown was 1 millisecond, even if the actual amount of time consumed was 10 microseconds (or less). The 1 numbers in the output from the older could easily have been 0.02, which would be closer to the output from the new version. Maybe, but that still doesn't explain why there is a factor of 2000 between some conversions and others. And 4001, 4002 and 4003 are remarkably like a big round number plus a tiny offset! I would agree with the OP that the values shown look suspicious and would bear some investigating... I believe the way it gets calculated was also changed a bit. You'll commonly see numbers that are near multiples of 1000. If I'm not mistaken these are the duration of a context switch (or several context switches), which means that with this output, you can guess that his kernel is probably compiled with CONFIG_HZ_250. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsteriskNow install addons despite license conflict with FFA and G.729
On 07/19/2011 01:02 PM, Michael wrote: On Tue, Jul 19, 2011 at 3:34 PM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: You don't need to install asterisk-addons to be able to store CDRs; you need them to be able to store CDRs in MySQL specifically. If you choose another database that doesn't have licensing restrictions that interfere with usage of non-GPL modules, then you'll be fine. Asterisk 1.6.2.19 includes CDR modules for PostgreSQL and FreeTDS (Microsoft SQL Server), and also generic ODBC support which can be used to connect to MySQL if you wish. Doesn't FreePBX CDR page/engine require MySQL CDRs? Yes, but you don't have to use cdr_mysql to insert into a MySQL database. The cdr_odbc module works just fine for that. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.4 - Google iCal not working
That's not the password. I switched it to that in the config file for realism. I always give some honey out to those who have a sugar tooth. Any ideas on the fix? On Mon, Jun 27, 2011 at 3:24 AM, Matt Darnell mattdarn...@gmail.com wrote: When i reload asterisk, calendar show calendars does not show this. What I am missing? I really need to get this to work! You are missing that you should take out passwords from config files. Hope your gmail account didn't get hacked. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.4 - Google iCal not working
I am trying to integrate Asterisk 1.8.4.2 with Google iCal and I have been unsuccessful. libical-0.44.tar.gz - installed neon-0.29.5.tar.gz - installed i did a make clean, make make install in asterisk. make menuselct [*] res_calendar [*] res_calendar_caldav [*] res_calendar_ews [*] res_calendar_exchange [*] res_calendar_icalendar [calendar1] type = ical url = https://www.google.com/...basic.icshttps://www.google.com/calendar/ical/llc.jason.llc%40gmail.com/public/basic.ics user = llc.jason.llc%gmail.com ;secret = supersecret ; web password secret = Supp0rt1 refresh = 15 ; refresh calendar every n minutes timeframe = 60 When i reload asterisk, calendar show calendars does not show this. What I am missing? I really need to get this to work! Thanks, --J -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Jabber / facebook chat?
On 05/17/2011 07:18 AM, Stefan Gofferje wrote: On 04/17/2011 02:13 AM, Stefan Gofferje wrote: has anyone managed to establish an XMPP connection to the facebook Jabber servers? I'd like to send messages on missed calls vie FB. I finally figured it out. For facebook chat to work you have to use usetls = no usesasl = yes Chat.facebook.com offers TLS but it seems, it's incompatible to res_jabber. To clarify, does that mean that you were able to successfully use facebook chat with sasl? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on digium repo
On 05/16/2011 08:36 AM, Jerry Geis wrote: I an running centos 5. I added this to the digium.repo file in /etc/yum.repos.d directory. [digium-current] name=CentOS-$releasever - Digium - Current baseurl=http://packages.digium.com/centos/$releasever/current/$basearch/ enabled=1 gpgcheck=0 #gpgkey=http://packages.digium.com/RPM-GPG-KEY-Digium Then I did yum install asterisk14 addons | 951 B 00:00 base | 2.1 kB 00:00 base/primary_db | 2.2 MB 00:03 digium-current | 1.1 kB 00:00 digium-current/primary | 33 kB 00:00 digium-current 260/260 extras | 2.1 kB 00:00 extras/primary_db | 244 kB 00:00 updates | 1.9 kB 00:00 updates/primary_db | 544 kB 00:01 Setting up Install Process No package asterisk14 available. What did I miss? jerry You missed the Asterisk repo. Replace all instances of digium.com with asterisk.org (and then Digium with Asterisk). packages.digium.com is Digium modules, such as FaxForAsterisk, whereas packages.asterisk.org is Asterisk, DAHDI, libpri, etc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lead time for RPM's?
On 05/12/2011 02:46 PM, Jason Parker wrote: I'll make it a point to respond to this email when the new builds are available. These builds are now available. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lead time for RPM's?
On 05/12/2011 02:40 PM, Cassius Smith wrote: Hi all Usually I build asterisk from source, but recently have been doing a couple of test installations with packages from the Digium repository. About how long does it take to get from new release announcement into the Digium RPM repository? Specifically 1.8.4 CentOS hasn't made it to the rpm repository yet. Cassius In most cases, we'll have RPMs built and available before the release notifications go out. However, we are currently in the process of rebuilding our build servers, so it has been delayed a few days. I expect that builds will be available in the next day or so. I'll make it a point to respond to this email when the new builds are available. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rates Importer Tool
I know most billing software sell this as a monthly service. You get cd-rom every month where they have collected the published tarrif tables filed with the FCC. You load it on the software to analyze call costs. I'm guessing this is a lot of labor hours/manual work thus they charge for providing it. In particular I am thinking of InforTel for Windows. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A E [Gmail] Sent: Monday, May 09, 2011 2:29 PM To: Commercial and Business-Oriented Asterisk Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Rates Importer Tool Hi All, new to the list. Wondering if anyone has / knows of, a good rate importer tool that can be used to standardize and normalize the ratesheets / rate decks etc. obtained from various carriers so they can be analysed and imported into a DB or be saved as a CSV or something? Thanks so much in advance aeg - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cannot install dahdi-linux on (old) PAE kernel.
On 05/06/2011 01:30 PM, Bob Beers wrote:
Not sure if this will work, but I'd try adding, before line 86:
#Workaround for PAE
%if %{paevar} == PAE
Provides: kmod-dahdi-linux
%endif
Can't actually test it myself, sorry.
- Bob
You'd probably want to modify the kmodtool that comes with it, to just always
provide kmod-dahdi-linux.
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[asterisk-users] ARA table definitions (1.8.*)
Where would one find, or better yet determine from code, all of the table definitions for ARA dynamic families? There seems to be some bits and pieces in various places around the internet, ie. voip-info, the definitive guide, ect. but nothing complete or definitive. I have wondered about this for years. Ideally we would have a script packaged with asterisk source, that could be run and would parse the source and generate the table create scripts, including all table columns, and save to file. We could then go in and customize the script from there, adding or removing columns as needed, ect. Thanks, Jason -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some errors
On 03/15/2011 12:34 PM, Fellipe Paes wrote: why I can't use _. in my dialplan? Because it matches everything. In this case, it's matching the 'h' exten. So when the call gets hung up, it goes to _. and does what you're seeing. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
On 02/23/2011 12:43 PM, vip killa wrote: I recognize all the options given yet as I explained before they are not viable. I do not have the resources to pay someone, I do not have the expertise to fix this issue because according to an asterisk developer any fix in that area would be deeply architectural in nature... what other options are there? Option 3 was wait for someone else with the skills and/or money necessary to fix it. Demanding that somebody fix an issue will not work in any community, open source or otherwise. You'll only be labeled a nuisance and ignored. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk18 rpm issues
On 02/02/2011 02:14 PM, Frank Liu wrote: Hi there, Per the instruction from http://www.asterisk.org/downloads/yum , I setup the yum repository on my Centos 5 x86_64 machine and did a yum install asterisk18 asterisk18-configs then I startup the asterisk (with no changes to config) just to see if it runs, but see below errors in the /var/log/asterisk/messages: [Jan 31 11:41:40] WARNING[2899] loader.c: Error loading module 'res_pktccops': /usr/lib/asterisk/modules/res_pktccops.so: cannot open shared object file: No such file or directory [Jan 31 11:41:40] WARNING[2899] loader.c: Error loading module 'chan_mgcp.so': /usr/lib/asterisk/modules/chan_mgcp.so: undefined symbol: ast_pktccops_gate_alloc I checked the system and can't find the file /usr/lib/asterisk/modules/res_pktccops.so at all. I double checked the rpm file downloaded by yum and res_pktccops.so is not in any rpms. Asterisk should still load fine with this warning. chan_mgcp wouldn't work, but that isn't used very often. I will take a look at it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
On 01/19/2011 12:18 AM, randulo wrote:
Although there's no requisite mention of ${Horrible_Dictator}, can't
we pretend there was, call a Godwin and kill this subject?
That would fall under Quirk's Exception: Intentionally invoking Godwin's Law to
attempt to kill a thread is rarely successful. :)
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Re: [asterisk-users] Asterisk 1.8.2 and digium yum repositories
On 01/19/2011 04:41 AM, Ishfaq Malik wrote: Hi Does anyone have any idea how long it will take for the new release of asterisk 1.8 to make it to the digium yum repositories? Thanks Ish They've been there since yesterday afternoon. It's possible that you hit the repository before the packages were there, causing the refresh timer to be extended (the default is probably 24 hours - but I'd have to check). If they still aren't showing up for you, you can run `yum clean metadata; yum update` -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unexpected dialplan match
On 12/20/2010 11:35 AM, Daniel Tryba wrote:
I was wondering why *...@default should match '_*[0-9a-zA-Z].*0.'
in 1.6.13. Who is making the parse error, * or me?
CLI dialplan show *...@default
'_*[0-9a-zA-Z].*0.' =
1. NoOp(${EXTEN}) [pbx_config]
2. Set(accountcode=${CUT(EXTEN,*,2)}) [pbx_config]
3. Set(extension=${CUT(EXTEN,*,3)}) [pbx_config]
4. Set(CDR(accountcode)=${accountcode}) [pbx_config]
7. ResetCDR() [pbx_config]
8. ...
'.' stops further matching. Your extension ends up being (effectively)
shortened to _*[0-9a-zA-Z].
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Re: [asterisk-users] DAHDI on VMWARE
On 12/02/2010 02:03 PM, Danny Nicholas wrote: Hi gang, We are moving our computers from a cluster of physical machines to a VMWARE server with virtual machines. We investigated and are looking to replace our TDM400P/TDM410P with AEX410P cards. Can we run asterisk with the DAHDI drivers from one of the Virtual machines or is DAHDI going to have to be a native process on the “REAL” machine? Thanks Danny Nicholas VMware has no type of PCI-passthrough feature that I'm aware of. There are virtualization environments that do, but the added overhead is going to make things extremely unreliable. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice quality assessment in Asterisk
Those boxes run around $50k USD, I've only seen them once back in the late 1990s. At work for customer consulting we have very expensive site licenses for Prognosis IPT Assessor which generate great looking pdf reports. We also use Cisco IOS IP SLA however it doesn't have a reporting mechanism. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bert Van Kets Sent: Friday, October 08, 2010 8:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voice quality assessment in Asterisk The professional way is to do a series of test calls, play a reference file and record the audio at the incoming side. You then use both files to calculate a MOS score. This method is used by telco's to do quality checks. https://secure.wikimedia.org/wikipedia/en/wiki/Mean_opinion_score http://voip.about.com/od/voipbasics/a/MOS.htm Bert On 08/10/2010 11:12, Sevana Oy wrote: Hi, How do you typically test voice quality in Asterisk? For example if you like to do load testing, or monitor voice quality and get notified if certain calls had bad quality for proactive maintenance? Thank you! Best Regards, Sevana Oy http://www.sevana.fi - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk SIP woes
Hi Guys, Hope fully somebody out there will have experienced this and can shed some light on how it was overcome. Current setup includes asterisk 1.6.2.11, GNU GK and a Quintum Tenor CMS on the same lan. Earlier I was unable to make a sip call from the CMS back to a sip client registered on my asterisk box. So I moved onto passing the call from the Quintum CMS to a Quintum Tenore DX which is also on the same lan and is registered as a sip client to the available asterisk machine but can't get it to route a call out to another sip client. Following are the resulting logs when this call fails. 10.152.0.7 Quintum CMS 10.152.0.120 Asterisk 10.152.0.155 GnuGK 10.152.0.248 Quintum DX ch |01/01| 2010/09/10|16:53:42:560 |h323[-1774588488] [0]: h323mgr:RcvIncomingCall [ch] Call Source 10.152.0.155:46124 ch |01/01| 2010/09/10|16:53:42:635 |CH: iprg name=IPRG-default. ocall[177]: FaxRelay:1 FaxModemCoding:0 RXGain:0 TXGain:0 IdleNoiseLevel:-7000 QOSType:0 QOSValue:176 ocall[177]: rejectNoCID:0 minimumANILength:1 rejectNoCIDCause:21 ocall[177]:RcvSetup() iprgIndex=0 numIncCalls=0/-1 maxTalkTime=0 extRouteReq=0. bandwidth info: max=-1 cur=12600. ocall [177]: Fast start element present. ch |01/01| 2010/09/10|16:53:42:640 |calling-called media type=9(4). called-calling media type=9(4). H323 [177]:Setting remote rtp port=10.152.0.7:10312 ps=0. ch |01/01| 2010/09/10|16:53:42:645 |ocall [177]:Remote side packet saver version = 3. translateCID =12345678. UNSPECIFIED_INDEX h323[177] [0]: tcall:doTranslation inc=1 iprgIndex=0. CallInfo [178]: origCalled.digit(61008) callingparty (12345678) . ch |01/01| 2010/09/10|16:53:42:655 |h323[178] [-1976852028]: ocall:stackSendCallProc ocall:setIPMedia():Setting My IP to 10.152.0.248 H323OrigCall::stackSendCallProc(), EncryptRTP(10356--0x147a) ch |01/01| 2010/09/10|16:53:42:660 |Routing requested for: Orig#=61008 NPI=1(public) TON=1 Normalized#=61008 NPI=1(public) TON=1 Incoming SRC:10.152.0.7 CallingParty:12345678 Route code= selected TG=0 0 match(es) found: Route response [178]: result=0 cause=34. use the cause from previous attempt ifone available CallInfo[178]: fail event. cause=34 legno=0 leg=0 sentLeg=0. CallInfo [178]: discTickm(-179930606)connTickm(0) duration (0) . CallInfo[178]: send eventFailed 0. h323[178] [-1976852028]: ocall:stackSendRelease Any ideas on what i can try please? let me know if you need additional configuration files please. thank you. Regards, Jas - IMPORTANT NOTICE : The information in this email is confidential and may also be privileged. If you are not the intended recipient, any use or dissemination of the information and any disclosure or copying of this email is unauthorised and strictly prohibited. If you have received this email in error, please promptly inform us by reply email or telephone. You should also delete this email and destroy any hard copies produced. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial_exec_full problems with TDM400 - getting critical.
Hi, I thought you'd cracked it, I simply turned off all sip by removing the sip.conf but after a few more days it did the same. I've set logging permanently on again. Any other suggestions? Cheers, Jason. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]on Behalf Of A J Stiles Sent: 17 August 2010 10:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dial_exec_full problems with TDM400 On Tuesday 17 Aug 2010, Jason Morgan wrote: Hi, I was running asterisk 1.4, but recently upgraded to 1.6 (for fax support) at the same time as moving from Ubuntu hardy to I have a single TDM400P rev I with two fxo and two fxs modules, these were working perfectly for years on Asterisk 1.4 using Zaptel drivers with Oslec. Now I've moved to 1.6 so I am using Dahdi. Distribution is stock ubuntu package. After several hours (perhaps 24 or so, not nailed it down precisely) incoming calls are not answered and outgoing calls get dial_exec_full. Incoming calls are reported to either A:just ring and ring, or B:get an engaged tone. Strangely when this happens asterisk DOES see the incoming call in situation A, but fails to answer. What tests can I do to resolve this as it is very inconvenient as we are missing a lot of calls? Have you got any extensions defined that aren't physically connected to anything? I replaced an ancient Asterisk version with a bang-up-to-date 1.6 version I built myself, and was getting similar symptoms to what you describe. It seemed not to be freeing up channels it was trying to associate with non-existent devices. I made sure that every entry in sip.conf had a corresponding phone plugged in somewhere, then went through the dialplan and removed all references to anything that wasn't mentioned in sip.conf. (And there were a few.) It seems to have stayed up since then . -- AJS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opensource Speech recognition for Asterisk
I'm not aware of an open source speech product. Some great examples where speech recognition works well are 1-800-USA-RAIL, Microsoft/Cisco corporate directory 425-882-8080 you can say the employees name and be connected and those works great, 1-800-Goog-411 also works well. Windows 7 Speech Recognition, Dragon Natually Speaking work pretty good. Vonage does a good enough job of sending my home voicemails to my email in Speech to Text, I use this app daily, rarely having to listen to actual voicemails. What Speech-Text doesn't convey is anger/happiness, etc. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Sunday, August 22, 2010 11:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Opensource Speech recognition for Asterisk On Saturday 21 August 2010 17:21:30 Zeeshan Zakaria wrote: I yet have to see ANY working speech recognition software, free or not. This technology is nothing more than a joke so far, not practical at any level. As for free, there is nothing decent. Actually, speech recognition works fine across the board AS LONG AS you use a limited grammar set. It's the arbitrary language speech recognition that needs to be trained to a particular voice. However, arbitrary language isn't normally a common case for IVR systems, which need a limited set of responses in order to decide the proper branch in a decision tree. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dial_exec_full problems with TDM400
Hi, I was running asterisk 1.4, but recently upgraded to 1.6 (for fax support) at the same time as moving from Ubuntu hardy to I have a single TDM400P rev I with two fxo and two fxs modules, these were working perfectly for years on Asterisk 1.4 using Zaptel drivers with Oslec. Now I've moved to 1.6 so I am using Dahdi. Distribution is stock ubuntu package. After several hours (perhaps 24 or so, not nailed it down precisely) incoming calls are not answered and outgoing calls get dial_exec_full. Incoming calls are reported to either A:just ring and ring, or B:get an engaged tone. Strangely when this happens asterisk DOES see the incoming call in situation A, but fails to answer. What tests can I do to resolve this as it is very inconvenient as we are missing a lot of calls? At the moment I have a terminal open all the time with verbose=10 and debug=10, sadly this log is not written to the logfiles so is lost when the terminal exists (perhaps there is a way round this, I don't know) Shutting down asterisk and restarting dahdi removes the problem for another day. Asterisk is version 1.6.2.5-0ubuntu1 Dahdi is version 2.2.1 Any help appreciated. I am at a complete loss what to do, except go back to the old 1.4 server. Cheers, Jason. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial_exec_full problems with TDM400
Hi AJ, Surely this is a really bad bug if unconnected SIP devices ( a very likely occurance ) can take out trunk lines. Anyway what you say is true, there are several sip phones defined and not all are physically present all the time. I'll remove definitions of all sip phones for a while and see what that does. Cheers, Jason. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]on Behalf Of A J Stiles Sent: 17 August 2010 10:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dial_exec_full problems with TDM400 On Tuesday 17 Aug 2010, Jason Morgan wrote: Hi, I was running asterisk 1.4, but recently upgraded to 1.6 (for fax support) at the same time as moving from Ubuntu hardy to I have a single TDM400P rev I with two fxo and two fxs modules, these were working perfectly for years on Asterisk 1.4 using Zaptel drivers with Oslec. Now I've moved to 1.6 so I am using Dahdi. Distribution is stock ubuntu package. After several hours (perhaps 24 or so, not nailed it down precisely) incoming calls are not answered and outgoing calls get dial_exec_full. Incoming calls are reported to either A:just ring and ring, or B:get an engaged tone. Strangely when this happens asterisk DOES see the incoming call in situation A, but fails to answer. What tests can I do to resolve this as it is very inconvenient as we are missing a lot of calls? Have you got any extensions defined that aren't physically connected to anything? I replaced an ancient Asterisk version with a bang-up-to-date 1.6 version I built myself, and was getting similar symptoms to what you describe. It seemed not to be freeing up channels it was trying to associate with non-existent devices. I made sure that every entry in sip.conf had a corresponding phone plugged in somewhere, then went through the dialplan and removed all references to anything that wasn't mentioned in sip.conf. (And there were a few.) It seems to have stayed up since then . -- AJS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 fax between ATA's and Asterisk and Cisco PGW 2200
WireShark does a good job showing the T38 communication. Most products you can also set packet redundancy to send 2 packets. Your setup was T.38 ATA to T.38 Gateway with PRI/ANALOG/PSTN/G.711. I've heard various problems with SIP/PSTN and faxing, around jitter/packet loss and it's not supported by Verizon SIP and others. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of P Z Sent: Thursday, July 29, 2010 7:49 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] T.38 fax between ATA's and Asterisk and Cisco PGW 2200 To provide a reliable fax solution for users connected to a Asterisk 1.6.2.6 server i have tested a few T.38 capable ATA's: - Patton M-ATA - Grandstream HandyTone 486 - Fritz!Box 7170 I have tried Asterisk 1.6.2.6 compiled with SpanDSP-0.0.6pre17 and also Asterisk 1.6.2.6 with Fax for Asterisk installed. These Asterisk servers are connected to a Cisco PGW 2200 + AS5400XM. Sending fax messages from all ATA's to the PSTN (so ATA - Asterisk - PGW - PSTN) failed with a variety of error messages so i tested the different steps one by one. ATA's - Asterisk ReceiveFax: So far i have only succeeded in sending fax messages from the Fritz!Box 7170 to both Asterisk configurations using the ReceiveFax application. Sending fax messages from the other ATA's to Asterisk using the ReceiveFax application failed. ATA's - PGW: To exclude Asterisk i have connected the ATA's directly to the PGW; no success either. Asterisk - PGW: To exclude the ATA's i used the Asterisk SendFax application to send a TIFF file to a landline each time with a different fax machine connected to it. Results: Asterisk 1.6.2.6 compiled with SpanDSP-0.0.6pre17 : asterisk[1367]: WARNING[18591]: app_fax.c:223 in phase_e_handler: Error transmitting fax. result=19: Received other than DIS while waiting for DIS. asterisk[1367]: WARNING[18591]: app_fax.c:820 in transmit: Transmission failed asterisk[1367]: WARNING[18906]: app_fax.c:223 in phase_e_handler: Error transmitting fax. result=20: Received no response to DCS or TCF. asterisk[1367]: WARNING[18906]: app_fax.c:820 in transmit: Transmission failed asterisk[1367]: WARNING[18986]: app_fax.c:223 in phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely. asterisk[1367]: WARNING[18986]: app_fax.c:817 in transmit: Transmission error Asterisk 1.6.2.6 with Fax for Asterisk : asterisk[7092]: WARNING[3167]: res_fax.c:1529 in sendfax_t38_init: Audio FAX not allowed on channel 'SIP/out.to.pgw-000b3f49' and T.38 negotiation failed; aborting. asterisk[7092]: ERROR[3167]: res_fax.c:1650 in sendfax_exec: error initializing channel 'SIP/out.to.pgw-000b3f49' in T.38 mode asterisk[7092]: VERBOSE[3226]: -- FAX handle 0: [ 028.000627 ], entering CLOSING state asterisk[7092]: VERBOSE[3225]: -- Channel 'SIP/out.to.pgw-000b3f72' FAX session '11' is complete, result: 'FAILED' (FAX_FAILURE_TRAINING), error: '3RD_FRM_CHECK_ERROR', pages: 0, resolution: 'unknown', transfer rate: '2400', remoteSID: '' asterisk[7092]: WARNING[3272]: res_fax.c:1529 in sendfax_t38_init: Audio FAX not allowed on channel 'SIP/out.to.pgw-000b3f8b' and T.38 negotiation failed; aborting. asterisk[7092]: ERROR[3272]: res_fax.c:1650 in sendfax_exec: error initializing channel 'SIP/out.to.pgw-000b3f8b' in T.38 mode My questions: - Does anyone have experience with T.38 fax with a setup like this: ATA - Asterisk - PGW - PSTN? - Does anyone have experience in connecting Asterisk to a Cisco PGW 2200 + AS5400XM? - Are there any tools to debug T.38 traffic? Thanks! - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX authentication oddity - Known issue? Fixed?
On 07/28/2010 11:32 AM, Tilghman Lesher wrote: They permit what packets will even reach user2 It should also be pointed out that the config option is permit, and not allow. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ringback tone after MOH, before queue member bridged
I normally work with other 3rd party IVRs, usually once the Agent is Reserved we signal the phone system to play Music on Hold while it's ringing the Agent. The trick here is to replace the Music on Hold with a fake ring file. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ad...@3a.hu Sent: Friday, July 23, 2010 3:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ringback tone after MOH, before queue member bridged Good morning, i've noticed many times that there are IVRs that play a ring tone just before bridging me to an agent. My asterisk does not behave like this but i've always wanted to. I'm now playing with 1.6.2.9 and i've read in queue's doc: http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue R — stops moh and rings once an agent is ringing (Asterisk Trunk) (in queue's optinal parameters). Could someone please explain this line to me? I've set this option, i have a softphone and an ATA registered to *, pure SIP, nothing more. It's not working, either i'm using the r option, which disables MOH and just rings, or i'm using R which gives me MOH but no ringing. It's nothing major, it just would be nice to have. thanks adam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice prompts
On 07/19/2010 01:23 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias Sent: Monday, July 19, 2010 1:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Voice prompts Have now installed a swedish prompt set In /var/lib/asterisk/sounds/se I run elastix And set Language=se in /etc/asterisk/sip.conf But not work \-\- \-\- Show your CLI output so we see that you are getting correct playback Here's a QD snippet to let you do a verification Exten = 1234,1,answer Exten = 1234,n,Set(CHANNEL(language)=se) Exten = 1234,n,playback(tt-monkeys) Exten = 1234,n,playback(vm-goodbye) Exten = 1234,n,hangup Danny, When bottom posting, something you should keep in mind is that a -- on a line by itself causes most email clients to consider anything below it a signature (a sane client will lighten the text, and it won't appear when you hit reply). It would make things much nicer if you were to also remove that part of the signature on your replies. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] centos 5 rpm pacakges (add asterisk16-xmpp module)
On 07/15/2010 08:16 AM, Vasiliy G Tolstov wrote: Hello. Who can add asterisk16-xmpp module to packages.asterisk.org or build asterisk with support xmpp and update packages? Thank You. This is something we've been considering for a while. It should make its way onto the list shortly. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Complex Dialplan Help Needed
I think you need to ask your SIP provider about Redirecting Header, ask what they support and how-to. I work more with Cisco CallManager and SIP Rediversion Header is new in CallManager 8x. Not sure about Asterisk. We have this same problem with Cisco Mobility/Single Number Reach, providers usually won't accept a Calling Party Number that isn't in your range, some will. http://www.voip-info.org/wiki/view/RDNIS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Geoffrey Yeoh Sent: Monday, July 12, 2010 6:10 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Complex Dialplan Help Needed Hello all, I have a project which requires me to rout calls from ten blocks of sequential numbers i.e. 02081000100 - 02081000200 (each block - 100 numbers) coming in from a telco gateway via Dahdi-SS7 to 10 specific numbers outside the box through two to three SIP trunks (trunk 2 and 3 will be spare capacity/redundant for trunk 1). CLI is crucial here as I need to forward the CLI of the numbers from the blocks of numbers from the SS7 gateway, not the CLI of the originating caller. The Asterisk is behind a firewall with NAT setup. The traffic is one way only. Calls going to the switch goes to Asterisk, Asterisk accepts the call, looks at the CLI from the line (not the caller), routs the call to its assigned outside number through the primary trunk. If primary trunk is unavailable, trixbox will then rout the call to the spare trunks on the list. Hope anyone who has setup this before could give me some good tips on how to set this up. Geoffrey -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi problems with kernel 2.6.32
On 05/26/2010 08:00 PM, cov...@ccs.covici.com wrote: From another thread, I blacklisted netjet and now things are working. But I wonder what is going on here and where did netjet come from -- it doesn't look like an dahdi module to me. It comes from mISDN. It is a very badly misbehaving module. IIRC, it wildcards a large portion of tigerjet PCI IDs. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] include sip configuration from another file in sip.conf
On 05/12/2010 01:03 PM, Robert Wagner wrote: Hi, when i include a sip configuration from another file in my sip.conf using #include /etc/asterisk/sip-sipgate.conf everything seems to be working. The peer is listed when i execute sip show peers and Status is OK. But the peer is not listed using sip show registry. I need to place the register = ... in the sip.conf to make it work. Is this working as expected or is it a bug? Working as expected. When you #include a file, the #include line is replaced with the contents of the file. Meaning your register line is likely being placed inside the previous context. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] REALTIME in 1.2
I am trying to change a 1.6 realtime statement into a 1.2 realtime
statement and I know much has changed. I wish I could just upgrade, but
alas not right now.
exten =x,n,Set(NULL1=${REALTIME(schedules,id,${SCHEDULE})})
comes back with
pbx.c:1371 ast_func_read: Function REALTIME not registered
I am not stuck with realtime, I just have a mysql database with info
that changes and needs to update the dialplan accordingly.
Jason
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[asterisk-users] Recording music in Queue
I know that this is a feature but I would like to have the hold music recorded while a person is on hold. So I know the agent put them on hold and not just muted. I have monitor-join=yes monitor-format=wav in my queues.conf any ideas? Per http://www.asteriskguru.com/tutorials/queues_conf.html The best part is no recording will be initiated while the people are listening to music on hold Jason Confidentiality Statement Notice: This email is covered by the Electronic Communications Privacy Act, 18 U.S.C. 2510-2521 and intended only for the use of the individual or entity to whom it is addressed. Any review, retransmission, dissemination to unauthorized persons or other use of the original message and any attachments is strictly prohibited. If you received this electronic transmission in error, please reply to the above-referenced sender about the error and permanently delete this message. Thank you for your cooperation. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mISDN installation via yum
Michael Nausch wrote: HI, I tried to install asterisk and mISDN via http://www.asterisk.org/downloads/yum My machine is running with kernel-2.6.18-164.15.1.el5.i686 Packages for that kernel version were missing. That was an oversight and has been corrected. A `yum update` should be enough to solve this for you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change in menuselect handling of sound files (in 1.6.1.X)
Olivier wrote:
Hi,
Between 1.6.1.9 and 1.6.1.18, handling of menuselect has changed in such
a way that I cannot script non-english sound files downloading anymore.
The following used to work (unattended) with 1.6.1.9 (for instance):
cd /usr/src/asterisk-${ASTERISK_VERSION}
./configure
make menuselect.makeopts
echo MENUSELECT_CORE_SOUNDS=CORE-SOUNDS-EN-GSM CORE-SOUNDS-FR-GSM
menuselect.makeopts.defaults
make USER_MAKEOPTS=menuselect.makeopts.defaults menuselect.makeopts
make
make install
Now, with 1.6.1.18, CORE-SOUNDS-FR-GSM is not downloaded anymore.
I quickly compared both Makefile contents but it's too complex for me.
How should I change my script to download sounds files ?
Regards
Remove this line:
make menuselect.makeopts
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[asterisk-users] D-Channel Span Up without Down
I am getting a bunch of Primary D-Channel on span 1 up but there was not a down message before that. Is this normal? Confidentiality Statement Notice: This email is covered by the Electronic Communications Privacy Act, 18 U.S.C. 2510-2521 and intended only for the use of the individual or entity to whom it is addressed. Any review, retransmission, dissemination to unauthorized persons or other use of the original message and any attachments is strictly prohibited. If you received this electronic transmission in error, please reply to the above-referenced sender about the error and permanently delete this message. Thank you for your cooperation. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.6.1/1.6.2 binary packages
Pablo Ruiz wrote: Hello, Does anyone know when we will see asterisk 1.6.1 (and/or 1.6.2) binary packages at packages.asterisk.org http://packages.asterisk.org? Greets. Packages for 1.6.2 will be available Real Soon Now. It's near the top of my short list. They exist, and are sitting in a(n internal) testing repository. Mostly, I just need to make sure upgrades go smoothly. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.6.1/1.6.2 binary packages
bruce bruce wrote: Thanks for the update Jason, How do the upgrades work if v1.6.0 is already install and one wants to upgrade to 1.6.2 (once it's available)? yum upgrade asterisk* ??? Thanks It should be as easy as a `yum update`. That's the goal, anyways. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime Issue
It seems that my realtime is not assigning channel variables correctly.
INFO
Asterisk 1.6.0.26
Exten.conf
exten = _X.,1,NoOp()
exten = _X.,2,Set(DEVICE=${CUT(CHANNEL,,1)})
exten = _X.,3,Set(NULL=${REALTIME(agents,device,${DEVICE})})
exten = _X.,4,NoOp(DEVICE is ${DEVICE})
exten = _X.,5,NoOp(USERNAME is ${USERNAME})
exten = _X.,6,NoOp(username is ${username})
CLI
-- Executing [...@default:1] NoOp(SIP/1156-55ce, ) in new
stack
-- Executing [...@default:2] Set(SIP/1156-55ce,
DEVICE=SIP/1156) in new stack
-- Executing [...@default:3] Set(SIP/1156-55ce,
NULL=username=john.smith,name=John
Smith,department=Dept_A,routable=no,extension=1234,device=SIP/1156,voice
mail=no,monitor=yes,visible=yes,date_modified=2010-02-09 14:12:01,) in
new stack
-- Executing [...@default:4] NoOp(SIP/1156-55ce, DEVICE is
SIP/1156) in new stack
-- Executing [...@default:5] NoOp(SIP/1156-55ce, USERNAME is )
in new stack
-- Executing [...@default:6] NoOp(SIP/1156-55ce, username is )
in new stack
So I can see it is getting info from the database in Line 3
But only the direct set variable command (Line 2) and Result (Line 4)
work
Lines 5 and 6 do not get the john.smith assigned
Help
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[asterisk-users] Software for my laptop to send Fax via H.323 ?
I'm trying to test a Diaglogic BrookTrout SR140 card. It uses H.323. Trying to find a way I could use my laptop to send a fax over H323 to the BrookTrout card for testing. Any thoughts? Normally I'd setup a FXS interface on a Cisco router and setup a h323 dial peer to the BrookTrout, but I didn't the router with me! - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as a skinny/sccp client?
Brian J. Murrell wrote: I wonder if Asterisk's skinny/sccp channel driver could be used as a client to register with a Cisco PBX. That is, along with a SIP client, say, have Asterisk and said SIP client stand in for a Cisco phone, or an IP Communicator. Anyone done this? Cheers, b. No, this isn't currently possible. I did ponder this for a while, but my conclusion was that the effort required to do so would far outweigh any benefit you'd gain from it. Cisco has been moving to SIP for a very long time. There aren't any phone features that Asterisk could emulate that would make this any better than SIP (or even anything approaching parity). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phones won't stop ringing
I'm experiencing runaway ringing too, can we make this a class action against someone? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff Brower Sent: Wednesday, March 10, 2010 10:20 PM To: Chris Owen Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Phones won't stop ringing Chris- Sounds like the Toyota bug has migrated to Asterisk... it's mutated into runaway ringing :-) -Jeff Sorry for my attempt at levity; just couldn't help it plus I'm sure Digium guys will know how to resolve. We're having an issue that isn't easily googleable so I thought I might might try here. We have several customers who want all their extensions to ring on incoming calls. Frankly I think it is craziness to ring 11 extensions all at once but that is how they want it. We're doing this by creating an incoming route that goes to a hunt list containing all the extensions. This normally works fine but occasionally when someone picks up the call other phones don't seem to realize the call has been answered and will continue to ring. On at least once occasion I saw a call that went to voicemail and all the phones continued to ring. When this happens the phones will continue to ring forever. The only way to stop them from ringing is to pickup the handset at which time they realize there is no call and reset. I'm pretty sure the underlying cause of this problem is funkiness in their network but it just seems to happen too easily and then once it stops it won't stop.Even if this is caused by network issues is there anything I can do to mitigate the problem. Just seems wrong that the phones would continue to ring forever. Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Identify scripts connecting to the asterisk manager
Is there any easy way to identify which script or service is connecting to the Asterisk manager? Somewhere on my system a script or service is trying to connect with a bad user name or password. I get the following error: connect attempt from '127.0.0.1' unable to authenticate I thought maybe I could do a tcpdump on port 5038 and try to fish out the bad username or password but I wasn't able to see any passwords or usernames in plain text. Any way I could maybe change the logging in Asterisk to show me the username that is not able to authenticate? - Jason -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk RPM's
Jay Vocaire wrote: Thanks for researching this for me. If you actually look at the link you sent me, you will see that the latest is: asterisk16-core-1.6.0.21-1_centos5.x86_64.rpm 20-Jan-2010 15:45 11M So, we come back to my original question: is there a reason for the delay on getting the RPM's out? Btw- I am doing yum update, it seems to agree with the above, that the latest RPM is .21. Thanks. -Jay Usually the RPMs are available at the same time new source tarballs are released. This time, that was not the case. Updated packages are available now, however. To force a refresh of repository information and upgrade, you can run `yum clean metadata; yum update`. -- _ -- 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 built kmod-dahdi-linux for PAE kvariant from SRPM
stephen.hindma...@bt.com wrote: rpmbuild --bb ~/localrpms/SPECS/dahdi-linux-kmod.spec snip error: Failed build dependencies: kernel-devel = 2.6.18-164.11.1.el5 is needed by dahdi-linux-kmod-2.2.1-1_centos5.2.6.18_164.11.1.el5.i386 Add a --target=i686 to your rpmbuild line. -- _ -- 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] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory
Brian wrote: Each time the server is rebooted Asterisk duly deletes the manually created /var/run/asterisk directory - quite why it does this I just don't know - perhaps it is a bug? Your assumption is incorrect. Some Linux distributions will empty /var/run/ on boot, just as they do with /tmp/. I do believe you're right, however, in suggesting that there is a bug in Asterisk. It appears that Asterisk creates /var/run/asterisk/ during install and assumes that it will always exist. Some of the sample init scripts (Debian) create that directory before starting Asterisk. This should be done in all of them (or in Asterisk itself, maybe?). Please report an issue on http://issues.asterisk.org/ -- _ -- 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] Could Asterisk be crashing under high context switches?
Hello! I have been struggling with Asterisk 1.6 and DAHDI for the past few weeks. We are an outgoing call center with 30 internal analog phones hooked up to 2 Rhino CB24 channel banks. The banks are connected to a Rhino R4T1 card in a Dell 2950 server with 8 gigs of RAM. The 2 other ports on the R4T1 go to our 2 PRIs. In this configuration, we have trouble maintaining stability. It may be fine for days, but soon the load slowly creeps up on the server from below 1 all the way up to 6 which is when no one can dial out and asterisk pretty much has to be killed to be stopped. We also have bandwidth.com set up as a SIP provider. If we use bandwidth.com, stability is greatly improved. I installed munin on the phone server yesterday and noticed something dramatic, I think! Asterisk became unstable 3 times yesterday. 2 of those times, the number of context switches went to almost 80k the first time, then over 70k the second. First question - is this abnormal for around 20 ongoing recorded calls? I did a little bit of searching and found this: http://wiki.sangoma.com/files/wanpipe-linux-asterisk-tutorials/How_to_Reduce_Asterisk_System_Loads.pdf It talks about zaptel/DAHDI chunk size and that directly affects system load. Second question - the document explains how to change the chunk size for Sangoma hardware. Is there a general way to do that for DAHDI? Thanks is advance! Jason Martin Metrix Matrix, Inc. 785 Elmgrove Rd, Bldg 1 Rochester, NY 14624 Office: 888-865-0065 x202 Mobile: 585-705-1400 ___ -- 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] Setting outgoing callerid on when using a PRI
Doug Lytle wrote: Dave Fullerton wrote: Note num and not number I don't know if that was a change from 1.4 to 1.6 or if Doug mistyped it. Not a mistype. I've been using number all along, but looking at the docs shows that I've been incorrect. It must concatenate the number down to num. Looks like I've got a little modifying to do this evening: core show function CALLERID livonia*CLI -= Info about function 'CALLERID' =- [Syntax] CALLERID(datatype[,optional-CID]) [Synopsis] Gets or sets Caller*ID data on the channel. [Description] Gets or sets Caller*ID data on the channel. The allowable datatypes are all, name, *num*, ANI, DNID, RDNIS. Uses channel callerid by default or optional callerid, if specified. Doug The documentation is correct, but the way the check really works, is that it reads the first 3 chars and matches it to num. This means that num, number, and numnumnumIloveapplesauce would all technically match. ___ -- 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] Best QoS for Linux
We use 3Com managed gigabit switches that support QoS and priority for VoIP. 3Com Unified Gigabit Wireless PoE Switch 24 and 3Com Baseline Switch 2924-PWR Plus Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com Michelle Dupuis wrote: Spinning off from another topic...what are people using for QoS / Shaping? I'm using Wondershaper script with OK results...but I'd like better. Ideas? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
