http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user hourly 1 1901-01-01T00:00+00:00 http://gmane.org/img/gmane-25t.png http://gmane.org http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220290 _______________________________________________ OCG Technical Support 2008-12-04T04:14:22 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220289 _______________________________________________ Jason Aarons (US 2008-12-04T02:44:02 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220288 _______________________________________________ OCG Technical Support 2008-12-04T02:14:37 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220287 Yeah, it would be best, but i'm really not aware of anything like that. Perhaps app_jack in 1.6 could do something (but don't ask me how, i've just heard rumors of it). The hold/background stuff is not my field, i just spitted out ideas of "how i would solve it". I looked at available commands, and if you say MusicOnHold doesn't stop, you have to terminate it somehow. Regards, Atis Atis Lezdins 2008-12-04T02:05:22 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220286 _______________________________________________ hakem Ta 2008-12-04T01:41:32 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220285 Do you thik theres no chance to do it directly from dialplan like if I use PlayTones, and the call will follow to the next line on the dialplan? I think that the best solution would be make a play musiconhold but not wait indefinitely, something like StartMOHAsync and StopMOHAsync. What do you think? Thanks for your solution. -----Original Message----- From: asterisk-users-bounces< at >lists.digium.com [mailto:asterisk-users-bounces< at >lists.digium.com] On Behalf Of Atis Lezdins Sent: miércoles, 03 de diciembre de 2008 10:31 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Parking calls On Thu, Dec 4, 2008 at 1:25 AM, Sebastian <scgm< at >adinet.com.uy> wrote: Ok, a simplified sample (i used PHP because i use it daily, but any language is good): context incoming { _X. => { Answer(); System(channel-waiting.php ${CHANNEL}); MusicOnHold(); } } context continue { _X. => { // you reached your condition Playback(tt-monkeys); Dial(SIP/something) Sebastian 2008-12-04T01:30:45 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220284 -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I'm have a bit of a problem with temporary greetings. I'm using 1.6.0.1 with IMAP storage. If I go into comedian mail and record a temporary greeting, I get a Message [0] on my IMAP server. In addition, I get /var/spool/asterisk/voicemail/default/134/temp.[gsm|wav|WAV] files. I have imapgreetings=no in the voicemail configuration file, as well as temporary greeting notification. If I go into commedian mail and delete my temporary greeting, Alison tells me that the temporary greeting has been deleted. I then am immediately told that I have a temporary greeting set. The files are still in the spool directory and Message [0] is still in my mailbox. If I delete Message [0], I still get the notification. If I delete the files in /var/spool/asterisk/voicemail/defaults/134/, I no longer get the notification. It appears as though * is looking for the temporary greeting on the local box, which is what I'd expect because of my configuration option. It also a Barry L. Kline 2008-12-04T01:22:09 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220283 Installing 1.4.23-rc2, I actually looked at the startup and saw this warning: WARNING[10730]: loader.c:359 load_dynamic_module: Error loading module 'app_directory.so': /usr/lib/libc-client.so.2007: undefined symbol: mm_dlog I'm running Fedora Core 9, with libc-client 2007d. googling didn't help, so what's the problem? Do I need a more recent (different) libc-client? sean _______________________________________________ sean darcy 2008-12-04T00:36:34 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220282 Ok, a simplified sample (i used PHP because i use it daily, but any language is good): context incoming { _X. => { Answer(); System(channel-waiting.php ${CHANNEL}); MusicOnHold(); } } context continue { _X. => { // you reached your condition Playback(tt-monkeys); Dial(SIP/something); } } then a channel-waiting.php would store ${CHANNEL} name somewhere in database. Then, assuming you can execute some code WHEN you change the database value you wanted to monitor in loop, you launch a script that sends AMI Redirect action. For example: <? $channel = 'SIP/123-abc'; // retrieved from DB where set by channel-waiting.php require "phpagi-asmanager.php"; $as = new AGI_AsteriskManager(); $res=$as->connect("localhost","username","password"); if($res==FALSE) { echo "Connection failed.\n"; } $res=$as->send_request("Redirect", array( "Channel"=>$channel, "Context"=>"continue", "Exten"=>"123", "Priority"=>"1" ) ); if($as->resp_is_success($res)){ echo "it worked!"; } Atis Lezdins 2008-12-04T00:30:58 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220281 Yep, those are fine and as I say, it does actually park the call because I can hang up and type 701 and get the call back, but my only problem is it hangs up immediately instead of playing the announcement. on Wednesday 12/03/2008 "Eric \"ManxPower\" Wieling"(eric< at >fnords.org) wrote > By "legacy phone" I assume you have an analog card connected to your > Asterisk server. I've not used analog phones with Asterisk in many > years, but IIRC you need transfer=yes and threewaycalling=yes in > zapata.conf/chan_dhadi.conf. You would then do a 2nd flash to complete > the transfer. On Polyom phones you do Transfer button/dial number/hear > parking slot/Transfer button again/Hang up. > > John covici wrote: > > I do the following from the legacy phone: hit theflash and get a > > dialtone from the call, dial 70, the call is parked, but hangs up from > > me immediately -- isn't this an attended transfer? > > > > on Wednesday 12/03/2008 "Eric \"ManxPower\" Wieling"(eric< at >fnords.org) wrote > > > John covici 2008-12-04T00:22:49 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220280 By "legacy phone" I assume you have an analog card connected to your Asterisk server. I've not used analog phones with Asterisk in many years, but IIRC you need transfer=yes and threewaycalling=yes in zapata.conf/chan_dhadi.conf. You would then do a 2nd flash to complete the transfer. On Polyom phones you do Transfer button/dial number/hear parking slot/Transfer button again/Hang up. John covici wrote: Eric "ManxPower" Wieling 2008-12-03T23:33:29 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220279 I don't understand how can I solve my situation with this -----Original Message----- From: asterisk-users-bounces< at >lists.digium.com [mailto:asterisk-users-bounces< at >lists.digium.com] On Behalf Of Atis Lezdins Sent: miércoles, 03 de diciembre de 2008 03:48 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Parking calls On Wed, Dec 3, 2008 at 7:27 PM, Sebastian <scgm< at >adinet.com.uy> wrote: further could follow AMI action Redirect - http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Redirect Of course you would need some script to send this action, but as long as you control writes to database it shouldn't be a problem. All you need is to store ${CHANNEL} name of current channel before entering MusicOnHold(). Also you could take a look at GROUP_COUNT function, perhaps it in some way can help you :) Regards, Atis Sebastian 2008-12-03T23:25:06 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220278 Sounds like you have some kind of call limit on your legacy pbx side. You'll need to get a consultant for your legacy pbx to tell you why 200 is a magic number. I've run into all kinds of problems with getting asterisk to talk to Inter-tel. Found all kinds of buried limits, which weren't documented, but we discovered when the Inter-Tel would act up when a certain number of calls would go into that part of the system. The worst limits would crash large parts of the Inter-Tel and drop all calls. Not a very nice way to discover bugs in a proprietary PBX. Be glad you at least get a busy, which you can check for with Asterisk, and route around. Our solution has been to add hardware to the Inter-Tel only when absolutely necessary, and do everything possible to keep calls in Asterisk and only send those calls into Inter-Tel after screening with IVRs, making them queue, etc. I think a similar strategy is what you should consider. And of course, our overall strategy is to dump the Inter-Tel as soon as possible, whi David Backeberg 2008-12-03T23:05:17 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220277 Ironically, much of the disagreement comes from everyone being *right*. Seriously. Philisophically, Asterisk is a tool chest, not a true product. As an analogy, if one wants to sit, one can either buy a chair or get a saw/hammer/nails/lumber and build one. Both will do the job. Asterisk is more like the saw/hammer/nails/lumber. Two wit: Andrew: you are right. For folks wanting a simple AMA-style billing CDR, it would be ideal if we leave that in the system as is. Even if it is broken for many other uses of Asterisk. I don't need it myself, but I see your point. regs: you are right. For folks wanting more advanced billing or intercarrier compensation, a standard CDR isn't going to cut it. We need to pull in N events, tied together by some kind of unique ID. Customized programming would then do the analysis and bill generation. Steve's system should help make that custom programming easier. Grey man: you are right. The direction of a call leg is easy to determine from the point of view of aste JD 2008-12-03T22:47:18 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220276 _______________________________________________ David fire 2008-12-03T22:46:24 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220275 ----- regs< at >kinetix.gr escribió: Here is a small chan_h323.so install guide: http://www.ecualug.org/?q=2008/04/18/comos/asterisk_14_agregando_soporte_h323_chan_h323so_en_asterisk_14 Saludos, Guillermo V. Salas 2008-12-03T22:42:15 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220274 _______________________________________________ David fire 2008-12-03T22:34:49 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220273 _______________________________________________ regs< at >kinetix.gr 2008-12-03T22:10:27 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220272 The Asterisk.org development team has released Asterisk version 1.6.0.3-rc1. This release is available for immediate download from http://downloads.digium.com/. This release candidate follows on the recent (broken) release of 1.6.0.2 with multiple fixes. This release also marks the first time that we are creating release candidates for bugfix releases in the 1.6 branch. For a full list of the changes in this release, please see the ChangeLog: http://svn.digium.com/view/asterisk/tags/1.6.0.3-rc1/ChangeLog?view=markup Thank you for your continued support of Asterisk! _______________________________________________ Asterisk Team 2008-12-03T21:30:49 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220271 _______________________________________________ Geraldo Coelho 2008-12-03T21:17:58 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220270 _______________________________________________ Tony Nichols 2008-12-03T20:54:21 Search the mailing list at Gmane query http://search.gmane.org/?group=$group=gmane.comp.telephony.pbx.asterisk.user