Re: [asterisk-users] twice normal beep before busy tone ??
Matt wrote: -- What this means is that if the call is busy, it will play busy tones, if the call is ringing it will play ringing, congestion, congestion etc. The reason you are hearing silence is that Asterisk doesn't know what the status of the call is before that. The cell phone provider will likely take up to 3 seconds to tell your machine what is happening with the call. If you use the 'r' option then it will play ringing tones even if the phone is busy. well that means, that if I have a bad phone line (meaning poor quality) and I remove the r, I will definitely have silence, when I call cell phone numbers and may have silence on calls to normal phones. If I leave the r in the dial string, this removes the silence and adds the ring tone, with the disadvantage that I will even hear the ring tone on calls to busy numbers. If that is true, the whole problem is related to the quality of the phone line, which prevents asterisk from getting the right status fast enough. Regards, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ -- 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] Phishing attempt
Matt Riddell wrote: On 6/11/2008 8:37 p.m., Thomas Kenyon wrote: Jeff LaCoursiere wrote: I didn't realize only 4% of the world's population lived in North America! Learn something every day. Sorry that was my bedtime maths, the figure is just over 4.5%. 4.5611893661578069635904176186202% To be slightly more accurate :) Or to be outright pedantic 4.5380853065046927068273204778542%. According to figures from the US census bureau for figures projected as being the start of this month and 8:39 GMT this morning respectively. World: 6,733,867,928 USA:305,588,671 According to google's figures (July 2007 est.) USA: 301,139,947 World: 6,602,224,175 Title: Census Bureau Home Page US Census Bureau Skip this top of page navigation FAQs Subjects A to Z Help SEARCH: Skip this left side navigation New on the Site Data Tools American FactFinder [EMAIL PROTECTED] Catalog Publications Are You in a Survey? About the Bureau Regional Offices Doing Business with Us Related Sites Skip this center section 2010 Census NewsBecome a Census Taker American Community Survey Census 2000 People Households Estimates Projections Housing Income | StateMedianIncome Poverty HealthInsurance International Genealogy More Business Industry Economic Census GetHelpwithYourForm EconomicIndicators NAICS Survey of Business Owners Government E-Stats ForeignTrade|ExportCodes LocalEmployment Dynamics More Geography Maps TIGER Gazetteer More Newsroom Releases Facts For Features MinorityLinks BroadcastPhotoServices Embargo/News Release Subscription More Special Topics Census Bureau Data and Emergency Preparedness CensusCalendar Training ForTeachers Students StatisticalAbstract FedStats USA.gov Skip this left side navigation Data Finders Population Clocks U.S. 305,588,703 World 6,735,039,780 08:43 GMT (EST+5) Nov 06, 2008 Population Finder Enable _javascript_ for access to this tool or see data available from the American Community Survey. city/ town, county, or zip or state Select a state Alabama Alaska Arizona Arkansas California Colorado Connecticut Delaware District of Columbia Florida Georgia Hawaii Idaho Illinois Indiana Iowa Kansas Kentucky Louisiana Maine Maryland Massachusetts Michigan Minnesota Mississippi Missouri Montana Nebraska Nevada New Hampshire New Jersey New Mexico New York North Carolina North Dakota Ohio Oklahoma Oregon Pennsylvania Puerto Rico Rhode Island South Carolina South Dakota Tennessee Texas Utah Vermont Virginia Washington West Virginia Wisconsin Wyoming Find An Area Profile with QuickFacts For the following combo box, to make a selection, press enter then alt plus down arrow and use the up and down arrows. Select a state to begin Select a state Alabama Alaska Arizona Arkansas California
[asterisk-users] asterisk.conf ==== maxload
Dear list anyone know wich is the limit of maxload into asterisk.conf ? Also the meaning is related to RAM ? or CPU ? Regards Andrea ___ -- 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] Phishing attempt
On Thu, Nov 06, 2008 at 08:42:51AM +, Thomas Kenyon wrote: Matt Riddell wrote: On 6/11/2008 8:37 p.m., Thomas Kenyon wrote: Jeff LaCoursiere wrote: I didn't realize only 4% of the world's population lived in North America! Learn something every day. Sorry that was my bedtime maths, the figure is just over 4.5%. 4.5611893661578069635904176186202% To be slightly more accurate :) Or to be outright pedantic 4.5380853065046927068273204778542%. According to figures from the US census bureau for figures projected as being the start of this month and 8:39 GMT this morning respectively. World:6,733,867,928 USA: 305,588,671 This omits Canada. Not to mention not all of the US is in North America. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phishing attempt
Thomas Kenyon wrote: Or to be outright pedantic 4.5380853065046927068273204778542%. I apologise for attaching the files, It was unintentional. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and rawplayer
BJ Weschke wrote: Ade Vickers wrote: -Original Message- Hi Folks, I'm using the rawplayer program to provide my music-on-hold, and it works very well, with one small drawback: every time I reset Asterisk, for any reason, the MoH resets to the beginning of the list. Since MoH isn't used that often, it basically means the same track is played over over again... What I'd like to do is have rawplayer continuously playing away in the background, even if it's playing to itself only, so there's an excellent chance that any caller who will be given the pleasure of my MoH choices, will get a different tune to the one s/he heard last time... This would probably involve some kind of IPC named pipe or other inter process method of getting the data from pt A to pt B to work. While technically possible, it's not a trivial amount of work to get it going in the codebase. You might be better off with something like streaming MP3 over http or something else like that if you're looking for something with no code modifications. Hm, I was ideally looking for something with no code modifications; e.g. a phantom channel which simply played music to itself, setup when Asterisk starts, or even with manual intervention (e.g. I dial a number, and rawplayer starts up). Are you really resetting Asterisk that much that this becomes a problem? If so, why? My Asterisk install is mainly used for inter-office communications, allowing the Spanish branch to use the UK landline, and testing/experimentation. As such, I frequently do things which require a full restart, or I get it tangled up to the point where it needs a restart. The hold music rarely plays, but because rawplayer always picks the files in the same order, it's almost always track 1 that's playing when I *do* put someone on hold, or whatever; I'd prefer it to be a random start point. Cheers! Ade. ___ -- 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] HD Voice conference Friday Nov 7th @ 12 Noon EST
Hi, This week's VoIP Users Conference will be tie the Talkshoe PSTN/SIP ULAW conference bridge to the ZipDX.com G.722-capable bridge. It may be a little crazy, but I'm looking forward to getting some explanations from David Frankel about the effects of wide band (or as Polycom calls it, HD Voice) on conferencing. David can probably answer any technical questions you may have about WB audio and maybe share some of the tweaks they do to keep their service working efficiently. I know for example that they have done a lot of experimentation with jitter buffers and such to get the lag down as low as possible. As usual, the conference is open to all, your questions and shared experiences are welcome. Recordings made from both bridges will be made available and if possible, some samples of technical data of the wideband end might be viewable. That would come from the ZipDX end, not mine. I've invited Dave Nelsen, CEO of Talkshoe who has worked in the telcom industry for years and if Dave has any time to help us get info from their Talkshoe bridge, that would be interesting too. I think this might be a fun, if slightly chaotic session so be there live if you can. If you want to try to hook up your asterisk server as a client on ZipDX, write me or mgraves and we'll get that info to you. See you Friday! 9AM PST, 10 MST, 11 CST, 12 EST, 17:00 UTC 5PM UK/Portugal, 18h Fr/Sp/It 19h Tel Aviv Call info: http://bit.ly/voip IRC: irc.freenode.net #voip-users-conference PSTN: (724) 444-7444 DTMF 22622# 1# http://food4wine.ning.com forums, blogs, conference agenda ___ -- 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] Spam from DIDForSale [EMAIL PROTECTED]
didforsale.com have just sent me SPAM to the email address I use here. What a bunch of stupid fuckwits. How to get a 100% cast-iron guarantee that I'll never used their services. Morons. Gordon ___ -- 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] TDM400 with FXS some handsets not ringing
On Thu, Nov 06, 2008 at 07:26:06AM +, Gordon Henderson wrote: Are you using a ring-adapter for the UK that includes the capacitor to put the ringing current on pin 3? Something like this: http://www.voipon.co.uk/rj11-adaptor-with-ring-capacitor-p-278.html Or take the output of the TDM card and punch it into a BT master socket and take the phones off that. A lot of modern (ie. cheap import) phones don't need it, but some still do. My refurb. 746 needs this. (1960's rotary dial) Gordon Of course, the UK ringer! Why didn't I think of that? Doh! Thanks Gordon, I'll order a pair. Mike. ___ -- 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] Spam from DIDForSale [EMAIL PROTECTED]
Gordon Henderson wrote: didforsale.com have just sent me SPAM to the email address I use here. What a bunch of stupid fuckwits. How to get a 100% cast-iron guarantee that I'll never used their services. Morons. Likewise. Bails Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner at Circlemail and is believed to be clean. ___ -- 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] ExtenSpy? am I doing it correctly?
Hi Steve. I'm still trying the same because I'm interested in the subject. For what I can understand the ExtenSpy application is working properly if the selected extension receives a call. Seems not working, instead, if the selected extension originates the call. My actual setup is like that: Ext12(Soggiorno) == Ext13(Camera) ^ | Ext911- ExtSpy(12) Here is the log when the 13 calls the 12 and 911 is called by an other phone (StudioAV): -- Executing [EMAIL PROTECTED]:1] Dial(SIP/Camera-08231e60, SIP/Soggiorno) in new stack -- Called Soggiorno -- SIP/Soggiorno-082560f8 is ringing -- SIP/Soggiorno-082560f8 answered SIP/Camera-08231e60 -- Packet2Packet bridging SIP/Camera-08231e60 and SIP/Soggiorno-082560f8 -- Executing [EMAIL PROTECTED]:1] ExtenSpy(SIP/StudioAV-0822f350, 12) in new stack -- SIP/StudioAV-0822f350 Playing 'beep' (language 'it') -- SIP/StudioAV-0822f350 Playing 'spy-sip' (language 'it') == Spying on channel SIP/Camera-08231e60 Unfortunately, in the opposite direction: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/Soggiorno-0822f350, SIP/Camera) in new stack -- Called Camera -- SIP/Camera-08231e60 is ringing -- SIP/Camera-08231e60 answered SIP/Soggiorno-0822f350 -- Packet2Packet bridging SIP/Soggiorno-0822f350 and SIP/Camera-08231e60 -- Executing [EMAIL PROTECTED]:1] ExtenSpy(SIP/StudioAV-082560f8, 12) in new stack -- SIP/StudioAV-082560f8 Playing 'beep' (language 'it') == Spawn extension (from-sip, 911, 1) exited non-zero on 'SIP/StudioAV-082560f8' == Spawn extension (from-sip, 13, 1) exited non-zero on 'SIP/Soggiorno-0822f350' The application ExtSpy seems to hang just before playing the 'spy-sip' and I can't hear anything coming from the selected extension. I'm using Asterisk version Asterisk 1.4.20.1 built by root @ Gateway on a i686. Is this the correct behavior or a bug? Thank you and best regards. Marco Signorini. Steve Gladden wrote: Scratching my head and trying this. Asterisk Version: Asterisk 1.4.21.2 Tried: exten = 4771,1,ExtenSpy([EMAIL PROTECTED]) exten = 4771,2,Hangup Also tried: exten = 4771,1,Answer exten = 4771,2,ExtenSpy([EMAIL PROTECTED]) exten = 4771,3,Hangup Also tried many variations including option ,b I think most calls I make are 'bridged' extensions 4771 and 4724 are both in mbb context. Tried 'cycling' though the channels and volule * # no change. Test: 4724 places outbound or extension call. I dial 4771 from 4772 I expect to hear audio from 4724's in progress call but hear nothing. I hear a recording beep when I dial 4771. I expect to hear audio from call being made from ext. 4724 I've obviously got this wrong or the feature is not working :-) Ao far I've been unable to find much information on the net of anyone documenting a problem or a working configuration. Is there something I'm completely missing here? Thanks! Steve ___ -- 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] Errors in console for zap
Hello, Today I saw about 40 calls drop on my asterisk box. Its doing Zap to SIP w/ g729 compression. Wasnt sure what the problem is and now I'm monitoring the console and I see these strange errors. I'm running Asterisk 1.2.24 Nov 6 05:09:09 WARNING[2581]: chan_zap.c:7919 pri_fixup_principle: Can't fix up channel from 12 to 6 because 6 is already in use Nov 6 05:09:09 WARNING[2581]: chan_zap.c:9050 pri_dchannel: Hangup on bad channel 0/6 on span 1 -- Moving call from channel 5 to channel 3 Nov 6 05:09:11 WARNING[2581]: chan_zap.c:7919 pri_fixup_principle: Can't fix up channel from 5 to 3 because 3 is already in use Nov 6 05:09:11 WARNING[2581]: chan_zap.c:9118 pri_dchannel: Hangup REQ on bad channel 0/3 on span 1 !! Got reject for frame 29, retransmitting frame 29 now, updating n_r! !! Got reject for frame 29, retransmitting frame 30 now, updating n_r! -- Moving call from channel 12 to channel 6 Nov 6 05:09:13 WARNING[2581]: chan_zap.c:7919 pri_fixup_principle: Can't fix up channel from 12 to 6 because 6 is already in use Nov 6 05:09:13 WARNING[2581]: chan_zap.c:9050 pri_dchannel: Hangup on bad channel 0/6 on span 1 -- Moving call from channel 9 to channel 7 Nov 6 05:09:15 WARNING[2581]: chan_zap.c:7919 pri_fixup_principle: Can't fix up channel from 9 to 7 because 7 is already in use Nov 6 05:09:15 WARNING[2581]: chan_zap.c:9118 pri_dchannel: Hangup REQ on bad channel 0/7 on span 1 -- Moving call from channel 30 to channel 2 Nov 6 05:09:15 WARNING[2581]: chan_zap.c:7919 pri_fixup_principle: Can't fix up channel from 30 to 2 because 2 is already in use Nov 6 05:09:15 WARNING[2581]: chan_zap.c:9050 pri_dchannel: Hangup on bad channel 0/2 on span 1 -- Moving call from channel 3 to channel 11 Nov 6 05:09:16 WARNING[2581]: chan_zap.c:7919 pri_fixup_principle: Can't fix up channel from 3 to 11 because 11 is already in use Nov 6 05:09:16 WARNING[2581]: chan_zap.c:9050 pri_dchannel: Hangup on bad channel 0/11 on span 1 -- Moving call from channel 28 to channel 24 Nov 6 05:09:16 WARNING[2581]: chan_zap.c:7919 pri_fixup_principle: Can't fix up channel from 28 to 24 because 24 is already in use Nov 6 05:09:16 WARNING[2581]: chan_zap.c:9118 pri_dchannel: Hangup REQ on bad channel 0/24 on span 1 -- B-channel 0/6 restarted on span 1 == Spawn extension (default, 4066767780, 1) exited non-zero on 'Zap/6-1' -- Hungup 'Zap/6-1' -- Moving call from channel 30 to channel 2 Nov 6 05:09:19 WARNING[2581]: chan_zap.c:7919 pri_fixup_principle: Can't fix up channel from 30 to 2 because 2 is already in use Nov 6 05:09:19 WARNING[2581]: chan_zap.c:9050 pri_dchannel: Hangup on bad channel 0/2 on span 1 -- Moving call from channel 3 to channel 11 Nov 6 05:09:20 WARNING[2581]: chan_zap.c:7919 pri_fixup_principle: Can't fix up channel from 3 to 11 because 11 is already in use Nov 6 05:09:20 WARNING[2581]: chan_zap.c:9050 pri_dchannel: Hangup on bad channel 0/11 on span 1 -- Moving call from channel 20 to channel 28 Nov 6 05:09:21 WARNING[2581]: chan_zap.c:7919 pri_fixup_principle: Can't fix up channel from 20 to 28 because 28 is already in use Nov 6 05:09:21 WARNING[2581]: chan_zap.c:9118 pri_dchannel: Hangup REQ on bad channel 0/28 on span 1 -- Moving call from channel 31 to channel 9 Nov 6 05:09:21 WARNING[2581]: chan_zap.c:7919 pri_fixup_principle: Can't fix up channel from 31 to 9 because 9 is already in use Nov 6 05:09:21 WARNING[2581]: chan_zap.c:9118 pri_dchannel: Hangup REQ on bad channel 0/9 on span 1 -- Moving call from channel 11 to channel 10 Nov 6 05:09:22 WARNING[2581]: chan_zap.c:7919 pri_fixup_principle: Can't fix up channel from 11 to 10 because 10 is already in use ALSO SEEING THESE: Nov 6 05:01:46 ERROR[2577]: chan_sip.c:11577 sipsock_read: We could NOT get the channel lock for SIP/ibellvps- 0a43a900 - Call ID [EMAIL PROTECTED] Nov 6 05:01:46 ERROR[2577]: chan_sip.c:11578 sipsock_read: SIP MESSAGE JUST IGNORED: SIP/2.0 Nov 6 05:01:46 ERROR[2577]: chan_sip.c:11579 sipsock_read: BAD! BAD! BAD! -- Channel 0/15, span 4 got hangup request, cause 16 These errors only appear when I'm going over 30 channels, but I never had these issues before. Any ideas? Thanks in Advance, Jon ___ -- 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] RFC: multiple packages editing asterisk config files
Hi I'm lately bothered with the need to provide a set of Asterisk configuration files in a package that will be good for a wide range of Asterisk users. Asterisk configuration files support #include and a number of other interesting tricks, as mentioned in http://svn.digium.com/svn/asterisk/branches/1.4/doc/configuration.txt [0]. Let's start with manager.conf . Let's start with the simplest possible variant: ;;; manager.conf [general] enabled = yes bindaddr = 127.0.0.1 ; here come also a number of other remmed-out values for a human admin ; to edit ;webanbled = yes ;port = 5038 ; The default #include manager.d/*.conf ; Here the human admin can add complete sections: ;[admin] ;secret = xx ;read = all ;write = all Some Asterisk configuration interface (let's call our fictional one astcfg) can then create: /etc/asterisk/manager.d/astcfg.conf (which can also be a symlink to a directory where astcfg can actually write[1]) /etc/asterisk/manager.d/astcfg.conf [general](+) ; Those settings don't necessarily make sense. They are here to ; demonstrate how configuration parsing works bindaddr = 0.0.0.0 port = 3030 [astcfg] secret = 209348 read = all write = all Now for a more complicated example. sip.conf . sip.conf gives us a little extra pain that most users have a matching 'register =' entry. But we already learned how to do that: an extra [general](+) section[2]. As we can clearly see, it is very simple to automatically add extra sections and to add extra directives to sections. It is impossible to cancel sections and to cancel directiver (or reset to default e.g: reset the port setting so that the port in bindaddr would take effect, or vice versa) directives. I wonder if this is an actual limitation, and if so: if there is a simple way to overcome it. Problems: 1. voicemail.conf . It is accessed directly from oh so many places. Teaching all of them to respect the cool asterisk configuration files tricks (for rewriting!) is a futile attempt. Workarounds: update password with an external script, and only use the existing Asterisk interfaces to check for ovicemail authentication. Practical? What other problems would such a method have? [0] Note that this is a link to the file from 1.4. In 1.6 the file is in TeX format that is slightly less readable. Any simple way to reference directly to the relevant chapter from the generated Asterisk Book? [1] Preventing astcfg from write access to the Asterisk config files is not a real protection, because: 1. If the use of '#exec' is enabled, the astcfg can force Asterisk to run some script of its choosing e.g. to edit other configuration files. 2. If astcfg is allowed to manipulate the dialplan in any way (e.g.: originate calls, it still has complete control). This may, however, save the need to run apache as asterisk. [2] If there are too many of those, we should ask ourselves how to fit those register lines into the peer entries in order to simplify the configuration parsing. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom's lose BLF after Asterisk restart
Hi, We have an issue where Polycom's lose BLF functionality after a reboot. The only way to fix it is to reboot the Polycoms. Anyone else have this issue? We are using 1.4.18. If I run 'sip show subscriptions' all the subscriptions come back after the restart but the lights on the phones do not work. Any help would be appreciated. have the same issue with grandstreams and thomson (at least on st20XX) if we restart asterisk, phones don't renew subscriptions ... didn't search too hard, but i haven't found neither an option in asterisk nor on the phone to force resubscriptions ... cheers -- Daniele Santi .o. [EMAIL PROTECTED] ..o () ascii ribbon campaign Linux User #415108 ooo /\ www.asciiribbon.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
Re: [asterisk-users] Polycom's lose BLF after Asterisk restart
Mr Shunz wrote: Hi, We have an issue where Polycom's lose BLF functionality after a reboot. The only way to fix it is to reboot the Polycoms. Anyone else have this issue? We are using 1.4.18. If I run 'sip show subscriptions' all the subscriptions come back after the restart but the lights on the phones do not work. Any help would be appreciated. have the same issue with grandstreams and thomson (at least on st20XX) if we restart asterisk, phones don't renew subscriptions ... didn't search too hard, but i haven't found neither an option in asterisk nor on the phone to force resubscriptions ... Can you reboot the phones remotely? With snom it's quite easy to write a script to reboot all phones - you can put that in your boot scripts Ed ___ -- 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] crashes after upgrade from 1.2.16 to 1.4.21.2
Hi, After upgrading our server from asterisk 1.2.16 to 1.4.21.2 we experience crashes at random intervals with: [Nov 6 11:03:28] WARNING[12230] app_dial.c: Unable to forward voice frame read(0, unfinished ... +++ killed by SIGSEGV (core dumped) +++ Process 15755 detached On a second sister-machine with a mirror install we have the same problem. So it doesn't seem to be a hardware problem. This is with a TE410P card. Any idea? ___ -- 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] Recommend Wireless IP Phone
Use snom M3 Siemens got some problems. ___ -- 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] crashes after upgrade from 1.2.16 to 1.4.21.2
Louis-David Mitterrand wrote: Hi, After upgrading our server from asterisk 1.2.16 to 1.4.21.2 we experience crashes at random intervals with: [Nov 6 11:03:28] WARNING[12230] app_dial.c: Unable to forward voice frame Fresh install or upgrade? If it was over the top upgrade, it could be some modules from 1.2 that's causing it. Also, I'm assuming you read over the upgrade.txt. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phishing attempt
What about Mexico and Canada? Aren't they considered North America? j On Thu, 6 Nov 2008, Thomas Kenyon wrote: Matt Riddell wrote: On 6/11/2008 8:37 p.m., Thomas Kenyon wrote: Jeff LaCoursiere wrote: I didn't realize only 4% of the world's population lived in North America! Learn something every day. Sorry that was my bedtime maths, the figure is just over 4.5%. 4.5611893661578069635904176186202% To be slightly more accurate :) Or to be outright pedantic 4.5380853065046927068273204778542%. According to figures from the US census bureau for figures projected as being the start of this month and 8:39 GMT this morning respectively. World:6,733,867,928 USA: 305,588,671 According to google's figures (July 2007 est.) USA: 301,139,947 World: 6,602,224,175 ___ -- 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] Phishing attempt
My guess is that anything part of NAFTA is considered North America. It is kind of strange how a county with such a small percentage of the world population holds pretty much the entire world's markets and economy in it's hands. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Thu, Nov 6, 2008 at 7:24 AM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: What about Mexico and Canada? Aren't they considered North America? j On Thu, 6 Nov 2008, Thomas Kenyon wrote: Matt Riddell wrote: On 6/11/2008 8:37 p.m., Thomas Kenyon wrote: Jeff LaCoursiere wrote: I didn't realize only 4% of the world's population lived in North America! Learn something every day. Sorry that was my bedtime maths, the figure is just over 4.5%. 4.5611893661578069635904176186202% To be slightly more accurate :) Or to be outright pedantic 4.5380853065046927068273204778542%. According to figures from the US census bureau for figures projected as being the start of this month and 8:39 GMT this morning respectively. World:6,733,867,928 USA: 305,588,671 According to google's figures (July 2007 est.) USA: 301,139,947 World: 6,602,224,175 ___ -- 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] Recommend Wireless IP Phone
On Wed, 5 Nov 2008, Pedram M wrote: Any recommendations on good wireless SIP phones? VoIP Tech Chat did a review on the Linksys WIP 330: http://tinyurl.com/review330 and VoIP Supply has a new phone (haven't read any reviews) that has a new long-life battery. Fred Posner smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Recommend Wireless IP Phone
On Thu, Nov 6, 2008 at 4:56 AM, Pedram M [EMAIL PROTECTED] wrote: Any recommendations on good wireless SIP phones? I use a Siemens S675IP in our two person office. It performs very well, and has a built in answering machine which is of interest for us because we have several SIP accounts that are pay as you go, so no vmail. Also the S675IP (the 685 is the same plus bluetooth) is connected to our POTS line, another great advantage. All in all, I have it registered at 6 SIP providers. Battery life is fine, decent feature set, something to look into IMO. I hear there is soon to be a USA version. r ___ -- 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] tired of midget packet received warnings
Hi, When monitoring an asterisk through its iax2 port I get these warnings at the console: [Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process: midget packet received (1 of 4 min) This is triggered by the monitoring app sending a POKE to the iax port. The warning appears even without any '-v'. Is there a way to avoid these warnings? Or at least turn them off when at the console in non-verbose mode? Thanks, -- http://www.lesculturelles.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ODBCExec from Dialplan
Hi,
I'm trying to make odbcexec work with Asterisk 1.6.
I had the attached code (app_odbcexec, not the standard one) working great
with asterisk 1.2 an MSSQL Server on heavy load PBXs with no problem, I'm
trying to port this to asterisk 1.6 but I'm failing to do so.
I attach de working code in 1.2 (app_odbcexec) and my try to port it to 1.6
(app_odbcexec1.6).
Anyone can help??
Thanks
/*
* Asterisk -- A telephony toolkit for Linux.
*
* ODBC exec function
*
* Robert Hanzlik [EMAIL PROTECTED]
*
* This program is free software, distributed under the terms of
* the GNU General Public License
*
* Copyright (c) Digium
*
* Based on work by Mark Spencer and Jefferson Noxon - app_db.c
* and Brian K. West - app_dbodbc.c
*
*/
#include asterisk.h
#include asterisk/pbx.h
#include asterisk/module.h
#include asterisk/app.h
#include asterisk/channel.h
#include asterisk/config.h
#include sql.h
#include sqlext.h
#include sqltypes.h
#define AST_MODULE app_odbcexec
static char *tdesc = Database query functions for Asterisk extension logic;
static char *q_descrip =
ODBCquery(varname=query): Retrieves a value from the database query\n
and stores it in the given variable. Always returns 0. If the\n
query failes, jumps to priority n+101 if available.\n;
static char *e_descrip =
ODBCexec(query): Executes a database query. Always returns 0.\n
If the query failes, jumps to priority n+101 if available.\n;
static char *q_app = ODBCquery;
static char *e_app = ODBCexec;
static char *q_synopsis = Retrieve a value from a ODBC query;
static char *e_synopsis = Execute a ODBC query;
AST_MUTEX_DEFINE_STATIC(odbc_lock);
static SQLHENV HOdbcEnv;
static int ODBC_res; /* global ODBC Result of
Functions */
static SQLHDBC ODBC_con; /* global ODBC Connection
Handle */
static SQLHSTMT ODBC_stmt; /* global ODBC Statement Handle
*/
static char *config = odbcexec.conf;
static char *dsn = NULL, *username = NULL, *password = NULL;
static int dsn_alloc = 0, username_alloc = 0, password_alloc = 0;
static int connected = 0;
static int ast_odbcexec(const char *query, char *out, int outlen);
static int odbc_load_module(int);
static int odbc_init(void);
static int odbc_unload_module(void);
static int odbc_do_query(char *sqlcmd);
static void reconect(void);
void LogErrMsg(char * source, long rc,SQLSMALLINT HandleType,SQLHANDLE Handle);
void LogErrMsg(char * source, long rc,SQLSMALLINT HandleType,SQLHANDLE Handle)
{
SQLSMALLINT len;
SQLCHAR msg[200],buffer[200];
SQLCHAR sqlstat[10];
ast_log(LOG_ERROR, Error %s %d\n,source,rc);
SQLGetDiagRec(HandleType,Handle,1,
sqlstat, rc,msg,100,len);
ast_log(LOG_ERROR, %s (%d)\n,msg,rc);
}
static int odbcexec_exec(struct ast_channel *chan, void *data)
{
int arglen, res;
char *argv;
arglen = strlen (data);
argv = alloca (arglen + 1);
if (!argv) /* Why would this fail? */
{
ast_log (LOG_DEBUG, Memory allocation failed\n);
return 0;
}
memcpy (argv, data, arglen + 1);
ast_verb (3, odbcexec: query=%s\n, argv);
ast_mutex_lock(odbc_lock);
res = odbc_do_query(argv);
ast_mutex_unlock(odbc_lock);
if(res==-1) {
ast_verb (3, odbcexec: Query failed.\n);
/* Send the call to n+101 priority, where n is the current
priority */
if (ast_exists_extension (chan, chan-context, chan-exten,
chan-priority + 101, chan-cid.cid_num))
chan-priority += 100;
}
return 0;
}
static int odbcexec_query(struct ast_channel *chan, void *data)
{
int arglen;
char *argv, *varname, *query;
char dbresult[256];
arglen = strlen (data);
argv = alloca (arglen + 1);
if (!argv) /* Why would this fail? */
{
ast_log (LOG_DEBUG, Memory allocation failed\n);
return 0;
}
memcpy (argv, data, arglen + 1);
if (strchr (argv, '='))
{
varname = strsep (argv, =);
query = strsep (argv, \0);
if (!varname || !query)
{
ast_log (LOG_DEBUG, Ignoring; Syntax error in
argument\n);
return 0;
}
ast_verb (3, odbcquery: varname=%s, query=%s\n,
varname, query);
if (!ast_odbcexec (query, dbresult, sizeof (dbresult) - 1))
{
pbx_builtin_setvar_helper (chan, varname, dbresult);
ast_verb (3, odbcquery: set variable %s to
%s\n, varname, dbresult);
Re: [asterisk-users] Recommend Wireless IP Phone
On Thu, 6 Nov 2008 15:01:09 +0100, randulo wrote: On Thu, Nov 6, 2008 at 4:56 AM, Pedram M [EMAIL PROTECTED] wrote: Any recommendations on good wireless SIP phones? I use a Siemens S675IP in our two person office. It performs very well, and has a built in answering machine which is of interest for us because we have several SIP accounts that are pay as you go, so no vmail. Also the S675IP (the 685 is the same plus bluetooth) is connected to our POTS line, another great advantage. All in all, I have it registered at 6 SIP providers. Battery life is fine, decent feature set, something to look into IMO. I hear there is soon to be a USA version. I own both the snom m3 and a Siemens S685IP. Both are basically good devices, but each has its little issues. The S675/685IP supports G.722 which is great! But it has no mute button, which is a drag. Also, its less expensive. The snom m3 is smallish in the hand, or so my wife tells me. It's also a little too easy to turn the handset off by accident. However firmware development at snom is progressing nicely. Aastra and Polycom have new SIP/DECT offerings targeting SMBs. Have not had a chance to try these yet. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- 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] AEL NoOp not working
2008/11/5 Atis Lezdins [EMAIL PROTECTED] On Wed, Nov 5, 2008 at 5:28 PM, Olivier [EMAIL PROTECTED] wrote: 2008/11/5 Atis Lezdins [EMAIL PROTECTED] On Wed, Nov 5, 2008 at 12:39 PM, Olivier [EMAIL PROTECTED] wrote: Hi, I've new to http://www.voip-info.org/wiki/view/Asterisk+AEL2 I'm using NoOp and Verbose functions inside extensions.ael. Strangely, NoOp is not printing anything in Asterisk console while Verbose is working. Am I missing something obvious ? Hi, NoOp is not outputting anything, it's just does nothing, however you should still be able to see Executing NoOp(blablabla) in console, as it's a command. Yes, that's the point : I don't see anything in console (I wasn't expecting anything else to happen). Strange ... Ok, i played with this, and seems that Executing ... lines are shown in CLI only on verbose level 3 or higher. So, either start asterisk with asterisk -vvv or issue core set verbose 3 in CLI. Yes, you're right : NoOp needs verbosity of 3 and above. Thanks for helping. The surprising thing is that AEL Verbose prints output whatever the verbosity level is (even with 0). Would you qualify this as normal ? Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC: multiple packages editing asterisk config files
On Thu, Nov 06, 2008 at 12:57:15PM +0200, Tzafrir Cohen wrote: Hi I'm lately bothered with the need to provide a set of Asterisk configuration files in a package that will be good for a wide range of Asterisk users. Asterisk configuration files support #include and a number of other interesting tricks, as mentioned in http://svn.digium.com/svn/asterisk/branches/1.4/doc/configuration.txt [0]. Now let's look at indications.conf . indications.conf . This is essentially a data file. Indications should mostly have been provided by an external library. Indications from libtonezone and from indications.conf are the same. The sample^Wreference indications.conf begins, however with a warning: ; NOTE: ;When adding countries to this file, please keep them in alphabetical ;order according to the 2-character country codes! I looked at main/indications.c and res/res_indications.c and fail to see the reason for this limitation. The only configurable part in this file follows: [general] country=us ; default location Which would be easy to override anyway: [general](+) country = dk There is normally never a need to remove a value. Only to add a section or fix a wrong value. Hence I would suggest to replace the default settings for indications.conf with: ; This would be /var/lib/asterisk if your distribution did not change it ; to something like /usr/share/asterisk ;-) #include $AST_DATA_DIR/configs/indications.countries.conf [general] ;country = us -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommend Wireless IP Phone
On Thu, Nov 6, 2008 at 3:21 PM, Michael Graves [EMAIL PROTECTED] wrote: The S675/685IP supports G.722 which is great! But it has no mute button, which is a drag. Also, its less expensive. Truth be told, I hate that there's no mute button. Also, the handset isn't good enough to make a huge quality difference in g722, at least to my ear. I do believe it has a high bang for the buck value, though. /r ___ -- 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] tired of midget packet received warnings
Louis-David Mitterrand wrote: When monitoring an asterisk through its iax2 port I get these warnings at the console: [Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process: midget packet received (1 of 4 min) This is triggered by the monitoring app sending a POKE to the iax port. The warning appears even without any '-v'. Your monitoring app is not sending valid IAX2 packets to the server. If it was sending a true IAX2 POKE, it would be a valid packet and wouldn't generate this warning. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] ODBCExec from Dialplan
On Thu, 2008-11-06 at 12:16 -0200, Sebastian Gutierrez wrote: I'm trying to make odbcexec work with Asterisk 1.6. Why not just use the functionality of func_odbc already built into Asterisk 1.6? Is there something you gain by going with odbcexec that func_odbc doesn't provide? Also, just as a word of caution, please don't send a message to the list by replying to another thread in the list. It's not good etiquette. This causes your message to appear inside of another thread, and can be confusing to people reading the mailing list. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tired of midget packet received warnings
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote: Louis-David Mitterrand wrote: When monitoring an asterisk through its iax2 port I get these warnings at the console: [Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process: midget packet received (1 of 4 min) This is triggered by the monitoring app sending a POKE to the iax port. The warning appears even without any '-v'. Your monitoring app is not sending valid IAX2 packets to the server. If it was sending a true IAX2 POKE, it would be a valid packet and wouldn't generate this warning. Hi, Is POKE a generic udp thing or specific to iax? In the former case I'll probably be able to submit a patch to wmnetmon (great dockable applet I'm using). Thanks, -- http://www.lesculturelles.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC: multiple packages editing asterisk config files
On Thu, Nov 06, 2008 at 12:57:15PM +0200, Tzafrir Cohen wrote:
Hi
I'm lately bothered with the need to provide a set of Asterisk
configuration files in a package that will be good for a wide range of
Asterisk users.
Asterisk configuration files support #include and a number of other
interesting tricks, as mentioned in
http://svn.digium.com/svn/asterisk/branches/1.4/doc/configuration.txt [0].
Where can we find tools to parse Asterisk configuration files?
Asterisk does not provide any library or utility to read configuration
files. The asterisk-gui uses Asterisk itself to both read and write
configuration files.
The FreeIRIS project uses a set of perl module, that are also available
in CPAN, as Asterisk::config
http://search.cpan.org/dist/Asterisk-config/
http://www.freeiris.org/
(Never tried using them)
Next file to look at is modules.conf . modules.conf is one of those
files with a single section (or maybe two or three, but many entries
in that section). Mostly the order in modules.conf doesn't count.
However I beleive that in some specific cases the order of entries there
does count.
In addition, of of the entries there are of the same two keys ('load'
and 'unload'). Hence it's imporssible to override an earlier
assignment later. This makes module.conf very unmodular (pun not
intended) by nature.
The problem with modules.conf is that a mistake in it can easily get
Asterisk to either fail to load or behave very strange.
Can someone describe to me a real-life situation where the order of
directives in modules.conf matters?
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
___
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Re: [asterisk-users] tired of midget packet received warnings
On Thursday 06 November 2008 08:53:40 Louis-David Mitterrand wrote: On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote: Louis-David Mitterrand wrote: When monitoring an asterisk through its iax2 port I get these warnings at the console: [Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process: midget packet received (1 of 4 min) This is triggered by the monitoring app sending a POKE to the iax port. The warning appears even without any '-v'. Your monitoring app is not sending valid IAX2 packets to the server. If it was sending a true IAX2 POKE, it would be a valid packet and wouldn't generate this warning. Is POKE a generic udp thing or specific to iax? In the former case I'll probably be able to submit a patch to wmnetmon (great dockable applet I'm using). It's specific to IAX. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODBCExec from Dialplan
Ok, sorry for the response on the same thread. The main thing is that with this I set the Store Procedure or Query directly on the dialplan line, is easier to configure, change, manage, etc. I also know that works great with heavy load, and it reconnects when the network goes down and up. Can you help me porting this app? I think woun`t be difficult for someone that has port other app. Thanks -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jared Smith Enviado el: Thursday, November 06, 2008 12:51 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] ODBCExec from Dialplan On Thu, 2008-11-06 at 12:16 -0200, Sebastian Gutierrez wrote: I'm trying to make odbcexec work with Asterisk 1.6. Why not just use the functionality of func_odbc already built into Asterisk 1.6? Is there something you gain by going with odbcexec that func_odbc doesn't provide? Also, just as a word of caution, please don't send a message to the list by replying to another thread in the list. It's not good etiquette. This causes your message to appear inside of another thread, and can be confusing to people reading the mailing list. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Files
Local channel will help you send your call through the dialplan. You can make all your decision there. If it answers, then the specified application will be execute. Check this example http://www.astblog.com/2008/09/18/use-the-power-of-local-channels/ David Klaverstyn wrote: I have successfully created call files and I can get Asterisk to make calls based on those files. The problem I have is that it seems you need to use a Channel for the first leg of the call file. This means I have to use either a ZAP, SIP or IAX2 channel. What I would prefer to do is send the first leg of the call to a context and extension so I can send the call using DUNDi rather than a predefined channel. Once the call has been established then is should go to context, extension so the second leg of the call can be completed. Is it possible to send the first leg of a call file to DUNDi and if not aviable send over IAX2 or then ZAP? The call files seem to be limited to a channel and not allow the first leg of the call to be decided by the path of a context, extension. ___ -- 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 Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- 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] Asterisk trunking
Hello !
I am experiencing some problems with Asterisk trunking, this is the scenario:
There are 3 servers, a DID server provider (VOIP provider) which
delegates us a bunch of DID numbers to our asterisk server number one
(I will call it AA), from which I route the calls to Asterisk server
number 2 (I will call it BB), which then terminate on phone handsets.
The trouble is, that I probably didn't do a proper routing, when I was
trying out parked calls with key #, the phone which is connected to BB
was constantly trying to connect to AA.
Here is the Dialplan configuration:
Asterisk server number one (AA)
All outgoing calls need a prefix in front of the number
[buster]
; btc trunk
exten = _9.,1,DIAL(SIP/btctrunk/${EXTEN})
exten = _9.,n,Hangup()
; soft
; caller-id SOFT
;exten = _X.,1,Set(CALLERID(num)=01${CALLERID(num)})
exten = _X.,1,Set(CALLERID(num)=0158631${CALLERID(num)})
exten = _X.,n,DIAL(SIP/softnet/01${EXTEN})
exten = _X.,n,Hangup()
; popravek 0
;exten = _0.,1,Set(CALLERID(num)=018109${CALLERID(num)})
exten = _0.,1,Set(CALLERID(num)=0158631${CALLERID(num)})
exten = _0.,n,DIAL(SIP/softnet/${EXTEN})
exten = _0.,n,Hangup()
; voicemail
; exten = 700,1,VoiceMailMain()
;exten = _018109.,1,GotoIf($[${DIALSTATUS} = BUSY]?busy:unavail)
;exten = _018109.,n(unavail),Voicemail([EMAIL PROTECTED],su)
;exten = _018109.,n,Hangup()
;exten = _018109.,n(busy),VoiceMail([EMAIL PROTECTED],sb)
;exten = _018109.,n,Hangup()
exten = _018109.,1,Answer()
exten = _018109.,n,Playback(pbx-invalid)
exten = _018109.,n,Playback(vm-goodbye)
exten = _018109.,n,Hangup()
exten = _0158631.,1,Answer()
exten = _0158631.,n,Playback(pbx-invalid)
exten = _0158631.,n,Playback(vm-goodbye)
exten = _0158631.,n,Hangup()
; parked calls
include = parkedcalls
; telprom prenos
exten = 018109235,1,Answer()
exten = 018109235,n,Dial(SIP/telprom/70,130,rtk)
exten = 018109235,n,Hangup()
exten = 015863160,1,Answer()
exten = 015863160,n,Dial(SIP/telprom/5863160,130,rtk)
exten = 015863160,n,Hangup()
exten = 015863161,1,Answer()
exten = 015863161,n,Dial(SIP/telprom/5863161,130,rtk)
exten = 015863161,n,Hangup()
exten = 015863162,1,Answer()
exten = 015863162,n,Dial(SIP/telprom/5863162,130,rtk)
exten = 015863162,n,Hangup()
exten = 015863163,1,Answer()
exten = 015863163,n,Dial(SIP/telprom/5863163,130,rtk)
exten = 015863163,n,Hangup()
exten = 015863164,1,Answer()
exten = 015863164,n,Dial(SIP/telprom/5863164,130,rtk)
exten = 015863164,n,Hangup()
exten = 015863165,1,Answer()
exten = 015863165,n,Dial(SIP/telprom/5863165,130,rtk)
exten = 015863165,n,Hangup()
exten = 015863166,1,Answer()
exten = 015863166,n,Dial(SIP/telprom/5863166,130,rtk)
exten = 015863166,n,Hangup()
exten = 015863167,1,Answer()
exten = 015863167,n,Dial(SIP/telprom/5863167,130,rtk)
exten = 015863167,n,Hangup()
exten = 015863168,1,Answer()
exten = 015863168,n,Dial(SIP/telprom/5863168,130,rtk)
exten = 015863168,n,Hangup()
exten = 015863169,1,Answer()
exten = 015863169,n,Dial(SIP/telprom/5863169,130,rtk)
exten = 015863169,n,Hangup()
exten = 015863170,1,Answer()
exten = 015863170,n,Dial(SIP/telprom/5863170,130,rtk)
exten = 015863170,n,Hangup()
exten = 015863171,1,Answer()
exten = 015863171,n,Dial(SIP/telprom/5863171,130,rtk)
exten = 015863171,n,Hangup()
; lokalni telefoni
exten = 018109222,1,Dial(SIP/222,130,rtk)
exten = 018109222,n,Hangup()
exten = 018109223,1,Dial(SIP/223,130,rtk)
exten = 018109223,n,Hangup()
exten = 018109224,1,Dial(SIP/224,130,rtk)
exten = 018109224,n,Hangup()
exten = 018109225,1,Dial(SIP/225,130,rtk)
exten = 018109225,n,Hangup()
exten = 018109226,1,Dial(SIP/226,130,rtk)
exten = 018109226,n,Hangup()
exten = 222,1,Dial(SIP/222,130,rtk)
exten = 222,n,Hangup()
exten = 223,1,Dial(SIP/223,130,rtk)
exten = 223,n,Hangup()
exten = 224,1,Dial(SIP/224,130,rtk)
exten = 224,n,Hangup()
exten = 225,1,Dial(SIP/225,130,rtk)
exten = 225,n,Hangup()
exten = 226,1,Dial(SIP/226,130,rtk)
exten = 226,n,Hangup()
---
Asterisk server number two (BB)
All phones are connected to context buster
[buster]
exten = _9.,1,Dial(SIP/btctrunk/${EXTEN})
exten = _9.,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE})
exten = _9.,n,Hangup
; soft - nova pravila
exten = _X.,1,Dial(SIP/softlink/${EXTEN})
exten = _X.,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE})
exten = _X.,n,Hangup
; mobitel
exten = _7.,1,Dial(SIP/portech/${EXTEN:1})
exten = _7.,2,Hangup
exten = _+38651.,1,Dial(SIP/portech/${EXTEN})
exten = _+38651.,2,Hangup
exten = _+38641.,1,Dial(SIP/portech/${EXTEN})
exten = _+38641.,2,Hangup
exten = _+38631.,1,Dial(SIP/portech/${EXTEN})
exten = _+38631.,2,Hangup
exten = _+38640.,1,Dial(SIP/portech/${EXTEN})
exten = _+38640.,2,Hangup
exten = _+38670.,1,Dial(SIP/portech/${EXTEN})
exten = _+38670.,2,Hangup
exten = _0038651.,1,Dial(SIP/portech/${EXTEN})
exten = _0038651.,2,Hangup
exten = _0038641.,1,Dial(SIP/portech/${EXTEN})
exten = _0038641.,2,Hangup
exten = _0038631.,1,Dial(SIP/portech/${EXTEN})
exten = _0038631.,2,Hangup
exten = _0038640.,1,Dial(SIP/portech/${EXTEN})
exten = _0038640.,2,Hangup
exten =
Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]
On Thu, 6 Nov 2008, Gordon Henderson wrote: didforsale.com have just sent me SPAM to the email address I use here. What a bunch of stupid fuckwits. How to get a 100% cast-iron guarantee that I'll never used their services. Morons. The English have such a way with words :) I keep a local archive of the last 30 days list posts. Searching for didforsale.com shows: Buy unmetered VoIP DID from DidForSale.com is the signature for Jai Rangi [EMAIL PROTECTED]. A wolf in the sheep's pen? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] twice normal beep before busy tone ??
It sounds like you have analog lines. If that is the case, the silence you experience is Asterisk sending the DTMF down the line. Asterisk collects the DTMF and when you are done dialing it retransmits those digits down the analog line. I think each digit is by default 300ms. If you are dialing 10 digits you have a delay of 3.3 seconds while Asterisk is sending the DTMF. None of the VoIP protocols have his issue. ISDN PRI or BRI also does not have this issue. Stefan Guenther wrote: Matt wrote: -- What this means is that if the call is busy, it will play busy tones, if the call is ringing it will play ringing, congestion, congestion etc. The reason you are hearing silence is that Asterisk doesn't know what the status of the call is before that. The cell phone provider will likely take up to 3 seconds to tell your machine what is happening with the call. If you use the 'r' option then it will play ringing tones even if the phone is busy. well that means, that if I have a bad phone line (meaning poor quality) and I remove the r, I will definitely have silence, when I call cell phone numbers and may have silence on calls to normal phones. If I leave the r in the dial string, this removes the silence and adds the ring tone, with the disadvantage that I will even hear the ring tone on calls to busy numbers. If that is true, the whole problem is related to the quality of the phone line, which prevents asterisk from getting the right status fast enough. Regards, Stefan -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- 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] Spam from DIDForSale [EMAIL PROTECTED]
I'm glad I'm not the only one who got that. I sent them a nasty response earlier this morning... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Thursday, November 06, 2008 11:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED] On Thu, 6 Nov 2008, Gordon Henderson wrote: didforsale.com have just sent me SPAM to the email address I use here. What a bunch of stupid fuckwits. How to get a 100% cast-iron guarantee that I'll never used their services. Morons. The English have such a way with words :) I keep a local archive of the last 30 days list posts. Searching for didforsale.com shows: Buy unmetered VoIP DID from DidForSale.com is the signature for Jai Rangi [EMAIL PROTECTED]. A wolf in the sheep's pen? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Variable Scope Question
If I have a global variable in my dialplan and I change it, does that change immediately take affect for all calls that are active? Here is my situation. The company I work for has two office groups that share a building. The two offices are separate companies but support one another and want to be able to transfer calls as if they were all on the same phone system. Each company has 4 incoming voice lines and calls on those lines should be sent to the appropriate main menu. As it stands I have a context called internal that defines all of the internal extensions for both offices then I have two virtually operator contexts, two virtually identical mainmenu contexts and two virtually identical admin contexts that allow them to record the appropriate mainmenu greetings. What I'd like to do would be to consolidate the mainmenu and operator contexts and create a CompanyA and CompanyB context that sets a variable for the appropriate company then jumps into the mainmenu or operator context carrying that value so the correct greetings are played and the correct operator extension is used. The variable would need to be one that only affects the current call and no others since there is the potential to have 4 calls coming in to each office at the same time. Any ideas on the best way to handle this? Thanks, Brent ___ -- 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] Asking again about busylevel
I sent this email a few days ago but did not see any responses to it: I am running asterisk 1.6.0.1. I have a Grandstream GXP280 phone I use for testing. In addition I register a zoiper SIP soft phone. For the Grandstream I have busylevel=1 in sip.conf. If I place a call from the GXP280 to zoiper and then put that call on hold from the zoiper side and then call GXP280's extension, asterisk indicates the phone is ringing. As the GXP280 is a single line phone it does not ring the second call. I would have expected the call to get a busy as I have busylevel set to one. I have tried setting busylevel to two as well with the same result. Can someone let me know what I should look at to see why I am not getting a busy instead of ringing? Here are the definitions in sip.conf for the GXP280 and zoiper: [dickenson] type=friend context=empl nat=yes host=dynamic secret=password callerid=Jim Dickenson 108 [EMAIL PROTECTED] [GXP280] type=friend context=empl host=dynamic secret=password callerid=GXP280 109 [EMAIL PROTECTED] busylevel=1 I am wondering if anyone has been able to get a busy when calling a SIP phone. I seem to just get a no answer returned. -- Jim Dickenson mailto:[EMAIL PROTECTED] CfMC http://www.cfmc.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]
On Thu, Nov 6, 2008 at 2:11 PM, David Gibbons [EMAIL PROTECTED]wrote: I'm glad I'm not the only one who got that. I sent them a nasty response earlier this morning... I got the same crap from them. I can't imagine anyone buying from a company that spams subscribers of a mailing list to get business. But I guess spam exists for a reason; there's always some moron out there that purchases goods from these delinquents. - Gonzalo ___ -- 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] Variable Scope Question
On Thu, Nov 6, 2008 at 6:12 PM, Brent Davidson
[EMAIL PROTECTED] wrote:
If I have a global variable in my dialplan and I change it, does that
change immediately take affect for all calls that are active?
Here is my situation. The company I work for has two office groups that
share a building. The two offices are separate companies but support
one another and want to be able to transfer calls as if they were all on
the same phone system. Each company has 4 incoming voice lines and
calls on those lines should be sent to the appropriate main menu.
As it stands I have a context called internal that defines all of the
internal extensions for both offices then I have two virtually operator
contexts, two virtually identical mainmenu contexts and two virtually
identical admin contexts that allow them to record the appropriate
mainmenu greetings.
What I'd like to do would be to consolidate the mainmenu and operator
contexts and create a CompanyA and CompanyB context that sets a variable
for the appropriate company then jumps into the mainmenu or operator
context carrying that value so the correct greetings are played and the
correct operator extension is used. The variable would need to be one
that only affects the current call and no others since there is the
potential to have 4 calls coming in to each office at the same time.
Any ideas on the best way to handle this?
Hello,
there is not a definite best way, however variable approach sounds ok.
All you need is channel variable as opposed to global variable.
Whenever call starts (internal or external), you can match mask of DID
or CallerID (for internal extensions) and just execute
Set(__company=A). Two underscores means that this variable will be
inherited in every child channel, so wherever the call will go (within
Asterisk of course) you will have variable ${company}
For more information please see http://www.voip-info.org/wiki-Asterisk+variables
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]
Gotta love this list being farmed for spammers now. I am sure they call it targeted delivery or some such nonsense. I can't wait for capitalism to completely fail, then there won't be any spam. David Gibbons wrote: I'm glad I'm not the only one who got that. I sent them a nasty response earlier this morning... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Thursday, November 06, 2008 11:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED] On Thu, 6 Nov 2008, Gordon Henderson wrote: didforsale.com have just sent me SPAM to the email address I use here. What a bunch of stupid fuckwits. How to get a 100% cast-iron guarantee that I'll never used their services. Morons. The English have such a way with words :) I keep a local archive of the last 30 days list posts. Searching for didforsale.com shows: Buy unmetered VoIP DID from DidForSale.com is the signature for Jai Rangi [EMAIL PROTECTED]. A wolf in the sheep's pen? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- 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] Variable Scope Question
On Thursday 06 November 2008 10:12:11 Brent Davidson wrote:
If I have a global variable in my dialplan and I change it, does that
change immediately take affect for all calls that are active?
Here is my situation. The company I work for has two office groups that
share a building. The two offices are separate companies but support
one another and want to be able to transfer calls as if they were all on
the same phone system. Each company has 4 incoming voice lines and
calls on those lines should be sent to the appropriate main menu.
As it stands I have a context called internal that defines all of the
internal extensions for both offices then I have two virtually operator
contexts, two virtually identical mainmenu contexts and two virtually
identical admin contexts that allow them to record the appropriate
mainmenu greetings.
What I'd like to do would be to consolidate the mainmenu and operator
contexts and create a CompanyA and CompanyB context that sets a variable
for the appropriate company then jumps into the mainmenu or operator
context carrying that value so the correct greetings are played and the
correct operator extension is used. The variable would need to be one
that only affects the current call and no others since there is the
potential to have 4 calls coming in to each office at the same time.
Any ideas on the best way to handle this?
[companyA]
exten = _X.,1,Set(company=A)
exten = _X.,n,Goto(maincontext,${EXTEN},1)
[companyB]
exten = _X.,1,Set(company=B)
exten = _X.,n,Goto(maincontext,${EXTEN},1)
--
Tilghman
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Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]
On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote: Gotta love this list being farmed for spammers now. I am sure they call it targeted delivery or some such nonsense. I can't wait for capitalism to completely fail, then there won't be any spam. Socialism has already completely failed. What should we do, go back to a barter economy? :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variable Scope Question
Tilghman Lesher wrote:
[companyA]
exten = _X.,1,Set(company=A)
exten = _X.,n,Goto(maincontext,${EXTEN},1)
[companyB]
exten = _X.,1,Set(company=B)
exten = _X.,n,Goto(maincontext,${EXTEN},1)
I should probably also mention that I am using AEL for my dialplan.
(i'm a programmer and the AEL syntax feels more natural to me) . Does
the Set command work the same way in AEL as it does in the regular
.conf file?
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Re: [asterisk-users] Recommend Wireless IP Phone
Wi-Fi SIP phones aren't limited to hot spots. I am in the process of setting up asterisk for SOHO. At present I'm not even using VoIP trunking, only LAN to stns and I intend to use Wi-Fi instead of analog cordless phone. I got the Engenius one, and it works, but I haven't played with it much. I was disappointed that it only has a single line appearance, as part of my reason for going SIP was to allow the same features like say my 941. I also got their 600 mw access point, but haven't had time to try it. My goal is to cover out 3 acre property and the 1/2 mile road to the mailbox, including mountainous terrain. Maybe I'll share more when I actually get it all put together. Wilton ___ -- 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] Spam from DIDForSale [EMAIL PROTECTED]
http://en.wikipedia.org/wiki/Jacque_Fresco A resource based economy. Greg Woods wrote: On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote: Gotta love this list being farmed for spammers now. I am sure they call it targeted delivery or some such nonsense. I can't wait for capitalism to completely fail, then there won't be any spam. Socialism has already completely failed. What should we do, go back to a barter economy? :-) ___ -- 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 -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] twice normal beep before busy tone ??
If it is 300 ms, that is way to long. I don't know any CO grade receiver that can't decode in 80 ms and some can do 40. There is also a similar size gap between digits. Is there an option to start dialing as soon as enough digits are collected to guarantee a unique route? That has been the norm in traditional PABXs for 20 or 30 years, and combined with 80 ms duration, it can generally finish by the time the user has entered the last digit. Wilton ___ -- 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] Variable Scope Question
On Thursday 06 November 2008 11:41:08 Brent Davidson wrote:
Tilghman Lesher wrote:
[companyA]
exten = _X.,1,Set(company=A)
exten = _X.,n,Goto(maincontext,${EXTEN},1)
[companyB]
exten = _X.,1,Set(company=B)
exten = _X.,n,Goto(maincontext,${EXTEN},1)
I should probably also mention that I am using AEL for my dialplan.
(i'm a programmer and the AEL syntax feels more natural to me) . Does
the Set command work the same way in AEL as it does in the regular
.conf file?
AFAIK, yes, it does.
--
Tilghman
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Re: [asterisk-users] [OT] Capitalism (was: Spam from DIDForSale [EMAIL PROTECTED])
On Thu, Nov 6, 2008 at 7:50 PM, Anthony Francis [EMAIL PROTECTED] wrote: http://en.wikipedia.org/wiki/Jacque_Fresco A resource based economy. Greg Woods wrote: On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote: Gotta love this list being farmed for spammers now. I am sure they call it targeted delivery or some such nonsense. I can't wait for capitalism to completely fail, then there won't be any spam. Socialism has already completely failed. What should we do, go back to a barter economy? :-) Thanks for interesting link :) Didn't knew any such projects exist. I recently submitted idea for Google Project 10^100 which would help implementing Resource Basec Economy (i just didn't knew that such term exists). Can't wait January 27th.. :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] twice normal beep before busy tone ??
Eric wrote: It sounds like you have analog lines. If that is the case, the silence you experience is Asterisk sending the DTMF down the line. Asterisk collects the DTMF and when you are done dialing it retransmits those digits down the analog line. I think each digit is by default 300ms. If you are dialing 10 digits you have a delay of 3.3 seconds while Asterisk is sending the DTMF. None of the VoIP protocols have his issue. ISDN PRI or BRI also does not have this issue. GREAT! Our client has 4 ISDN lines and we are using a DIALOGIC 4Port BRI. But the exchange point (don't know how to translate Vermittlungsstelle) of the German Telecom is nearly 5 KM away. Maybe this distance causes the trouble? Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ -- 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] Recommend Wireless IP Phone
--Original Message Text--- From: Wilton Helm Date: Thu, 6 Nov 2008 10:34:35 -0700 Wi-Fi SIP phones aren't limited to hot spots. I am in the process of setting up asterisk for SOHO. At present I'm not even using VoIP trunking, only LAN to stns and I intend to use Wi-Fi instead of analog cordless phone. I got the Engenius one, and it works, but I haven't played with it much. I was disappointed that it only has a single line appearance, as part of my reason for going SIP was to allow the same features like say my 941. I also got their 600 mw access point, but haven't had time to try it. My goal is to cover out 3 acre property and the 1/2 mile road to the mailbox, including mountainous terrain. Maybe I'll share more when I actually get it all put together. Wilton Please do report on your progress. I've tried a few Wifi SIP handset in my home office and for now I've settled upon DECT as a better solution. But new products emerge and Wifi handsets will eventually catch up. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- 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] Phishing attempt
Jeff LaCoursiere schrieb: What about Mexico and Canada? Aren't they considered North America? Canada: yes. Mexico: depends on the definition I guess. http://en.wikipedia.org/wiki/North_America#Countries_and_territories Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] Spam from DIDForSale [EMAIL PROTECTED]
I think I'll take the occasional spam and keep my freedoms and civil liberties... Tell Kim Jong Il I said hello though! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Francis Sent: Thursday, November 06, 2008 11:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED] Gotta love this list being farmed for spammers now. I am sure they call it targeted delivery or some such nonsense. I can't wait for capitalism to completely fail, then there won't be any spam. David Gibbons wrote: I'm glad I'm not the only one who got that. I sent them a nasty response earlier this morning... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Thursday, November 06, 2008 11:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED] On Thu, 6 Nov 2008, Gordon Henderson wrote: didforsale.com have just sent me SPAM to the email address I use here. What a bunch of stupid fuckwits. How to get a 100% cast-iron guarantee that I'll never used their services. Morons. The English have such a way with words :) I keep a local archive of the last 30 days list posts. Searching for didforsale.com shows: Buy unmetered VoIP DID from DidForSale.com is the signature for Jai Rangi [EMAIL PROTECTED]. A wolf in the sheep's pen? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]
He's dead, if you look at the recent photos of him his shadow is not where it should be compared to other people in the photos. Its all photoshop'ed now :) David Gibbons wrote: I think I'll take the occasional spam and keep my freedoms and civil liberties... Tell Kim Jong Il I said hello though! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Francis Sent: Thursday, November 06, 2008 11:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED] Gotta love this list being farmed for spammers now. I am sure they call it targeted delivery or some such nonsense. I can't wait for capitalism to completely fail, then there won't be any spam. David Gibbons wrote: I'm glad I'm not the only one who got that. I sent them a nasty response earlier this morning... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Thursday, November 06, 2008 11:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED] On Thu, 6 Nov 2008, Gordon Henderson wrote: didforsale.com have just sent me SPAM to the email address I use here. What a bunch of stupid fuckwits. How to get a 100% cast-iron guarantee that I'll never used their services. Morons. The English have such a way with words :) I keep a local archive of the last 30 days list posts. Searching for didforsale.com shows: Buy unmetered VoIP DID from DidForSale.com is the signature for Jai Rangi [EMAIL PROTECTED]. A wolf in the sheep's pen? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *Singer Wang* /System and Database Engineer/ The Pythian Group Office: (613) 565-8696 x298 Toll Free: (877) 798-4426 x298 Fax:(613) 565-8710 Email: [EMAIL PROTECTED] MSN:[EMAIL PROTECTED] Yahoo: pythianwang AIM:pythianwang ICQ:201253 Gadu-Gadu: 6817795 Tencent QQ: 858310404 begin:vcard fn:Singer Wang n:Wang;Singer email;internet:[EMAIL PROTECTED] tel;work:(613) 565-8696 x298 x-mozilla-html:TRUE version:2.1 end:vcard ___ -- 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] ODBCExec and Asterisk 1.6 New Thread
Ok, sorry for the response on the same thread.
This is a new one.
The main thing is that with this I set the Store Procedure or Query directly
on the dialplan line, is easier to configure, change, manage, etc.
I also know that works great with heavy load, and it reconnects when the
network goes down and up.
Can you help me porting this app? I think woun`t be difficult for someone
that has port other app.
Attachments: (app_odbcexec) working great on 1.2
(app_odbcexec1.6) my try to port to 1.6
Any idea?? Anybody?
Should this be on development list?
Thanks
/*
* Asterisk -- A telephony toolkit for Linux.
*
* ODBC exec function
*
* Robert Hanzlik [EMAIL PROTECTED]
*
* This program is free software, distributed under the terms of
* the GNU General Public License
*
* Copyright (c) Digium
*
* Based on work by Mark Spencer and Jefferson Noxon - app_db.c
* and Brian K. West - app_dbodbc.c
*
*/
#include sys/types.h
#include stdio.h
#include asterisk/options.h
#include asterisk/config.h
#include asterisk/file.h
#include asterisk/logger.h
#include asterisk/channel.h
#include asterisk/pbx.h
#include asterisk/module.h
#include asterisk/pbx.h
#include stdlib.h
#include unistd.h
#include string.h
#include stdlib.h
#include pthread.h
#include sql.h
#include sqlext.h
#include sqltypes.h
static char *tdesc = Database query functions for Asterisk extension logic;
static char *q_descrip =
ODBCquery(varname=query): Retrieves a value from the database query\n
and stores it in the given variable. Always returns 0. If the\n
query failes, jumps to priority n+101 if available.\n;
static char *e_descrip =
ODBCexec(query): Executes a database query. Always returns 0.\n
If the query failes, jumps to priority n+101 if available.\n;
static char *q_app = ODBCquery;
static char *e_app = ODBCexec;
static char *q_synopsis = Retrieve a value from a ODBC query;
static char *e_synopsis = Execute a ODBC query;
AST_MUTEX_DEFINE_STATIC(odbc_lock);
static SQLHENV HOdbcEnv;
static int ODBC_res; /* global ODBC Result of
Functions */
static SQLHDBC ODBC_con; /* global ODBC Connection
Handle */
static SQLHSTMT ODBC_stmt; /* global ODBC Statement Handle
*/
static char *config = odbcexec.conf;
static char *dsn = NULL, *username = NULL, *password = NULL;
static int dsn_alloc = 0, username_alloc = 0, password_alloc = 0;
static int connected = 0;
static int ast_odbcexec(const char *query, char *out, int outlen);
static int odbc_load_module(void);
static int odbc_init(void);
static int odbc_unload_module(void);
static int odbc_do_query(char *sqlcmd);
static void reconect(void);
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
void LogErrMsg(char * source, long rc,SQLSMALLINT HandleType,SQLHANDLE Handle);
void LogErrMsg(char * source, long rc,SQLSMALLINT HandleType,SQLHANDLE Handle)
{
SQLSMALLINT len;
SQLCHAR msg[200],buffer[200];
SQLCHAR sqlstat[10];
ast_log(LOG_ERROR, Error %s %d\n,source,rc);
SQLGetDiagRec(HandleType,Handle,1,
sqlstat, rc,msg,100,len);
ast_log(LOG_ERROR, %s (%d)\n,msg,rc);
}
static int odbcexec_exec(struct ast_channel *chan, void *data)
{
int arglen, res;
char *argv;
arglen = strlen (data);
argv = alloca (arglen + 1);
if (!argv) /* Why would this fail? */
{
ast_log (LOG_DEBUG, Memory allocation failed\n);
return 0;
}
memcpy (argv, data, arglen + 1);
if (option_verbose 2)
ast_verbose (VERBOSE_PREFIX_3 odbcexec: query=%s\n, argv);
ast_mutex_lock(odbc_lock);
res = odbc_do_query(argv);
ast_mutex_unlock(odbc_lock);
if(res==-1) {
if (option_verbose 2)
ast_verbose (VERBOSE_PREFIX_3 odbcexec: Query
failed.\n);
/* Send the call to n+101 priority, where n is the current
priority */
if (ast_exists_extension (chan, chan-context, chan-exten,
chan-priority + 101, chan-cid.cid_num))
chan-priority += 100;
}
return 0;
}
static int odbcexec_query(struct ast_channel *chan, void *data)
{
int arglen;
char *argv, *varname, *query;
char dbresult[256];
arglen = strlen (data);
argv = alloca (arglen + 1);
if (!argv) /* Why would this fail? */
{
ast_log (LOG_DEBUG, Memory allocation failed\n);
return 0;
}
memcpy (argv, data, arglen + 1);
if (strchr (argv, '='))
{
varname = strsep (argv, =);
query = strsep (argv, \0);
if (!varname || !query)
{
[asterisk-users] [OT] Re: Phishing attempt
Steve Totaro schrieb: It is kind of strange how a county with such a small percentage of the world population holds pretty much the entire world's markets and economy in it's hands. pretty much the entire is not entirely true. :-) http://www.bpb.de/wissen/I6PFEV,0,WeltBruttoinlandsprodukt.html http://sun-bin.blogspot.com/2005/12/map-world-population-and-gdp-scaled.html Europe seems to be on par with North America and Asia is not small either. But yes, talking about countries you're right, more or less. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] ODBCExec and Asterisk 1.6 New Thread
On Thursday 06 November 2008 12:27:05 Sebastian Gutierrez wrote: The main thing is that with this I set the Store Procedure or Query directly on the dialplan line, is easier to configure, change, manage, etc. I also know that works great with heavy load, and it reconnects when the network goes down and up. You can do the same with func_odbc. While templates may make your job easier, you certainly can use whatever syntax you like. And func_odbc manages connections properly, as well. Please also note that a good majority of the folks who would be qualified to look at your app are forbidden to do so, as you have not signed a license for contributions, and even looking at unlicensed code may affect how we code. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED]
Greg Woods wrote: On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote: Gotta love this list being farmed for spammers now. I am sure they call it targeted delivery or some such nonsense. I can't wait for capitalism to completely fail, then there won't be any spam. Socialism has already completely failed. What should we do, go back to a barter economy? :-) ___ -- 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 WHAT?? The telco has raised the cost of my call to Italy from 1 hen per minute to 3! I'll be out of chickens in WEEKS if this keeps up. Next thing you know, the cable co will start demanding 1.5 cows per month. Of course, they don't care that I can't give them .5 cows without wasting a WHOLE one. It's terrible! I only wish there were something we could use for payment instead of commodities -- maybe some sort of note that took the place of a physical commodity! Hey...that sounds like a terrific idea! N. ___ -- 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] AEL NoOp not working
On Thu, 2008-11-06 at 13:55 +0100, Olivier wrote: Yes, you're right : NoOp needs verbosity of 3 and above. Thanks for helping. The surprising thing is that AEL Verbose prints output whatever the verbosity level is (even with 0). Would you qualify this as normal ? Olivier-- The Verbose() app behaves the same whether you call it from AEL or via extensions.conf, or any other method that is used to get dialplan stuff into Asterisk. Are you including the verbosity level? For instance, if you say Verbose(Hi there); the verbosity level is zero by default. If you want to restrict it 3 or more, then Verbose(3,Hello); should do the trick. murf -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Spam from DIDForSale [EMAIL PROTECTED]
Singer X.J. Wang wrote: He's dead, if you look at the recent photos of him his shadow is not where it should be compared to other people in the photos. Well that's just lovely. Kim Jong Il is now an immortal vampire. Better call the white house and tell them to replace the nuclear warhead with a bunch of garlic cloves and a wooden-stake claymore mine . ___ -- 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] RFC: multiple packages editing asterisk config files
On Thu, 2008-11-06 at 17:02 +0200, Tzafrir Cohen wrote:
On Thu, Nov 06, 2008 at 12:57:15PM +0200, Tzafrir Cohen wrote:
Hi
I'm lately bothered with the need to provide a set of Asterisk
configuration files in a package that will be good for a wide range of
Asterisk users.
Asterisk configuration files support #include and a number of other
interesting tricks, as mentioned in
http://svn.digium.com/svn/asterisk/branches/1.4/doc/configuration.txt [0].
Where can we find tools to parse Asterisk configuration files?
Asterisk does not provide any library or utility to read configuration
files. The asterisk-gui uses Asterisk itself to both read and write
configuration files.
Tzafrir--
Not entirely true. In trunk, I developed a set of files in utils
that allow you toread and write config files externally to asterisk.
See extconf.c; it's a pain in the kazoo to maintain, but could be
very useful for those who need to do external config file processing.
It could stand a little updating against the asterisk core, as
new things have been introduced since it was written.
I thought about writing a new parser, about using someone elses' code
that was similar, and my best estimate was that reusing the code in
Asterisk to do it was the cheapest alternative at the time.
The config file stuff is actually pretty feature packed!
murf
The FreeIRIS project uses a set of perl module, that are also available
in CPAN, as Asterisk::config
http://search.cpan.org/dist/Asterisk-config/
http://www.freeiris.org/
(Never tried using them)
Next file to look at is modules.conf . modules.conf is one of those
files with a single section (or maybe two or three, but many entries
in that section). Mostly the order in modules.conf doesn't count.
However I beleive that in some specific cases the order of entries there
does count.
In addition, of of the entries there are of the same two keys ('load'
and 'unload'). Hence it's imporssible to override an earlier
assignment later. This makes module.conf very unmodular (pun not
intended) by nature.
The problem with modules.conf is that a mistake in it can easily get
Asterisk to either fail to load or behave very strange.
Can someone describe to me a real-life situation where the order of
directives in modules.conf matters?
--
Steve Murphy
Software Developer
Digium
smime.p7s
Description: S/MIME cryptographic signature
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Re: [asterisk-users] twice normal beep before busy tone ??
Most IVRs want longer DTMF tone lengths. If you shorten the toneduration= then many IVRs won't work. Wilton Helm wrote: If it is 300 ms, that is way to long. I don't know any CO grade receiver that can't decode in 80 ms and some can do 40. There is also a similar size gap between digits. Is there an option to start dialing as soon as enough digits are collected to guarantee a unique route? That has been the norm in traditional PABXs for 20 or 30 years, and combined with 80 ms duration, it can generally finish by the time the user has entered the last digit. Yes. This feature is called overlap dialing and is only available on PRIs -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- 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] Spam from DIDForSale [EMAIL PROTECTED]
LOL, I love people like Jacque who have no clue that wolves exist. The assumption that evil does not exist in the hearts of men is niavity. What this discussion has to do with asteriskI have no clue... but entertaining none-the-less. http://en.wikipedia.org/wiki/Jacque_Fresco A resource based economy. Greg Woods wrote: On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote: Gotta love this list being farmed for spammers now. I am sure they call it targeted delivery or some such nonsense. I can't wait for capitalism to completely fail, then there won't be any spam. Socialism has already completely failed. What should we do, go back to a barter economy? :-) ___ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ExtenSpy? am I doing it correctly?
I had problems when I was playing with the ExtenSpy command as well. The issue for me was that the context for the extension that I was using was not the same as the one that Asterisk showed in the console output when I called the phone. This is because I have various contexts included in other contexts so it was a bit confusing as to which context the extension was in at some given point in time. After changing things to match contexts stuff worked as expected. -- Jim Dickenson mailto:[EMAIL PROTECTED] CfMC http://www.cfmc.com/ From: Marco Signorini [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thu, 06 Nov 2008 10:38:11 +0100 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ExtenSpy? am I doing it correctly? Hi Steve. I'm still trying the same because I'm interested in the subject. For what I can understand the ExtenSpy application is working properly if the selected extension receives a call. Seems not working, instead, if the selected extension originates the call. My actual setup is like that: Ext12(Soggiorno) == Ext13(Camera) ^ | Ext911- ExtSpy(12) Here is the log when the 13 calls the 12 and 911 is called by an other phone (StudioAV): -- Executing [EMAIL PROTECTED]:1] Dial(SIP/Camera-08231e60, SIP/Soggiorno) in new stack -- Called Soggiorno -- SIP/Soggiorno-082560f8 is ringing -- SIP/Soggiorno-082560f8 answered SIP/Camera-08231e60 -- Packet2Packet bridging SIP/Camera-08231e60 and SIP/Soggiorno-082560f8 -- Executing [EMAIL PROTECTED]:1] ExtenSpy(SIP/StudioAV-0822f350, 12) in new stack -- SIP/StudioAV-0822f350 Playing 'beep' (language 'it') -- SIP/StudioAV-0822f350 Playing 'spy-sip' (language 'it') == Spying on channel SIP/Camera-08231e60 Unfortunately, in the opposite direction: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/Soggiorno-0822f350, SIP/Camera) in new stack -- Called Camera -- SIP/Camera-08231e60 is ringing -- SIP/Camera-08231e60 answered SIP/Soggiorno-0822f350 -- Packet2Packet bridging SIP/Soggiorno-0822f350 and SIP/Camera-08231e60 -- Executing [EMAIL PROTECTED]:1] ExtenSpy(SIP/StudioAV-082560f8, 12) in new stack -- SIP/StudioAV-082560f8 Playing 'beep' (language 'it') == Spawn extension (from-sip, 911, 1) exited non-zero on 'SIP/StudioAV-082560f8' == Spawn extension (from-sip, 13, 1) exited non-zero on 'SIP/Soggiorno-0822f350' The application ExtSpy seems to hang just before playing the 'spy-sip' and I can't hear anything coming from the selected extension. I'm using Asterisk version Asterisk 1.4.20.1 built by root @ Gateway on a i686. Is this the correct behavior or a bug? Thank you and best regards. Marco Signorini. Steve Gladden wrote: Scratching my head and trying this. Asterisk Version: Asterisk 1.4.21.2 Tried: exten = 4771,1,ExtenSpy([EMAIL PROTECTED]) exten = 4771,2,Hangup Also tried: exten = 4771,1,Answer exten = 4771,2,ExtenSpy([EMAIL PROTECTED]) exten = 4771,3,Hangup Also tried many variations including option ,b I think most calls I make are 'bridged' extensions 4771 and 4724 are both in mbb context. Tried 'cycling' though the channels and volule * # no change. Test: 4724 places outbound or extension call. I dial 4771 from 4772 I expect to hear audio from 4724's in progress call but hear nothing. I hear a recording beep when I dial 4771. I expect to hear audio from call being made from ext. 4724 I've obviously got this wrong or the feature is not working :-) Ao far I've been unable to find much information on the net of anyone documenting a problem or a working configuration. Is there something I'm completely missing here? Thanks! Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-962 Asterisk
The linksys phones annoy me because they cannot implement southern hemisphere DST properly. Grr. (yes, you can do it with a hack - but why can't the phones just work?) PaulH Steve Anness wrote: Good Day, I have been tasked with fixing the time on our asterisk server. I am having a hard time finding documentation to tell my what asterisk uses to get its time information to push to phones (or a better question, where does the SPA-962 get its time information)? Basically, I can go under the settings of the phone and change the offset to set the correct hour, but it is still about 4 minutes fast. So the SPA-962 has an offset option, but to offset it from what? The time on the asterisk server? That isn’t right because my asterisk server has the correct time. To offset from GMT? No because I am +6 from GMT not +2. I can physically set the time, but that is a bitch when you have many phones, shouldn’t the phone be syncing with something? Any thoughts? I am not finding anything conclusive. Steve Anness ICT Support Analyst Humanitarian International Services Group ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-962 Asterisk
The linksys phones annoy me because they cannot implement southern hemisphere DST properly. I was shocked the first time I had to write firmware for an international project. Not only is there the southern hemisphere issue of opposite seasons, but just about anyone in the world with a legislative body has to prove their independence from everyone else by defining the dates a bit differently (not to mention time zones that differ by 15 or 30 minutes). Then the US came along and changed their rules after a million products already had them hard coded in silicon! It's a mess. I just wish we'd all forget about it entirely. Its a way to force people who don't like to get up early to do so anyway. A number of studies have been done on the increase in accidents and reduced worker productivity for a week or two after a change. The recent US change was supposed to save energy, but I suspect if one did a study, they would find that businesses just extended their hours to accommodate a diversity of people, thus increasing their energy consumption! Wilton ___ -- 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] Polycom 430 no hangup after SIP BYE, Status 481 instead
I found out what the problem was. It appears to be a bug in the Polycom 430 firmware. I have 2 lines on the phone and both of them use the same auth id but with different servers. It seems that if you make an outgoing call from the phone on line 2 and then called party hangs up. Asterisk says BYE and the Polycom looks at line 1 (because it has the same auth id as line 2) and says I don't have an active call on line 1 when the active call is on line 2. Kinda annoying, but easy enough to work around. I am in the middle of migrating systems so I can just change all my usernames on line 2 to be prefixed with 1 or something like that. On Mon, Nov 3, 2008 at 11:16 AM, Joel Pearson [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I have a really strange problem with a Polycom 430 phone and Asterisk 1.4.20. Currently If I dial the Polycom from my mobile phone answer the call on the Polycom and then hangup the mobile the call ends fine on the Polycom. But if I call from the Polycom to my mobile and then I hang up the mobile the Polycom thinks the call is still active. However doing a show sip channels shows the the call has ended. Further to that doing a tcpdump shows that Asterisk sends a SIP BYE to the phone but the phone responds with: Status 481 Call Leg/Transaction does not exist. The Polycom is currently associated with 2 sip servers (using 2 lines on the phone) because I am currently in the progress of migrating from one server to another. So the asterisk server is having issues with is on Line 2 and it works perfectly well on Line 1 with a completely different Asterisk server running 1.4.16.2. I haven't tried switching the lines around to see if its just a problem with it being on Line 2. The Polycom is running the latest Bootrom and Sip version. Does anyone have any idea what could be causing this? Cheers, -Joel ___ -- 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] ODBCExec and Asterisk 1.6 New Thread
Dou you have any example? Can I call directly to querys without the templates??? Thanks! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Tilghman Lesher Enviado el: Thursday, November 06, 2008 4:53 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] ODBCExec and Asterisk 1.6 New Thread On Thursday 06 November 2008 12:27:05 Sebastian Gutierrez wrote: The main thing is that with this I set the Store Procedure or Query directly on the dialplan line, is easier to configure, change, manage, etc. I also know that works great with heavy load, and it reconnects when the network goes down and up. You can do the same with func_odbc. While templates may make your job easier, you certainly can use whatever syntax you like. And func_odbc manages connections properly, as well. Please also note that a good majority of the folks who would be qualified to look at your app are forbidden to do so, as you have not signed a license for contributions, and even looking at unlicensed code may affect how we code. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and bigmem kernel
What happens?... I'm not sure. IAX peers work fine, but SIP users does not register. There are not firewalls blocking ports. But actually the problem is not the issue because I tried with normal kernel and doesn't work. It is not configuration because it worked on a virtual machine on VirtualBox. Even more strange, Trixbox DOES work. I think I'll continue with trixbox by the moment On Sun, Nov 2, 2008 at 11:18 PM, Tzafrir Cohen [EMAIL PROTECTED]wrote: On Sun, Nov 02, 2008 at 10:03:12PM -0600, Edgar Guadamuz wrote: Hi all, I installed asterisk 1.4.22 on a Dell poweredge 2950, with 4GB RAM. I used debian, but the default kernel doesn't recognize the 4GB, just 3, so I installled the linux-image-2.6.18-6-686-bigmem kernel, that do recognize the whole 4GB. Asterisk seems to be installed correctly, but I had two issues: (1) I had an error with zaptel. Asterisk didn't start with zaptel modules loaded. I had to rmmod zaptel to get asterisk running. lsmod | grep ^zaptel zttest -c 3 (2) SIP doesn't work What did you do? What did you expect to happen? What actually happened? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] ODBCExec and Asterisk 1.6 New Thread
On Thursday 06 November 2008 18:59:29 Sebastian Gutierrez wrote:
Dou you have any example? Can I call directly to querys without the
templates???
func_odbc.conf:
[EXEC]
read=${ARG1}
write=${ARG1}
dsn=something
extensions.conf:
exten = 123,1,Set(result=${ODBC_EXEC(SELECT foo FROM bar)})
--
Tilghman
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Re: [asterisk-users] Help on g729 CODEC
Hi All, I need a help on g729 codec.Is there any tool which can convert g711 codec into g729 codec and supports batch processing ? Thanks in advance vivek --- On Fri, 11/7/08, Edgar Guadamuz [EMAIL PROTECTED] wrote: From: Edgar Guadamuz [EMAIL PROTECTED] Subject: Re: [asterisk-users] asterisk and bigmem kernel To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, November 7, 2008, 9:45 AM What happens?... I'm not sure. IAX peers work fine, but SIP users does not register. There are not firewalls blocking ports. But actually the problem is not the issue because I tried with normal kernel and doesn't work. It is not configuration because it worked on a virtual machine on VirtualBox. Even more strange, Trixbox DOES work. I think I'll continue with trixbox by the moment On Sun, Nov 2, 2008 at 11:18 PM, Tzafrir Cohen [EMAIL PROTECTED]wrote: On Sun, Nov 02, 2008 at 10:03:12PM -0600, Edgar Guadamuz wrote: Hi all, I installed asterisk 1.4.22 on a Dell poweredge 2950, with 4GB RAM. I used debian, but the default kernel doesn't recognize the 4GB, just 3, so I installled the linux-image-2.6.18-6-686-bigmem kernel, that do recognize the whole 4GB. Asterisk seems to be installed correctly, but I had two issues: (1) I had an error with zaptel. Asterisk didn't start with zaptel modules loaded. I had to rmmod zaptel to get asterisk running. lsmod | grep ^zaptel zttest -c 3 (2) SIP doesn't work What did you do? What did you expect to happen? What actually happened? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help on g729 CODEC
On Thursday 06 November 2008 23:45:38 vivek rastogi wrote:
I need a help on g729 codec.Is there any tool which can convert g711 codec
into g729 codec and supports batch processing ?
for from in /full/path/to/directory/*.ul ; do
to=${from%%.ul}.g729
asterisk -rx file convert $from $to
done
assuming you have codec_g729.so loaded and at least one G.729 license
available at any one time.
--
Tilghman
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[asterisk-users] Use of optional new number in ISDN release 22
Hello, Has anyone got any ideea if I can use in Asterisk the new called party number optionally included in the diagnostic field for release cause 22 in ISDN? A callcenter gets lots of this messages from the telco and it would be nice if I could tell them that the number has changed, and that the number is x Thanks, -- Cristian Dimache ___ -- 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] [OT] Reporting Spam
All of you on this list are familiar with how DNS works. You probably use spam blocking lists (SBL) for your email servers? In case someone is interested in building accurate SBL, Spamcop is a service that allows you to easily report spam by sending in the headers. It can be automated by procmail or whatever as well. Spamcop does a deep analysis of the headers of a message and totally ignores what address it may be spoofing. It's very accurate and send out (mostly ignored, but occasionally acted upon) abuse notices. The spam I get from VoIP companies (like the recent one which I got twice) was reported. If only a few of you start reporting this and other spam you receive, it feeds the spamcop list which is very accurate, yet fair in the way it operates, unlike some SBL. My intention in posting this isn't so much to start the inevitable discussion where someone now tries to trash everything I just said, but to inform those who might be interested, that you can help me and everyone else by reporting spam to spamcop.net. The usual caveat: YMMV /r ___ -- 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
