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...ing back or recording messages. Here is an example when someone leaves a message for someone: ---------------------------------------------------------------------------- --------------------------------------------------------------------------- -- Executing [170@base-out:1] AGI("SIP/bruce.mcalister-09051118", "agi://10.7.0.136:4573?app=getvoicemailexten") in new stack -- AGI Script agi://10.7.0.136:4573?app=getvoicemailexten completed, returning 0 -- Executing [170@base-out:2] NoOp("SIP/bruce.mcalister-09051118", "Caller VoiceMail Extension = 3031") in new s...

Greetings, The software G.729 codec module from Digium has been updated for all platforms. There are x86_32 and x86_64 versions optimized for specific processors available for both Asterisk 1.6 and 1.4 for the following platforms. * Linux * Solaris 10 * FreeBSD 7.0 * FreeBSD 6.1 Changes: * For Asterisk trunk / 1.6, builds have been updated for CLI API changes. * All non-Linux

Greetings, The software G.729 codec module from Digium has been updated for all platforms. There are x86_32 and x86_64 versions optimized for specific processors available for both Asterisk 1.6 and 1.4 for the following platforms. * Linux * Solaris 10 * FreeBSD 7.0 * FreeBSD 6.1 Changes: * For Asterisk trunk / 1.6, builds have been updated for CLI API changes. * All non-Linux

Hello, we are using hardware array and its vendor recommends the following setting in /etc/system: set sd:sd_max_throttle = <value> or set ssd:ssd_max_throttle = <value> Is it possible to monitor *somehow* whether the variable becomes sort of bottleneck ? Or how its value influences io traffic ? Regards przemol

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi All, Is it at all possible to have the "maxmessage" setting on per user/mailbox value? We have a requirement whereby we want the global maxmessage setting to be 180 seconds per mail box, however, we would like to have certain users to be able to store longer voice mail messages. Is this at all doable in asterisk? Thanks Bruce

Hi All, I have been trying to get the Solaris version of the G729 codec to work with asterisk 1.2.17 and 1.2.22. However, I come up against the very same error every time I try to install it. Has anyone out there seen this error, taken from the asterisk console straight from startup: [codec_g729a.so] => (Annex A/B (floating point) G.729 Codec (optimized for i386)) Jul 19 14:11:23

...nk. I would much rather test a release version, and as such, need to wait for the release of 1.4.12, my question is, do we have a guestimate on when it will be released, 1 week, 2 weeks, a month? Thanks Bruce -------------- next part -------------- A non-text attachment was scrubbed... Name: bruce.mcalister.vcf Type: text/x-vcard Size: 264 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070924/405cad9d/attachment.vcf

Hi All, Does anyone know what the current status is of the G729 codec on Solaris? According to the following link: http://www.asteriskvoipnews.com/asterisk_releases/_digium_g729_codec_now_available_forsolarissparc.html there is a version available for SPARC processor's. However, I have just had a quick look around Digium's FTP server and cannot seem to find these codecs (supported or

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Hi All, I'd just like to verify what the correct operation of the timeout parameter is for the dial application. I'm not sure if I've encountered a bug or a configuration issue. When a sip phone is not responding to invites on an outbound call, the dial application still waits the duration of timeout before continuing with dialplan execution. I was under the impression that app_dial

...oakimsen) > 16. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. > (Andrew Joakimsen) > 17. Re: Park() help, extension not heard (Rob) > 18. pickupchan without bristuffed version? (Stefan Guenther) > 19. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. > (Bruce McAlister) > 20. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. > (Thomas Kenyon) > 21. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. > (Andrew Joakimsen) > 22. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. > (Thomas Kenyon) > > > ----------------...

...ction (Jared Smith) > 12. Re: Upgrade Procedure (Nitesh Divecha) > 13. Re: Gtalk/Jabber connect issues in 1.4.8 (Bruce Ferrell) > 14. Re: Upgrade Procedure (Jared Smith) > 15. Re: 1.4.X howto disable able xpp with ./configure (Tzafrir Cohen) > 16. Re: G729 copy protection (Bruce McAlister) > 17. Re: G729 copy protection (Bruce McAlister) > 18. PRI Card (mail-lists) > 19. Re: G729 copy protection (Jared Smith) > 20. Re: how to use call transfer (Gordon Henderson) > 21. Re: New book "Asterisk Cookbook" any good? (Kristian Kielhofner) > 22. Re: PRI Car...

Hi All, I have a quick question with regards the pbx_lua module. Would the lua dialplan have direct access to the odbc configuration within Asterisk, those odbc connections/dsn's that are defined in res_odbc.conf/extconfig.conf/cdr.conf? Thanks Bruce

Hello all, Does anybody know if is it possible to install dahdi on solaris 10? I've only found a zaptel modified code for solaris at solarisvoip site. I'd appreciate any comment or experience about asterisk + dahdi/zaptel on solaris.. Best regards, Caio

Hi All, I have a really strange issue occuring where if I run "show dialplan" or "dialplan show" or "dialplan show parkedcalls", then asterisk dumps core. It only appears to happen with contexts that are created within res_features. I am able to display all my other dialplans, but, every time I try to just do a normal "dialplan show" asterisk core dumps

...Sorry I wasn't up earlier in the morning to explain my question, but the rest of the list did for me. I was seeing two messages _every_ time you posted one. Like I said, I was just curious if that could have been the reason; it would have been an easy fix ;) No offense intended. Bruce McAlister wrote: > Mojo with Horan & Company, LLC wrote: >> Sorry that this is unrelated but, Bruce, do you double-click to send >> your messages? Just curious. >> > > Sorry that this is unrelated but, Mojo with Horan, do you wake up each > morning and think of a meaningf...

Asterisk 1.2.24 seems to crash repeatedly under any substantial call load (and sometimes without a substantial call load - just one SIP leg is enough to do it) when using the G.729 pre-compiled binaries from: http://asterisk.hosting.lv/ As per: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing Time to crash is variable, but seems to require at least an hour of production performance

Hi All, I am trying to build chan_h323 for use with asterisk 1.4.18 on Solaris 10. When I compile asterisk, the build fails at chan_h323 with: ---------------------------------------------------------------------- chan_h323.c: In function `reload_config': chan_h323.c:2863: error: `IPTOS_MINCOST' undeclared (first use in this function) chan_h323.c:2863: error: (Each undeclared

Hi All, We are currently evaluating the confbridge application while we prepare to upgrade our environment to asterisk v1.6.2.x. We have run in to two issues using it to kick/mute participants in a bridge and would like to ask for the experience of others running the application for any work-arounds. Firstly for kicking participants, would it be possible to use the softhangup application

Hi all, I have 2 quick question regarding the file location and post record command of the recording using automixmon in features.conf. With the normal monitor/mixmonitor applications you can change the location of where the recordings will be stored, by changing the MONITOR_FILENAME variable. I tried changing the TOUCH_MIXMONITOR_OUTPUT variable to include a path but it sill puts the recorded