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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

...I am having difficulty getting incoming PSTN calls working. I have set up an account with a third party provider. In my system, the user register with SER and use Asterisk for PSTN access, voicemail etc My provider told me to change my sip.conf as follows register => username:password@sip.blueface.ie/2093 ; To receive incoming calls specify this block and replace "yourcontext" for your dial plan. [blueface-in] type=peer host=sip.blueface.ie context=incomingpstn And then in my extensions.conf to have something similar to the following (or however I wanted to...

...es are occurring. Many thanks, Aisling. ;sip.conf [general] bindport=5064 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=alaw allow=gsm srvlookup=yes canreinvite=no; autocreatepeer=yes nat=yes dtmfmode=info insecure=very registerattempts=0 register => username:password@sip.blueface.ie/1234 ;To receive incoming calls specify this and replace "yourcontext-pstn" for your dial plan [blueface-in] type=peer host=sip.blueface.ie context=pstn [1234] type=friend username=1234 canreinvite=no context=pstn insecure=very ;callerid= "Ais" <1234> h...

...I am having difficulty getting incoming PSTN calls working. I have set up an account with a third party provider. In my system, the user register with SER and use Asterisk for PSTN access, voicemail etc My provider told me to change my sip.conf as follows register => username:password@sip.blueface.ie/2093 ; To receive incoming calls specify this block and replace "yourcontext" for your dial plan. [blueface-in] type=peer host=sip.blueface.ie context=incomingpstn And then in my extensions.conf to have something similar to the following (or however I wanted to...

...N calls working. I have >>set up an account with a third party provider. In my system, the user >>register with SER and use Asterisk for PSTN access, voicemail etc >> >>My provider told me to change my sip.conf as follows >> >>register => username:password@sip.blueface.ie/2093 >> >>; To receive incoming calls specify this block and replace >>"yourcontext" for your dial plan. >>[blueface-in] >>type=peer >>host=sip.blueface.ie >>context=incomingpstn >> >>And then in my extensions....

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

...allationen > Beratung Support > Voice-over-IP-Loesungen > ******************************************** > > > > > ------------------------------ > > Message: 19 > Date: Tue, 15 Jan 2008 09:01:25 +0000 > From: Bruce McAlister <bruce.mcalister at blueface.ie> > Subject: Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk > 1.2.x. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <478C7665.1000404 at blueface.ie> > Content-Type: text/plain; cha...

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

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

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 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

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

Hi All, I read rumors of a potential Asterisk 1.4.12 release for last week. I would like to start testing Asterisk 1.4 on Solaris, but, the fix for the segfault in res_features is only in the current development trunk. 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

-----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'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

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

Hi, Has anybody come across a situation where they were unable to register with Asterisk through a SIP stateless proxy server? I'm getting an error: "403 Authentication user name does not match account name" As far as I can tell the requests reaching Asterisk with and without the proxy are identical excepting the IP address the REGISTER request is coming from and the Via header