IRC log for #asterisk on 20090618

00:00.11alrscomfrey: don't remember
00:00.36comfreyi think i got it...
00:00.46comfreythe modules were inserted backwards...
00:00.50comfreyduh!
00:01.40jlntlol
00:03.10comfreyok, that should help a bit...
00:03.45jlntI'm still stuck on the reloading page, but asked that question in the wrong channel.
00:03.46drmessanoIs there any way I can choose the sound packages from the CLI when installing?
00:05.59[TK]D-Fenderdrmessano: "from CLI" meaning that you could point to a pre DL-d file?
00:06.19comfreyok, that is it.
00:06.20[TK]D-Fenderdrmessano: because you can already select for DL from menuselect
00:06.24comfreythanks gang.
00:06.32alrscomfrey: np
00:07.13comfreyloader.c:400 load_dynamic_module: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_masq_park_call
00:07.20comfreyanyone seen this?
00:07.34drmessano[TK]D-Fender: Something I can run as a command + switches.. perhaps a variant of configure
00:08.00alrscomfrey: I haven't, but I don't use DUNDI. If you don't use DUNDI you can disable the module in modules.conf
00:08.09fileload res_features.so before chan_dahdi.so
00:08.14[TK]D-Fenderdrmessano: Not that I've heard, rather more like complaints askeing the same as you
00:08.29drmessano[TK]D-Fender: Same boat for ilbc
00:08.35drmessanohmmm
00:17.54*** part/#asterisk Tar-Get (n=Tar-Get@83.101.36.144)
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01:07.51[8none1]Will asterisk work with the Polycom server side DND feature?
01:08.10[8none1]I'm getting a 489 Bad Event error when I try using it
01:12.43[TK]D-Fender<PROTECTED>
01:12.51[8none1]thx
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01:53.39piper69ok guy i need someone to look at this simple sip and extension i have questions http://pastebin.ca/1464352
01:56.53AlmightyOatmealdoes anyone have any experience with cisco ip phones? i got the sip firmware loaded on a 7911G but it keeps saying "Unprovisioned" and I don't see how my config file is wrong... i obtained working copies from some wikis and my * server doesn't even see the ip phone try to connect
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01:58.07KyleKhow is it configured?
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01:58.22KyleKmaybe its failing at getting the file from the webserver/tftp/AVP
01:58.27AlmightyOatmealhttp://pastebin.ca/1464351
01:58.34AlmightyOatmealno, its not failing getting it via tftp
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02:02.37AlmightyOatmealConnection received from 192.168.1.20 on port 49152 [17/06 21:02:05.593]
02:02.37AlmightyOatmealRead request for file <CTLSEP001C58577242.tlv>. Mode octet [17/06 21:02:05.609]
02:02.40AlmightyOatmealFile <CTLSEP001C58577242.tlv> : error 2 in system call CreateFile The system cannot find the file specified. [17/06 21:02:05.609]
02:02.43AlmightyOatmealConnection received from 192.168.1.20 on port 49153 [17/06 21:02:05.828]
02:02.46AlmightyOatmealRead request for file <SEP001C58577242.cnf.xml>. Mode octet [17/06 21:02:05.828]
02:02.49AlmightyOatmealUsing local port 1529 [17/06 21:02:05.828]
02:02.51AlmightyOatmeal<SEP001C58577242.cnf.xml>: sent 15 blks, 7281 bytes in 0 s. 0 blk resent [17/06 21:02:05.906]
02:02.54AlmightyOatmealsorry :(
02:03.04AlmightyOatmealmy right mouse button got a little excited
02:08.23AlmightyOatmealholy balls i think i fixed it!
02:10.09AlmightyOatmealbut now whenever i tried to dial it brings me to a voicemail prompt
02:10.59AlmightyOatmeallooks like i need to tweak an outbound call rule for this phone?
02:13.08AlmightyOatmealso to modify an outbound call rule so i have to dial 9 before the number do i do something like exten => _91NXXNXXXXXX or exten => 9_1NXXNXXXXXX ?
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02:15.07*** mode/#asterisk [+o Deeewayne] by ChanServ
02:20.42AlmightyOatmealthe 7911 keeps wanting to dial an extension rather than an external phone number.. i can't quite figure this out.. i even made an outbound dialing rule in asterisk like exten => _91NXXNXXXXXX but when i hit '9' it says extension 9 was not found :(
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02:27.31Dovidhi - been having trouble figuring out the error: WARNING T.30 ECM carrier not found
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02:29.56jblackmany phones have an internal dialplan that takes some masaging
02:30.13AlmightyOatmealyeah, thats what i'm figuring which is a pain in the booty hehe
02:30.38AlmightyOatmealbut if i could just get extension 9 to roll with dialing outbount numbers, this cisco phone would be happy as a clam on prozac
02:30.38jblacksome you can override it with dialing the number, then send/dial/whatever
02:30.56AlmightyOatmeali'm not that lucky here, it still takes the first digit and thinks its an extension
02:31.06jblackoh, cisco? I think that company makes crap
02:31.34jblackway up in bigiron land they're ok, but in the consumer space, they're a disaster
02:32.21AlmightyOatmealyeah no joke
02:32.41AlmightyOatmeali spent days getting this phone to work, now that it half-ass does i'm like yaay
02:32.56AlmightyOatmeali'm thinking of buying a simplier sip phone off ebay once i get some money this weekend from my rents
02:33.00jblackreturn it through their front window, and get a polycom
02:33.11AlmightyOatmeallol yeah, i'm eyeballing a cheap polycom
02:33.24jblackreally configurable, you'd like it
02:34.18jblackon the downside, you'll want a dhcp server and a http/ftp or scp server, though you don't _need_ them
02:34.47jblackscp shoudl be easy. Who doesn't have ssh?  ;)
02:35.09thuddwhirrdoes anyone know if there is a dialplan application, like background, that can be interupted by speech as well as dtmf?
02:35.24jblackyou can check the answering machine thing.
02:35.31jblackbut I don't know if that'll cover what you need
02:35.36jblackamd I think it's called
02:35.50thuddwhirryeah, it doesn't have a playback component
02:36.06thuddwhirri'm trying to hack together the equivelent of "barge"
02:36.29thuddwhirri.e. asterisk starts playing instructions, and if the person starts speaking, it terminates playback
02:39.13thuddwhirroo.  BackgroundDetect looks about right
02:39.53AlmightyOatmealgrr, the cisco is dialing the phone number as an extension.. i must me missing somethign in my extensions.conf
02:40.45AlmightyOatmealnvrmnd
02:40.50AlmightyOatmealgot it
02:41.12AlmightyOatmeali win! :D
02:44.15[8none1]I'm trying to integrate my asterisk queues with my agents jabber status. Has anyone done this and if so can you provide an example?
02:44.47[8none1]I want the system to pause and unpause the agents when the sign off or go to an away status
02:45.13[8none1]I've read this : http://www.voip-info.org/wiki/view/Asterisk+Jabber
02:45.41[8none1]And I see the jabberstatus app but I'm not sure how to integrate this with queue pausing
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03:02.09drmessanoso what happens if I just happen to have a cdr_odbc.conf and a cdr_mysql.conf ?
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03:13.57srf21c~book
03:13.57infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
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03:21.52orangeservicegreets
03:25.05orangeservicehave a weird extension matching issue... wondering if anybody has seen anything like it before. I am matching on _9., some numbers I dial match every time, but some numbers (starting with 9) never match at all in the debug.
03:27.08Dovidcan anyone help with "load_dynamic_module: Error loading module 'app_fax.so': libspandsp.so.2"
03:27.25Dovidi found the file on my server. why would asterisk not 'see it" ?
03:27.44drmessanoWhich version of Asterisk?
03:28.27Dovid1.6.0.10
03:28.27drmessano1.6.?
03:28.29drmessanoOk
03:28.39Dovidhad it b4
03:28.43Doviddont remembre what i did to fix it
03:28.53Dovidi think its more of a linux issue as to where asterisk is lookig
03:28.56Dovidlooking* for it
03:29.30drmessanoOnly issue I know of is spandsp 0.0.6 wouldnt work on 1.6, which has been fixed
03:29.53Dovidi did it on another machine (same set up) with no issue
03:30.22Dovidi think it has to do where asterisk is looking for the lib. any way i can figure out where asterisk is looking for i ?
03:30.39Dovidit*
03:31.07drmessanoAsterisk shouldn't have passed the ./configure stage if it couldnt find it.. Maybe you need to go back that far.. Did you upgrade spandsp and not recompile?
03:31.26Doviddidnt think of that ;)
03:31.31Dovidlet me re-compile asterisk
03:31.52drmessanoI believe 0.0.5 vs 0.0.6 have different APIs, which is why there was a "bug" previously in using the newer
03:32.16Dovidusing the latest from steve's site
03:32.30drmessanoI bet you had 0.0.5 previously
03:32.45drmessano(I just went through this 2 hours ago)
03:32.52Dovidapp_fax was in make menuselect
03:33.05Dovidjust complied 0.0.6 and noew - recompiling asterisk
03:33.20Dovidbeen a long day ;)
03:33.28Dovidi have in /usr/src 0.0.4 and 0.0.6
03:33.32drmessanoah
03:33.41drmessanoyep, thats definitely your issue
03:33.51Dovidi think i did /usr/src/span*0*
03:34.09Dovidissue that the old modules r in there or asterisk is doing soething wrong ?
03:34.38drmessanoIf you installed spandsp 0.0.6 and didnt recompile asterisk, asterisk is still compiled with 0.0.4 support
03:35.00Dovidwill see as soon asterisk is done compliling ;)
03:35.38Dovidi didnt think i compiled it, in the past if it was too old asterisk didnt even 'see it'
03:36.31Dovidmeeeeh
03:36.32Dovidsame error
03:37.06drmessanocd to the spandsp 0.0.4 src and make uninstall
03:37.20drmessanothen go to the 0.0.6 and configure / make / make install
03:37.26drmessanothe go back to asterisk
03:37.29drmessanothen
03:37.39drmessanoconfigure / make / make install
03:37.45Dovidi just did that
03:37.57Dovidroot@flig [/]# ls -l /usr/local/lib/libspandsp.so.2
03:37.57Dovidlrwxrwxrwx  1 root root 19 Jun 17 23:25 /usr/local/lib/libspandsp.so.2 -> libspandsp.so.2.0.0*
03:38.05Dovidis this suppsoed to be a sym link ?
03:38.39drmessanolrwxrwxrwx 1 root root 19 Jun 17 20:33 /usr/local/lib/libspandsp.so.2 -> libspandsp.so.2.0.0
03:39.13Dovidwat do u get for: ls -l /usr/local/lib/libspandsp.so.2.0.0 ?
03:39.31Dovidi wonder why the * at the end
03:39.39drmessano-rwxr-xr-x 1 root root 1513423 Jun 17 20:33 /usr/local/lib/libspandsp.so.2.0.0
03:41.04Dovidhttp://pastebin.com/m4c5f5bd
03:41.08Dovidthey all have * at the end
03:41.12Dovidmaybe thats the issue
03:41.21Dovidstrange - never seen that b4
03:41.42Dovidor is that because of the sym link ?
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03:42.21Dovidnope. on my other box i have same sym links but no *
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03:48.09Dovidseems there is a -F option in sym link
03:48.16Dovidnot sure why span dsp did it
03:48.43DeeewayneDovid, ./configure --prefix=/usr
03:48.57Dovidin spandsp ?
03:48.57Deeewaynefor spandsp ^
03:49.02Dovidok. let me try that
03:49.09Dovidwut does that do ?
03:49.26Deeewayneit will put the lib into /usr/lib instead of /usr/local/lib
03:49.31Dovidah ok
03:50.11Dovidwuts causing it to use /usr/local/lib ? distro ?
03:50.41Deeewaynenot sure
03:50.56Dovidok. thanks. i am trying that now
03:51.21Dovidweird. on another box i have /usr/local/lib (where the file is) and it works.
03:52.19Dovidyea. tha fixed it
03:52.20Dovidthanjs
03:52.23Deeewaynenp
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03:54.49Dovidwow. my first t.38 with asterisk and it worked. :)
03:54.55Dovidthis is a great day for me ;) thanks a lot
03:55.25Dovidalso - whats this error ? didnt see much on google other than others asking about it
03:55.25Dovid[Jun 17 23:52:53] WARNING[20998]: app_fax.c:128 span_message: WARNING T.30 ECM carrier not found
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04:59.49OrNixHello. Tell me please, how can i bind chan_h323 on 0.0.0.0:1720? I need to receive h323-calls on few interfaces of asterisk-server. (sorry for my poor english. I have too little practice)
05:06.11kaldemarOrNix: change bindaddr and port in /etc/asterisk/h323.conf
05:09.15OrNixkaldemar, "this SHALL contain a single, valid IP address for this machine". By default chan_h323 can't bind on 0.0.0.0, but may be are there another ways?..
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05:11.35kaldemarOrNix: you could always use iptables, but it's a bit of a hack.
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05:13.27OrNixkaldemar, do you mean DNAT to address, on which asterisk is listening 1720/tcp ?
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05:14.10kaldemaryes, but then you'd have to mangle the h.323 packets too, which is ugly. there's an iptables module for it though.
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05:16.40OrNixkaldemar, i've got an idea to run another copy of asterisk on necessary interface, bind its SIP-module to 127.0.0.1:5081 and link with main asterisk :)
05:19.04OrNixbut.. overhead
05:21.46kaldemarh323.conf accepts 0.0.0.0 and it listens to 0.0.0.0 too.
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05:22.50kaldemarOrNix: i recall having some trouble with the address it binds RTP to with multiple interfaces. but i'd give it a try anyway.
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06:00.14OrNixkaldemar, my asterisk dies with bindaddr = 0.0.0.0, when recieves incoming call
06:06.36kaldemarh.323 has always been nice
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06:08.49OrNixchan_ooh323 can listen 0.0.0.0, but i don't like it
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06:16.24BoardyYesterday is was helped with setting up my MOH. It is working, but not when I call from an external line
06:16.41Boardydoes anybody have a clue where I should start searching?
06:16.48Boardy(for a solution)
06:18.34Boardywhen I turn on "verbose", I can see that the MOH is started, but I don't hear anything.
06:21.59BoardyI will check back later when more people are awake.
06:27.59Boardybye bye
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07:01.12sujith_hhi all. I added 2 extensions using freepbx. 100 and 101. When i used ekiga to connect them i am getting 'remote user cleared the call'. I tried to call 101 from 100, like sip:101@192.168.10.1
07:01.31sujith_hcan anyone help me what could be the issue?
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07:02.57sujith_hthis is how my sip_additional.conf looks like http://pastebin.ca/1464526
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07:10.23angryuserhi all
07:10.54angryuserBobPierce, check that you routed you udp ports
07:11.01angryuseryour*
07:11.24angryuserBobPierce, ah sorry Boardy is gone...
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07:15.29sujith_hangryuser: can you help me? I had added 2 extensions using freepbx. Now my extensions are registered in the ekiga. But i am not able to call. My sip_additional.conf looks like this http://pastebin.ca/1464526
07:16.47angryusersujith_h, what is happening in cli when you call ? give me output of "sip show peers" command
07:21.10sujith_hangryuser: i am getting http://pastebin.com/m28bf49af
07:22.19sujith_hangryuser: oops this is the one i get http://pastebin.com/m44ce2d47
07:22.21angryusersujith_h, to get to "CLI"  type "asterisk -rvvvvvv"   and then sip show peers
07:23.12angryusersujith_h, after what pass a call and copy all output what you get in CLI console to pastebin
07:24.06sujith_hangryuser: this is the output i get for sip show peers http://pastebin.com/m4cf7d220
07:24.19sujith_hangryuser: how to pass call in cli?
07:24.39sujith_hangryuser, *CLI
07:24.57angryusersujith_h, look like you sip chan is not loaded
07:25.54sujith_hangryuser: I am new to this. What i want to do is to have 2 extensions and then to call them.
07:26.07angryusersujith_h, type "modules show like sip"
07:26.34angryusersujith_h, have you installed by youself ? from iso or compiled ?
07:27.07kaldemarfreepbx...
07:27.48sujith_hangryuser, i had installed asterisk from the distro ubuntu. And then by using freepbx i am trying to configure asterisk to make calls \
07:28.50sujith_hangryuser: http://pastebin.com/m460c4637 is the output of 'modules show like sip'
07:29.43angryusersujith_h, ah its "module show like sip"
07:30.28sujith_hangryuser: this is the output i get http://pastebin.com/m7116f170
07:31.20angryusersujith_h, hm, chan sip is here, type 'help' and paste output
07:31.30angryusersujith_h, what version shave you installed ?
07:32.10angryusersujith_h, use of web interface is a bad idea when you want to learn, why dont you want to compile form sources first ?
07:33.14sujith_hangryuser: the source version is 1.4.17 and the output of help is http://pastebin.com/m5ca02b3b
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07:35.53angryusersujith_h, source version 1.4.17 and zaptel/dahdi version ?
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07:36.45sujith_hangryuser: oops zaptel and dahdi are not installed :(
07:37.44angryusersujith_h, it is  not a problem with sip to sip, type "module load chan_sip.so"  what happens ?
07:38.43sujith_hangryuser: this is the result http://pastebin.com/m46e6fd45
07:39.14angryusersujith_h, and now "sip show peers"
07:40.02sujith_hangryuser: this is the output http://pastebin.com/m69e408af
07:40.14sujith_hangryuser: and yah 100 and 101 are online :)
07:40.24angryusersujith_h, call
07:42.07sujith_hangryuser: nope am not able to :(
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07:44.08angryusersujith_h, i recommend you to start learning asterisk without web interface first,
07:44.25angryuser~book
07:44.25infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
07:44.37sujith_hok
07:44.49angryusersujith_h, you wil see after it will be simple
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07:50.26pifcan I rm -rf voicemail messages without problem?
07:50.42pifeven if asterisk is running?
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08:06.56KyleKpif: it reindexes them
08:07.27KyleKalso just make sure nobody is using the specific voicemail folder
08:09.05pifthx
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08:27.41Madkisshi folks
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08:29.54geningreetings folks
08:30.10genini got a quick asterisk video call question if anyone is alive
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08:30.17geninhi
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08:34.53KermitTheFraggergenin: just ask, if anyone knows the answer, he or she will speak up
08:35.34geninwe are trying to push a video call from asterisk to a 3g phone but when the mobile rings, it shows up as a data call and not a video call
08:35.56geninany ideas why?
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09:07.26jerryeguruwhat does one mean when they say buy termination. what exactly do they mean by termination
09:09.15jerryegurui am trying to set up a calling card solution with asterisk
09:12.12Chainsawjerryeguru: Termination means "making things end up on the old-fashioned phone network", generally.
09:12.35Chainsawjerryeguru: My phone system here is terminated on 4 ISDN BRIs.
09:13.16Chainsawjerryeguru: I have Patton SmartNode hardware doing the work, but you can also pay a SIP provider to do it for you, where you just authenticate and talk SIP to them and they do the heavy lifting.
09:14.16jerryeguru<PROTECTED>
09:15.07geninanyone have H324 working on asterisk and if so a pastebin of your sip.conf and extensions would be nice
09:15.08genin:)
09:15.26geninbefore one of our people jumps out of a window
09:18.04jerryeguruChainsaw: if i am to setup a calling card solution, why would i need to terminate if the voip digital interface card can also provide dialing to any phone number
09:18.42Chainsawjerryeguru: You may be terminating on that card then. What card is exactly? ISDN BRI/PRI, T1, E1?
09:19.16jerryeguruChainsaw:  E1
09:19.37Chainsawjerryeguru: Okay, so your phone system is terminating on a single E1 using some sort of digital interface card.
09:19.56jerryeguruChainsaw:  yes
09:20.10jerryeguruChainsaw:  actually i have 2 E1s
09:20.20Chainsawjerryeguru: Then you wouldn't need an external VoIP provider to do termination for you.
09:20.31jerryeguruChainsaw:  ok. thanks
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09:38.53casixhello
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09:39.29casixis it possible to define variables for every users in sip.conf and use this variables in the dialplan?
09:40.05angryusercasix, not in sip.conf, you can you astdb fo that
09:40.10angryuseruse*
09:41.21casixyes, but i would like to define this in the sip.conf because it is more clear, all options of the user are together
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09:47.14twanny796is there a way to do voip behind a symmetrical NAT?
09:48.50MaliutaLap~sipnat
09:48.50infobot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
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10:18.19tzafrir_laptopcasix, setvar
10:18.57tzafrir_laptop(also in iax.conf . as of 1.6: also in chan_dahdi.conf and probably in some other channel drivers)
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10:33.15casixtzafrir_laptop: thx :)
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11:38.00aksynhi guys - should an ast_translator's destroy callback only ever get called at the end of a call?
11:38.29aksyn(sorry, just realised there's an #asterisk-dev)
11:47.51chrisdushello, I have a B410P card with 2 spans coming in, on the S/T box the LINE leds are green, the leds on the card are both green, but when I run dahdi_scan one of the spans shows up as alarms=UNCONFIGURED ... Any idea what can cause this? The other span is alarms=OK ..
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11:54.51yanghello mort_gib
11:55.01mort_gibHi Yang
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11:57.03thomashave my chan_sip.c patched
11:57.10thomasnow compile...
11:57.14thomasis it posible to load this
11:57.16thomaswithout
11:57.20thomasshutdown asterisk?
11:57.36thomas<PROTECTED>
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12:00.47*** join/#asterisk Slashman (n=Slash@ariane.fimasys.com)
12:03.38*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)
12:03.42*** join/#asterisk Heretic (n=BuRn@ZA1-securenode.echelon.co.za)
12:06.14p4tr0p1I have a Digium Wildcard TE110P T1/E1 R2 signaling, zap end openr2. I've configured my extension with SetCallerPres(prohib) to don't show my ID when I'm calling pstn, but I'm still seeing my pbx ID when I'm calling my cell phone =/ Anybody?
12:09.26[TK]D-Fenderp4tr0p1: Plan B : Try setting the callerID to something like "00000000"
12:11.14p4tr0p1[TK]D-Fender, I did it after start searching ;o/
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12:23.20joobieTK, know of a way to find out what the longest hold time of a queue is via AMI?
12:23.31joobieQueeuStatus seems to only report the average wait time
12:24.16[TK]D-Fenderjoobie: There is via CLI, threefore there is via AMI.  Think on this a little
12:24.22[TK]D-Fendertherefor*
12:28.35chrisdusWhat would be a reason for not having a chan_dahdi.conf in my /etc/asterisk directory?
12:28.53chrisdus(fresh install btw..)
12:30.55SuPrSluGdid you make config?
12:32.18SuPrSluGit doesn't install config files by default, in case you already have some you don't want overwritten
12:32.49[TK]D-Fenderchrisdus: Depends what version of *, and whether or not you built the samples
12:33.03tzafrir_laptopchrisdus, or alternatively, write one
12:35.43SuPrSluGdoh, make config instals the init scripts make samples installs default config files
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12:37.46p4tr0p1[TK]D-Fender, you don't have a Plan C ??? huehuehueuhehue
12:38.04[TK]D-Fenderp4tr0p1: Plan C : Get a PRI and lead a happy life.
12:38.30p4tr0p1libpri?
12:42.32p4tr0p1hehehehehe
12:43.04p4tr0p1tks man!
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12:52.35devyllchan_dahdi.c:2789 pri_find_dchan: No D-channels available!  Using Primary channel 16 as D-channel anyway!
12:52.45devyllcan anybody tell me why do I get that ?
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12:54.11[away]Shazaump4tr0p1 yo :D
12:54.33p4tr0p1[away]Shazaum, haha!
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13:01.29joobieergh back
13:01.31joobiethanks TK
13:01.39joobieheros about to start, im out
13:01.40joobiepeace
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13:15.04Kattyis armed.
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13:15.48wonderworldhi, i transfer an incoming call to another extension with a snom 320 phone. i am wondering how to set a variable that would contain the SIP-username of MY phone (the phone that initiated the transfer)
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13:16.40Kattynail file, nail buffer, cutical cleanser, nail cuticle remover, base coat, white nail polish, top coat. guess what i'm doing today!
13:16.52filedon't nail me!
13:17.19Kattyit starts with FRENCH
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13:18.16[away]Shazaumwonderworld: set(callerid(name)=...)
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13:20.53wonderworldShazum: i am doing it with a macro...... a call comes in, someone presses transfer and the "quick dial button" on the phone. phone goes to the pressed extension in the dialplan. THERE i must find out somehow who transfered the call...
13:21.16wonderworldit must be variable.... can't set it for every phone
13:21.55wonderworldi tried to use the BLINDTRANSFER var, but it's empty
13:23.09[away]Shazaumi dont know ur dialplan
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13:25.47kaldemarwonderworld: set the variable for the calling channel before the first dial.
13:27.19*** join/#asterisk b1shop (n=chad@c-67-173-5-134.hsd1.il.comcast.net)
13:27.46b1shopmorning all.  i am looking to batch convert over 1K .g729 files to wav or mp3.  ideas?
13:32.28[TK]D-Fenderb1shop: sox <-
13:32.59b1shoptried that.  sox formats: no handler for file extension `g729'
13:33.50[TK]D-Fenderb1shop: asterisk -rx "convert abc.g729 abc.wav"
13:35.33kaldemar"file convert .."
13:36.38b1shophmm.  i need a running asterisk server for this?
13:38.18b1shopnevermind.  forgot sudo.  more coffee
13:39.15b1shopUnable to open input file: foo.g729
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13:41.05[TK]D-Fenderb1shop: Probably plenty of other apps to do this, but Feel free to have a good cry over it :)
13:41.23b1shopugg
13:41.27*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)
13:41.40[TK]D-Fenderb1shop: And * can only convert if it can read it.
13:42.08b1shopi just need to be able to play some of them to figure out what they are
13:42.22b1shopi thought VLC, etc.. but nothing can play them
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13:45.04b1shopgrr.  * can't convert them either
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13:48.17[away]Shazaumok
13:48.50Corydon76-digb1shop: do you have g729 licenses?
13:49.25b1shopCorydon76-dig: nope.  actually didnt know i needed one.  hmm.
13:49.41[TK]D-Fenderb1shop: * can... if you installed the codec <-
13:49.49Corydon76-digb1shop: Correct.  As a patented codec, you need a license to compress or decompress the codec
13:49.55[TK]D-Fender[09:41]<[TK]D-Fender>b1shop: And * can only convert if it can read it. <- did tell you
13:50.40b1shopsorry.  didnt know it was a licensed codec
13:51.03Corydon76-dighttp://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC if you would like to purchase a license
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13:52.26jerryegurui am looking for an open source asterisk client that can run on mobile phones, preferably a java application
13:54.48b1shop$10 is fine for the license if it does what i need...
13:55.37kaldemarjerryeguru: try fring
13:57.34kaldemaroh my, fring is closed.
13:58.35jerryeguruoh!
13:59.29Kattybuffs [TK]D-Fender's nails.
13:59.42[TK]D-FenderKatty: MANicure!
13:59.55Kattyi went and got stuff for a french manicure this morning.
14:00.07Kattyi decided i was going to check out the other half of my brain which i usually ignore.
14:00.13[TK]D-Fendergoes to find the remaining members to form a Poison tribute band...
14:01.46Kattyit is truly suprising how informative youtube can be.
14:01.55Kattyespecially with everyone doing product reviews and tutorials.
14:02.51[TK]D-FenderKatty: Yup, saved me from buying a synth I was looking at...
14:03.05Katty[TK]D-Fender: this girly thing is most... fascinating.
14:03.08Katty[TK]D-Fender: i've been missing out.
14:07.40superbeefWhat's the backwards compatibility lack between asteirsk 1.4.x to asterisk 1.2?
14:07.46superbeeflack=like
14:08.00[TK]D-Fendersuperbeef: Concerning?
14:09.09superbeef[TK]D-Fender: I've got several early era 1.2.x boxes out that also have amp 1.x, and I'm doing a refresh, and replacing those hosts with newer builds. I want to make sure 1.4.x boxes can play nicely with the 1.2.x until I replace them all
14:09.43[TK]D-Fendersuperbeef: there are config changees and AMP.. lets not go there.
14:09.53*** join/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net)
14:09.55[TK]D-Fendersuperbeef: Your GUI OWNS you.
14:10.25superbeef[TK]D-Fender: haha.. i've managed ot upgrade 1 or 2 boxes from amp 1.x to current freepbx without issue
14:10.36superbeef[TK]D-Fender: i've also had some failed occurrences
14:10.48gr0mitok, anyone recommend a channelbank for 100 analogue extensions?
14:10.50superbeef[TK]D-Fender: i take it that it would probably be safest to re-enter everything by hand
14:10.52[TK]D-Fendersuperbeef: taht is the issue.  tis config differences
14:11.57superbeef[TK]D-Fender: so the real concern is configs written by the GUI rather than asterisk versions?
14:12.30[TK]D-Fendersuperbeef: huh?
14:13.02[TK]D-Fendersuperbeef: 1.2 vs 1.4 have config differences, and AMP may have been using more 1.0 rev code which was deprecated in 1.2 and removes in 1.4
14:14.46Kattyhoray!
14:14.49Kattynails are all buffed!
14:15.07superbeef[TK]D-Fender: so would you consider it unwise to have 1.4x boxes and older 1.2x boxes attempt to communicate with each other, or as long as I define the IAX2 trunks properly on the new boxes, it will be fine
14:15.34[TK]D-Fendersuperbeef: Generally not a big deal.
14:18.15jerryeguruok i hv googled my fingures off for a java client for mobile phones that can be used to dial into asterisk
14:18.21jerryeguruany ideas on this ppl
14:18.55Katty[TK]D-Fender: these are SOOOO pretty :>
14:19.01superbeef[TK]D-Fender: do you typically build your asterisk deployments from source or use packages.. I've been building from source, but I'm going to swtich to Ubuntu 8.04LTS boxes, and I was entertaining letting ubuntu maintain the packages
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14:19.09Katty[TK]D-Fender: they're reflective!!!!
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14:20.20Joe_CoThey, so I have a phone that registers when it's next to me in the office, but not from home. It's using stun to get the ip, but it's still not registering. Here's the sip debug of it:
14:20.23[TK]D-Fendersuperbeef: Source only, and wouldn't consider Ubuntu
14:20.31Joe_CoThttp://asterisk.pastebin.com/d3e1427
14:21.28superbeef[TK]D-Fender: wouldn't consider ubuntu as an OS, or their packages
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14:22.33[TK]D-Fendersuperbeef: YES
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14:23.40superbeef[TK]D-Fender: haha....   My current boxes are unspported fedora.. that's brutal, then I built a gentoo box for my Hub, but I'm always so afraid of breaking something when I upgrade with gentoo on production servers
14:28.15Joelsuperbeef I use ubuntu with no issues.
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14:31.06BoardyYesterday is was helped with setting up my MOH. It is working, but not when I call from an external line.
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14:31.15BoardyDoes anybody have a clue what can be the cause?
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14:32.25eppigyI need to cut back on caffiene
14:32.45Boardyon the CLI the MOH is started, but I don't hear the music.
14:34.05Kattyeppigy: YOU
14:34.10Kattyeppigy: are you rationing my hugs?!
14:34.36*** part/#asterisk Joe_CoT (n=joecot@ubuntu/member/pdpc.bronze.joeterranova)
14:34.45eppigyhuggles Katty
14:34.50Kattyhugs eppigy
14:34.53Katty:>
14:34.54Kattybetter.
14:35.04eppigymake innapropriate noises
14:35.11eppigypoors spelling
14:35.12Katty:<
14:35.15eppigy!!!
14:35.25eppigyoh man
14:44.33Madkissmiaow.
14:45.08jerryeguruanybody knows a java asterisk client on mobile phones
14:45.15brahDoes txfax have limitations on faxing multiple pages?
14:45.38eppigyeyes Katty's nails
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14:55.34Kattyeppigy: i've only got them buffed, and the cuticles taken care of.
14:55.45Kattyeppigy: it's been busy
14:56.15brahWell, it does fax multiple pages.
14:56.19brahBut it's erratic, as expected.
14:56.31brahFax Transmission complete, check return code.
14:56.33brahWhere do I check it?
14:57.53Kattyeppigy: now base coat is on
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15:05.03eppigyleans close to Katty smelling her hair
15:05.13Katty...
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15:05.34Kattypersonal bubble, sir!
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15:06.22eppigyD:
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15:08.29[TK]D-Fenderpulls out a giant needle and eyes Katty's bubble with an evil grin
15:08.38KattyAHHHHHHHHHHHHHHHHHHHHHHHHHHHHHH
15:08.42Kattydies.
15:09.46eppigyI am all about invading personal space
15:09.56eppigyI mean given the right circumstances
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15:14.57rue_mohrtodays question is re hwec taps, where is it specified how many are available, this card can do 24, we use 6 and were getting a long delay echo the users are annoyed with
15:15.51[TK]D-Fenderrue_mohr: 24?  Supposed to be 128
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15:16.46rue_mohrmaybe I understood how many channels the hwec counld do
15:17.07rue_mohranyhow, I'd like to know what its set up to be able to do, and dial it up if I can
15:17.09*** join/#asterisk awkfu (n=awkfu@66.162.90.56)
15:17.19[TK]D-Fenderrue_mohr: Also depends on which revision of that card you're using.
15:17.42[TK]D-Fenderrue_mohr: the older one used the shitty VPM which got replaced by an Otasic based unit
15:17.53hescousing aplay to play a vm received by email, I get: "can't play WAVE-file format 0x0031 which is not PCM or FLOAT encoded".  Can anyone advise how to convert this into an accessible format at a cli prompt?
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15:18.18rue_mohrI used the guidelines on a webpage to adjust incomming 1mw to 14844 on the dahdi monitor, at that I have to turn the rx gain on the polycom to 11db for the users to hear properly
15:18.31rue_mohrvpm
15:18.35rue_mohr....
15:19.20rue_mohrvpmnlpthresh.... damnit
15:19.27leevi'm using asterisk 1.4.24.1 with gui 2.0.4 on freebsd.  when a call comes in I get:
15:19.28leev[Jun 18 09:47:07] WARNING[90224] app_dial.c: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
15:19.28leev[Jun 18 09:48:28] WARNING[90224] ast_expr2.fl: ast_yyerror():  syntax error: syntax error, unexpected '=', expecting $end; Input: = 1
15:19.39leevany ideas how to fix that?
15:19.59rue_mohrtk, you mean after all that freaking big deal about me buying a hewc, your telling me I have the wrong freaking one!?
15:20.23rue_mohrtk, is this a hardware change!? not just firmware?
15:20.49[TK]D-Fenderleev: that is 2 separate problem.  first requires seeing what caused the error along with channel debug.  the token error is a dialplan expression failure.  PASTEBIN the complete call attempt and dump the peer you are dialing.
15:20.51[TK]D-Fender~pb
15:20.51infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/
15:20.54e0n`hmm, how would I use sendtext or console send text to send out a text message for a reporting system I wrote
15:21.04[TK]D-Fenderrue_mohr: HARDWARE.  Completely different EC board
15:21.08rue_mohrdamnit
15:21.23[TK]D-Fenderrue_mohr: Go verify which one you have precisely
15:21.27rue_mohrso I'm suppiosed to shell out another freaking $800 for a different board?!
15:21.59rue_mohrits the  vpmnlpthresh paramiter thats been giving us all that trouble
15:22.06rue_mohrso I suspect we have the old card
15:22.08rue_mohr!?
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15:23.29[TK]D-Fenderrue_mohr: Go verify.
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15:24.30rue_mohrso I have to go and take the computer down and tear it open again, or can you tell me something on the software I can look for
15:24.43tzafrir_laptoprue_mohr, what computer is it?
15:24.50leev[TK]D-Fender: do i just do 'asterisk -rvvvv' and and just paste the output from that?
15:25.11[TK]D-Fenderleev: PASTEBIN.  Do not spam in here.
15:25.22tzafrir_laptopFor that ammount of channels, consider using a software EC
15:25.24[TK]D-Fenderrue_mohr: The module should appear in dmesg
15:25.26rue_mohrits a compact evil
15:25.35leevyeah, i meant pastebin, but is that the output i want?
15:25.48brahWhat goes after the pipe in the Data field on a call file?
15:25.54[TK]D-Fenderleev: verbose 10, and all the details i jsut asked for
15:26.12rue_mohr(compaq evo)
15:26.29[TK]D-Fenderbrah: depends on the app you're calling
15:26.39rue_mohrits strapped to the wall with large metal brackets and lagbolts
15:27.15brahCool
15:27.37rue_mohrits a pain to take the lid off while leaving it running
15:27.42rue_mohrand hooked up
15:27.43devyllI can't seem to place outgoing calls through trunk. I use digium board TE220 with dahdi. app_dial.c:1237 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
15:27.54brahMore specifically, here
15:27.57brah"Data:/tmp/faximage.tif|caller"
15:27.58rue_mohrdevyll,
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15:28.19rue_mohrif you say    core show channeltypes    you will not see the dahdi driver in there
15:28.28rue_mohrit didn't load cause there was an error
15:28.35rue_mohrbest of luck working out what the error was
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15:29.05leev[TK]D-Fender: http://pastebin.com/m3d67b78f
15:29.17[TK]D-Fenderbrah: Go read the apps instructions.
15:29.53devyllrue_mohr
15:30.21devyllrue_mohr : dahdi is loaded. it appears. devicestate: no. indications: yes. transfer: no
15:30.31[TK]D-Fenderleev: Why are you showing me 2 SIP peers when its an ***IAX2*** peer that fails?
15:30.33rue_mohrfor me it happened when I changed drivers and the install files didn't update the modules configuration to load new echo modules to ...
15:30.37rue_mohrk, I cant help
15:30.38*** join/#asterisk SAT1 (i=d945ad33@gateway/web/ajax/mibbit.com/x-54c29220a55a61e4)
15:31.03[TK]D-Fenderdevyll: What good is that error message when we can't see what caused it?
15:31.06SAT1Hi everyone, is there any free SIP web based dialer?
15:31.41*** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be)
15:31.54tzafrir_laptopAny idea if the YeaLink user spamming voip-info is really someone from YeaLink? Are they that stupid?
15:32.10rue_mohrVPMADT032: Present and operational (Firmware version 107) [TK]D-Fender what am I looking for?
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15:33.22leev[TK]D-Fender: sorry.  i'm quite new to this.  http://pastebin.com/m2c6e9def
15:33.40devyll[TK]D-Fender. ok, can you tell me what means devicestate:no ? . I actually need to know where to start ? the interface seems up. i can recieve calls from outside and dahdi_cfg -vvv is ok , dahdi_tool shows green light, dmesg is ok . only thing I have is logs from /var/log/asterisk and dmesg. which is what i posted above.
15:33.42EmleyMoorSIP response 603 "Declined" where the client is X-Lite - what does it mean?
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15:34.11batphonestill having hell getting RSA auth to work for IAX
15:34.29batphoneevery time i make a call over the trunk it says something about accepting unauthenticated call
15:34.57batphonein caps
15:35.06batphonemy phone system yells at me :<
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15:36.29[TK]D-Fenderrue_mohr: Yes, what you're looking for... I'm a little gray on if thats the better one, or if that only applied to the digital cards
15:36.59[TK]D-Fenderleev:   Addr->IP     : (Unspecified) Port 0 <-- you are calling an IAX2 peer that has not registered
15:37.17leev[TK]D-Fender: they are both registered as sip devices though
15:37.23leev'sip show peers' has them both there
15:37.41[TK]D-Fenderleev: but you are not DIALING 2 SIP devices.  You are dialing 1 SIP and 1 IAX2 <-
15:37.54EmleyMoorI notice the MWI flag is stuck on on the X-lite that's giving 603
15:38.06[TK]D-Fenderleev:  -- Executing [6001@DLPN_Default:1] Dial("SIP/6000-28726000", "SIP/6001&IAX2/6001") in new stack
15:38.13leevthats what i don't understand.  how do i make them both sip?
15:38.43[TK]D-Fenderleev: this is your config, go change it.  If you don't know where in the GUI to go change that, ask in their channel as GUI's are not supported here
15:38.56leevalright
15:39.12leevthanks
15:41.35leev[TK]D-Fender: think i found it, thanks
15:42.46angryuserhi all, i am searching fo asterisk built in, somethinkg like IP04 from atcom have someone tested it ? http://www.atcom.cn/En_products_IP04.htm
15:43.01angryusersomething*
15:43.34angryuserAre there any analog products (besides digium appliance)
15:43.38angryuser?
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15:49.35brahIt's not faxing this particular pdf... meanie..
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15:57.05markithi, just curious, any simple way to know if a registar is using asterisk for providing voip?
15:57.25rue_mohryes, get them to put you on hold :)
15:58.09markitrue_mohr: lol, they could have changed voices (expecially because it's italian)
15:58.20rue_mohrno I meant the music
15:58.23markitnothing like telnetting to some port?
15:58.39markitah, I see... same problem, they could have changed with custom ones
15:58.52rue_mohrcould have...
15:59.13rue_mohrarg, I know there is a echo suppression level setting here somewhere
15:59.56markitok, not important at all, thanks
16:01.07rue_mohranyone know where the echo suppression agresiveness level is set?
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16:13.58jerryeguruany open source mobile sip dialers
16:14.03jerryeguru?
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16:18.37CanWoodHey Folks.  I'm looking to make a limited account in manager.conf that will allow ActivaTSP to connect and dial for an extension range.  Does anyone have a link to robust documentation for the opions under each [user] listing?  Could I enable dialing do it with just "read = call"? can it be limited by extension?
16:21.07russellbthe short answer is that you can't.
16:21.11CanWoodheh
16:21.21russellbsorry :-(
16:21.33CanWoodhey, better to know than to invest days in trying to figure out how
16:21.45CanWoodI appreciate the candor
16:21.51russellbyeah, manager permissions are not fine grained enough
16:22.02russellbonce you open up the ability to originate calls, you can't really control what is done with it
16:22.19russellbincluding originating a call to execute System(rm -rf /)
16:22.22russellbso ... yeah.
16:23.07CanWoodevery example I've seen of a manager.conf includes "read = system,call,log,verbose,command,agent,user" and "write = system,call,log,verbose,command,agent,user"
16:23.18CanWoodso I figured with just call, it could be configured safely
16:23.33CanWoodshort of reading the source, it's hard to find good docs on it though
16:23.37russellbnods
16:23.57russellbI'm not sure what the o'reilly book has on this, that'd be the other place i would check
16:24.06CanWoodchecked there first
16:24.12russellbah.
16:24.12CanWoodgood suggestion though.  tks
16:24.19russellbmost people use manager the way you've seen it
16:24.46russellbit's not really built for access other than extremely trusted access ..
16:25.01russellbruns off to go test something in another office
16:25.03CanWoodheh.  And most people log in to their Windows boxes as Administrator accounts :)
16:25.12[TK]D-FenderCanWood: Write a proxy that scans.
16:25.16russellbgood luck
16:25.37leev[TK]D-Fender: i get this error now. http://pastebin.com/m5da7a19f
16:25.50leev6000 can call 6001.  but when 6001 tries to call 6000, it doesn't work
16:26.47[TK]D-Fenderleev: Still looks like its calling.
16:26.57CanWood[TK]D-Fender: if I had access to the servers, I'd consdider it.  I'm researching on behalf of someone who's asterisk hosting provider said no to giving them acces to AMI.  With the security model it provides, it sounds like their hosting company gave the right answer
16:27.08[TK]D-Fenderleev: and thats a dialplan error for trying to set the DYNAMIC FEATURES var to "blank"
16:27.48[TK]D-FenderCanWood: Noone said you had to put the proxy on the server itself.  It need only be the poitn of access for the other endpoints
16:29.08CanWood[TK]D-Fender: But when that proxy receives scans accepts and then tries to forward the request to initiate the call, it would then need AMI access to the server at the host's endpoint.  That's what they won't give us
16:29.52*** part/#asterisk nextime (n=nextime@unaffiliated/nextime)
16:29.58[TK]D-FenderCanWood: You CAN set up an AMI account right?
16:30.21CanWoodnot on the server where the extensions are defined.  No
16:30.34*** part/#asterisk jerryeguru (n=Administ@41.222.2.65)
16:30.46[TK]D-FenderCanWood: Well I guess the point of limiting something you can't even create is a moot point.
16:31.33CanWood[TK]D-Fender: If I could provide a safe configuration for the hosting provider to consider and review, I thought they may flex on the "no AMI access" standpoint.  Thant's the angle I was coming from
16:31.51[TK]D-FenderCanWood: they'd have to install the proxy then.  Oh well..
16:32.17CanWood[TK]D-Fender: exactly.  Still, thanks for the suggestion.  Any ideas are appreciated
16:32.29[TK]D-FenderCanWood: run your own damn server :p
16:32.40CanWoodI have one.  That's why the person is looking to me for advice ;)
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16:49.05adr3nalin3does the vpmadt032 work with a TDM800P card?
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17:00.49e0n`hmm, I need to find a more efficient way to send out sms' via asterisk, I am not sure what type of equipment or what exactly i'll need I setup an SMS dialplan, but it doesn't SMS, it calls my external phone.
17:06.45alrse0n`: I've been using messagepub.com
17:06.51alrs7 cents each
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17:18.34juanIMPhi all, can anyone tell me which is the more stable release of 1.4.x ?
17:19.01seanbrightdoes anyone here have the tc400b?
17:19.32_ShrikEseanbright: yes
17:20.01seanbright_ShrikE: did it come with a standard sized bracket or just the low-profile one?
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17:20.48_ShrikEseanbright: We got the ones we have about a year an a half ago and they came with a standard bracket.
17:20.55seanbrightdamn.
17:21.09seanbrightnew ones are low-profile
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17:35.35MrTelephoneDoes anyone know if there is an extensions.conf variable for auth username for sip clients?
17:36.26Kattyhmm. lunch kinda sucked.
17:38.07[TK]D-FenderMrTelephone: "core show functions like SIP"
17:38.54eppigyKatty: D:
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17:45.24Kattyeppigy: IF YOU LIKE PINA COLADAS
17:51.01adr3nalin3I just installed a hardware echo canceller on a TDM800p and now when starting zaptel I get this error:  No hardware timing source found in /proc/dahdi, loading dahdi_dummy
17:53.30leifmadsenanyone ever do anything with video in asterisk? I'm curious if anyone has been able to record a video prompt before?
17:53.35leifmadsenor, video voicemail
17:54.11eppigyKatty: I drink margaritas from time to time
17:56.06Nuggetpiña coladas!
17:56.38superbeefleifmadsen: no, but it sounds way cool
17:57.15leifmadsensuperbeef: heh ya :)
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17:58.25Kattyi think someone missed my reference.
17:59.00Kattyleifmadsen: no luck with video and asterisk before.
17:59.06Kattyleifmadsen: but then again, i've not really played with it much.
17:59.12Kattyleifmadsen: also.
17:59.17Kattyleifmadsen: give me your bailey's
17:59.24leifmadsenKatty: ummm... no :)
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18:02.16Katty:<
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18:11.30`paulcould be offtopic: does anyone know a good US tollfree provider?
18:12.02Corydon76-digBinfone is what I use
18:12.17Corydon76-digI assume you mean with voip termination, right?
18:14.47jplankwith dahdi, dadhi_genconf configures the software echo can by default. Can having the software echo can and hardware echo can both enabled cause echo?
18:15.16jplankI didn't start to have echo problems on my FXO lines (2400p) until I upgraded to dahdi
18:15.42eppigyDONDE ESTA
18:16.50AlmightyOatmealno, don't touch me there!
18:16.58eppigy8[]
18:17.07AlmightyOatmealruns away crying
18:17.29eppigySTRANGER DANGER
18:18.03adr3nalin3could someone please help me get rid of HPEC?  Digium tech support is always giving me this call you back bs now
18:22.07Kattyconjuction junction, what's your function
18:22.10Kattyi need a pick me up
18:22.45tzafrir_laptopjplank, if you have a hardware echo canceller, it is used regardless of the existing of a software one
18:23.48tzafrir_laptopadr3nalin3, what do you mean by "get rid of hpec"? Is it dahdi or zaptel?
18:26.03eppigyKatty: I need a mellowing agent
18:26.12eppigyI am white knuckled at the wheel
18:26.48jplankits weird, up until the point that I upgraded from zaptel to dahdi I had no echo. Once I went over to dahdi (using pretty much the same config files just renamed) I'm getting echo
18:27.55adr3nalin3it is dahdi, digging deeper HPEC most likely isn't my problem.  whenever I start dahdi I get this-> Running dahdi_cfg:  DAHDI_CHANCONFIG failed on channel 1: No such device or address (6)
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18:30.03Kattyeppigy: nap
18:31.05adr3nalin3tzafrir_laptop: Do you have any idea what that might be caused by?  I just installed the hardware echo canceller today.  I also disconnected a couple of usb plugs to make the VPM card fit
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18:32.49eppigyKatty: I am taking a half day tommorow
18:33.00eppigybut then I am workign over the weekend :<
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18:34.38tzafrir_laptopadr3nalin3, please post your /etc/dahdi/system.conf
18:34.57adr3nalin3tzafrir_laptop: looks like the computer isn't seeing the card at all
18:35.14tzafrir_laptopand also the output of:  dahdi_hardware -v
18:35.43adr3nalin3tzafrir_laptop: dahdi_hardware shows nothing
18:36.10tzafrir_laptopis the card actually connected?
18:36.15tzafrir_laptopdoes it show on lspci?
18:36.17adr3nalin3I'm going to have the person that is on-site re-seat the card.
18:36.20adr3nalin3no on lspci
18:36.45adr3nalin3do you think disconnecting the USB connectors could/would have caused any IRQ problems?
18:37.43tzafrir_laptopadr3nalin3, I don't know those cards so I really have no idea
18:38.21tzafrir_laptopand this looks like a good time to head home
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18:39.37[pnp]tomasis there anywhere I can find a list or guide of variables for exten's, ie. _1X matches 1*****(to infinity) - I'm trying to find similar examples
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18:40.35[TK]D-Fender~book
18:40.35infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
18:40.37[TK]D-Fender[pnp]tomas: ^^^^
18:40.42jplankin the documentation for chan_dahdi.conf, for echocancel it mentions "There are no special settings required for ; hardware echo cancellers; when present and enabled in their kernel ; modules, " what module are they referring to?
18:40.49[pnp]tomasrgr, thanks Fender
18:41.03leifmadsenAsterisk 1.4.26-rc3 has just been released! You can find more information about it at http://www.asterisk.org/node/48601
18:42.24[pnp]tomasis the 2nd edition updated to reference dahdi instead of zaptel?
18:42.28leifmadsenno
18:42.39leifmadsendahdi did not exist when the 2nd edition was written
18:42.48[pnp]tomasgotcha
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18:51.10SomethingIsoddhello all does anyone here use freeradius with asterisk?
18:51.24marc7trying to compile asterisk-addons 1.6.1.0 on debian-lenny, getting errors compiling format_mp3.c in function mp3_open // request for member ‘ptr’ in something not a structure or union
18:51.59marc7any ideas on what package might be missing?
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18:59.27areaymy incoming calls are being dropped by my asterisk server... i noticed on my iax2 debug, it says "Accepting UNAUTHENTICATED call" ...could that be my problem?
19:00.04[TK]D-Fenderareay: No... they are being ACCEPTED <-
19:00.23[TK]D-Fender"Status = success" <- how do I fix this failure?!?!
19:01.08areaylol
19:01.10*** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net)
19:01.36areaywell what does this mean then: == Spawn extension (incoming, 05603******, 1) exited non-zero on 'IAX2/217.14.138.126:4569-10047'
19:02.20[TK]D-Fenderareay: Means it quit for some reason.  Of course you're not showing what its doing or any debug so I suppose you think we're PSYCHIC
19:06.20[TK]D-Fender\o/
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19:14.21spckafternoon guys
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19:15.35spcki'm playing with dynamic features in features.conf today and I don't seem to be able to setup something to use Goto or Gosub
19:15.49spckfrom within the applicationmap
19:16.01ruben23hi
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19:18.55spckhere's a sampling of the conf files and some cli output http://pastebin.com/d52e356a5 if anyone would like to look at it
19:19.01spckhi ruben
19:20.23ruben23<PROTECTED>
19:21.22spckgot me
19:21.29ruben23having this on the output on my cli http://pastebin.com/m14e8950f
19:25.06carrartell whoever 62.41.66.37 is to disable comfort noise since you don't support it
19:26.08[TK]D-FenderAnd just because your server isn't behind NAT doesn't mean your clients aren't and that you don't need to accomodate them as per the guide
19:31.33ruben23<PROTECTED>
19:32.25[TK]D-Fenderruben23: that is 1 peer, and you aren't showing the failed call
19:33.02AlmightyOatmealsomeone want to buy a cisco 7911G ip phone? i already have the firmware and configs for asterisk ready to roll
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19:33.13AlmightyOatmeali would keep it but the hard drive in my laptop died and i'm dead without it :(
19:34.20ChainsawAre you in the UK?
19:34.47ruben23this is the failed call http://pastebin.com/m7e717ba1
19:35.10AlmightyOatmealChainsaw: nope
19:36.00ChainsawNot interested then, sorry.
19:36.25AlmightyOatmeal:'(
19:36.33[TK]D-Fenderruben23: Doesn't show SIP debug to see why it "failed"
19:37.57Kattyzzzzzzzzzzzzzzzzzzzzzzzzzzzz
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19:38.38Kattyi think lunch made me all blahrghfffff this afternoon
19:38.39[TK]D-Fendergrounds Katty
19:38.42ruben23<PROTECTED>
19:41.08[TK]D-Fenderruben23: It shows * cancelling for some reason
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19:41.46[TK]D-Fenderruben23: deny=0.0.0.0/0 <-remove
19:41.48[TK]D-Fenderpermit=70.42.72.140/255.255.255.255 <- remove
19:42.07ruben23ok
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19:52.24ruben23<PROTECTED>
19:55.10ruben23still the client can hear us
19:55.14ruben23:-(
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20:00.19gr0mitanyone integrated * voicemail with a meridian option 11 Release 21?
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20:11.48spckis there any way to change what context you are operating in via the dialplan?
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20:12.51Kattytwitch
20:12.52neurosysspck: see the goto()
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20:12.57Kattyomg.
20:13.03Kattycaffeine IV stat!
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20:14.15spckahh you need to specify the extension and priority as well
20:15.26neurosysspck: yes
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20:23.24pittstainsI have a question about passing values from the AMI to the Dialplan... I'm using AMI's Originate command to initiate an external phone call and pass it to the extension of my choosing.  However, once the call is passed to the Dialplan, I can't seem to retrieve the number that Originate dialed.  Any suggestions?
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20:25.03pathhi guys
20:25.09pathI'm getting this weird error
20:25.11path[Jun 18 16:12:24] WARNING[22305]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0xa322b30 (len 396) to 200.24.227.34:5060 returned -1: Operation not permitted
20:26.21[TK]D-Fenderpittstains: set it as a var in your originate
20:26.27[TK]D-Fendercheckout time, later all
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20:34.26ruben23hi
20:36.35neurosysHeh now that TK is on his way home... can i ask without getting eaten alive: why do most of you feel polycom bests others with the hardphones?
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20:41.46Joelneurosys I like aastra the most *shrug*
20:42.03Joeland yes, putting [tk]d-fender on ignore is always a good idea.
20:42.25neurosysheh The only hardphone ive used are linksys :P esp. the spa-942
20:42.44pathnice phone ;)
20:44.09neurosysIts seems to work fine
20:44.47neurosysIm just wondering how the cisco's and polycoms are better
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20:48.48Joelcisco are junk imho
20:49.04Joelpolycom are nice, but I like aastra, as they seem to actually pay attention to the open source world
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20:57.45saxaok, what is the difference when I do iax2 show peer mybox and I see a line expire: 922
20:57.58saxaon the other box I do the same and I see expire: -1
20:59.17Joelno expire set would be my guess
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21:00.36saxaJoel: ok, this means exactly what ?
21:00.49saxano expire to registering ?
21:01.01Joel...
21:09.00KavanS[Jun 18 14:04:27] WARNING[27385]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'LOCAL' (cause 0 - Unknown)
21:09.03KavanS^^^ whoa!
21:09.07KavanSanyone have any ideas?
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21:11.36batphoneKavanS: please rebooting the PC
21:12.14KavanSbatphone, that's not what I wanted to hear :P
21:12.16QwellKavanS: is chan_local loaded?
21:12.17[TK]D-FenderKavanS:  Show us the F'n call :p
21:12.32KavanSchan_local is loaded, just wondering if I'm doing a jacked method of dialing
21:12.45[TK]D-FenderLOVES seeing the results of errors and never the SOURCE of them.
21:15.36KavanSlet me get the prior logs....yes chan_local is loaded
21:18.14batphoneKavanS: please let me know what version of Internet Explorer you are using?
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21:18.25batphone;)
21:18.40*** join/#asterisk l2trace99 (n=jr@75.112.140.9)
21:18.54batphonesry.. i just had to argue with some dude in bangladesh about getting a bios upgrade on a pc
21:19.08batphonehe told me bios upgrades are not supported by warranties
21:21.34tompawhi
21:21.55tompawwhat is the best way to ensure that a specified dialplan command will be executed after the call?
21:22.17tompawI have Dial(...) and after one of the sides hangs up I need to perform a MYSQL query.
21:22.27tompawIs there a way to send a macro pointer to Dial() command?
21:23.32[TK]D-Fendertompaw: "h" <- go read up on your dialplan basics & *'s list of Standard Extensions
21:24.38tompaw[TK]D-Fender: MYSQL is not a problem, only making sure that it gets executed
21:24.51tompawI guess I'm missing something basic about the dialplan itself.
21:26.42smpstompaw, yes, you are missing "h" extension
21:26.57KavanSbatphone, wtf?
21:28.42tompaw[TK]D-Fender, smps: a haaaa! thx!!
21:28.55tompawand sorry for my rtmf question.
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21:30.30batphonetompaw: just dont let it happen again, or else
21:30.32linageewhy isn't caller ID used more often as an authentication method for IVRs?
21:30.40batphoneits spoofable?
21:30.46linageeif people are worried about caller ID spoofing you can always use callback. :)
21:30.56nkohhthen you're having to call back :(
21:31.05linageecalls back: "to validate the selections you've made, please press 1"
21:31.28tompawhm. looks like my variables are destroyed when I go from "s" to "h"
21:32.12linageenkohh: call your credit card company, it calls you back, never have to enter any of your info or verify anything. that would be sweet.
21:32.56linageeprobably confusing for the consumer who doesn't understand caller ID can be spoof, but still sweet. :)
21:33.06nkohhwhat if you're on an office phone?
21:33.10nkohhi call most stuff while at work.
21:33.30linageenkohh: have it selectable online somehow to only work for cell phones? *shrug*
21:33.30nkohhmay as well fight with my credit card company on work time.lol.
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21:33.59nkohhlinagee: i guess
21:34.07linageenkohh: better yet, only do it when you know the number is a cell phone. use XXX-XXX prefixes to identify
21:34.21batphonei must have missed exactly what we are discussing
21:34.23linageeer, NPA-whatnot. however you call that
21:34.28batphonewhat is the purpose of this system again?
21:34.54linageebatphone: identify yourself in a database without having to enter countless numbers and such using caller ID and callback verification
21:35.22tompaw"h" works like a charm now. thank you.
21:35.25linageeheck, offer callback when a representative is available. that works nicely too.
21:35.29batphonelinagee: as nice as that sounds, someone could jack your cell phone and disrupt that whole process pretty easily
21:35.33linageetompaw: did you just hang me up? :)
21:35.50tompaw:-)
21:35.53linageebatphone: if someone jacked my cellphone and it's not locked, I have bigger things to worry about
21:36.07tompawhmm
21:36.11batphonelinagee: seems like a liability on both ends
21:36.13[TK]D-Fender[17:30]<linagee>why isn't caller ID used more often as an authentication method for IVRs? <- Sure, you can call me back at 911-555-1212
21:36.27tompawit doesn't work if I hang up during the dtmfs that are called as Dial(...,D()) parameter
21:36.40linagee[TK]D-Fender: put a bit of intelligence into that to avoid calling back 900-sex-chat and such. :P
21:36.44[TK]D-Fenderlinagee: Or one of the NANPA "no toll limit" zones that wil FUBAR them on their next bill
21:36.46batphonelinagee: let me know when you get that system online so i can point it at my 1-900 systems ;D
21:36.46tompawif I wait for the DTMFs to be dialed and call is established and hang up, "h" is executed
21:37.03tompawif I hang up during that D() process, it finishes the DTMF in the background
21:37.05linageebatphone: LOL. maybe you can with someone else's system? :)
21:37.07tompawand "h" is ignored.
21:37.14[TK]D-Fenderlinagee: Of abuse your IVR to DOS someone
21:37.36linagee[TK]D-Fender: ouch
21:37.38[TK]D-Fenderlinagee: Nice idea... except for it being stupid :p
21:37.38tompawshould I use "t" extension for that?
21:37.46linagee[TK]D-Fender: you have me stumped, sorry
21:38.20linagee[TK]D-Fender: oh wait no your idea would not work
21:38.31linagee[TK]D-Fender: this is for pre-exiting customers.
21:38.34batphonelinagee: as it is, whenever i am trolled by some IVR for my callback number for a poll or something stupid like that, i always put in the number i called
21:38.46linagee[TK]D-Fender: unless you brute force tons of numbers through it, but then you can rate limit things
21:38.50talirk81anyone have an idea why  "EXEC HANGUP"  from an AGI would be getting a -1   agi debug and  core set debug 9  dont give me any extra information on why it counldnt hang up the chanel
21:39.03linageebatphone: wow. does it call itself? :)
21:39.06[TK]D-Fenderlinagee: plenty of work, little benifit
21:39.19batphonelinagee: unless the sysops are braindead it probably doesnt
21:39.41linageebatphone: maybe it calls itself back and forth several times and ties up an endless number of lines? :)
21:39.42batphonelinagee: but its like every website that wants my personal infromation just to read some stupid forum post or something
21:39.58batphonelinagee: i always enter their own WHOIS information into their own web forms for them to spam
21:40.35linageewow you guys are tough. :)
21:40.51linageei guess it's a brutal brutal world out there in the telco system
21:41.41tompawwill "t" extension be executed if I terminate a call during dtmfs being dialed with Dial()'s D() ?
21:41.54batphonelinagee: now that the telephone system and internet are one in the same it is even worse than before
21:42.24batphonelinagee: not only are there phone numbers to abuse, but more often than not there is some server with an IP address somewhere attached to those numbers
21:42.42batphonelinagee: and, invariably, stupid people running said servers in many cases
21:43.43batphoneanyone remember that 10-10-811 number you could dial for long distance calls?
21:43.59batphoneif you dialed 10-10-811, then entered 10-10-811 as the number you wanted their system to dial for you
21:44.02batphonebad things happened
21:44.57linageebatphone: what happened?
21:45.13batphonelets just say that 10-10-811 ceased to work for a couple of hours
21:45.25linageebatphone: can you stack 10-10-XXX numbers and make providers charge each other? lol
21:45.58batphonelinagee: considering that they did not filter the own number, presumably at that time that would have worked
21:46.23linageeor will you just end up getting triple/quadruple charged? lol
21:46.23batphonei worked on an asterisk system once that was having load issues
21:46.23beek~BOOK
21:46.23infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
21:46.34tompawdamn, manual doesn't say anything about it
21:46.45batphonei went to see what was wrong and found out that someone had forwarded their recption phone to their main number
21:46.51batphonewell, the reception phone WAS the main number
21:47.21batphoneand i found out later that the load average on the providers broadsoft cluster went through the roof for a while
21:47.21linageebeek: nice. it's on safari. :)
21:47.27linageebeek: i just got the phone
21:47.29linageeer, book
21:48.22linageeasterisk motto: it's only software. lol
21:48.25beeklinagee: Thanks.   I was looking for the link for a reply to the mailing list.
21:48.59linageemark spencer has some quotes in the book? neat
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21:49.55tompaw[TK]D-Fender: "h" extension isn't powerful enough, do you know any tricks to tune it?
21:50.32SomethingIsoddhello all does anyone here use freeradius with asterisk?
21:50.34NuggetYou know how you can tell if a computer programmer is an extrovert?   He looks at YOUR shoes while he's talking to you.
21:50.51batphonehaha
21:51.10linageei'd bet you $10 that mark spencer has a team of students in india writing his code for him. :)
21:52.01linageei've seen him code in person before, just saying it would be more efficient to multiply your capacities to do something by outsourcing.
21:52.31[TK]D-Fendertompaw: how so?
21:52.43Nugget"efficient" and "outsourcing"?  heh.
21:52.48linageetompaw: crankslam?
21:53.00linageei mean crackslam
21:53.29linageetompaw: see futurama when hermes hangs up on the professor's arch nemesis.
21:53.49tompaw[TK]D-Fender: I have a Dial(blah,D(...)). If I hang up after the call has been established, "h" is executed properly.
21:54.13tompaw[TK]D-Fender: However, if I hang up during the D(...) string being called by Asterisk, "h" is ignored.
21:54.19linageetompaw: http://www.urbandictionary.com/define.php?term=crackslam
21:54.40tompawlinagee: I am familiar with that word ;)
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21:55.46tompawwhat happens then anyway? it's not documented anywhere in Dial's manual.
21:56.09tompawconsole only says "Destroying blah_trunk"
22:01.59tompawMaybe I should switch from Dial's D() to SendDTMF()?
22:02.08[TK]D-Fendertompaw: Doubt it highly.
22:02.35tompaw[TK]D-Fender: but then when I hang up during SendDTMF, there is a slight chance "h" gets executed.
22:02.44tompawI don't know, I'm just throwing ideas...
22:04.17tompaw"t" extension didn't help.
22:04.20[TK]D-Fenderis off to martial arts & stage-time
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22:04.28tompawgood luck
22:06.59*** join/#asterisk simNIX (n=simNIX@156-60.bbned.dsl.internl.net)
22:11.10tompawI will try Dial's M()
22:11.12tompawwell
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22:11.26tompawI would try M() if only I knew how to pass additional parameters to that macro.
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22:12.06tompawwith ^?
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22:18.12saxaok
22:18.29saxai have 2 boxes, connected by iax2
22:19.11saxawhen I want my local box call the remote one I do the Dial(IAX2/remotebox/101
22:19.16saxa)
22:19.44saxaso how do I make this extension call the Zap/1 external line ?
22:20.00saxashould I replace the 101 with Zap/1 ?
22:21.38saxaits alittle bit confusing to me the 2 extensions.conf files
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22:21.47saxathey should be similar
22:24.24tompawwhoa, it says SendDTMF returns -1 on failure or hangup
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22:24.36tompawthat -1 on hangup might do the trick maybe?
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22:29.54*** join/#asterisk areay (n=areay@188.220.16.92)
22:30.25areay[TK]D-Fender, i got it working... i had it on the wrong context lol...
22:31.01tompawhow do I capture the value returned by SendDTMF?
22:32.36areayi use iax for my incoming and outgoing calls... the iax.conf entries are exactly the same... but for incoming calls (from the pstn) i get one-way audio... i don't understand it because my outgoing calls (to the pstn) are fine
22:33.34*** join/#asterisk TheHypnotist (n=eyes@77-96-97-206.cable.ubr17.nmal.blueyonder.co.uk)
22:34.04TheHypnotistAnyone know where I kind find information for voice recognition of numbers?
22:34.15TheHypnotists/kind/can
22:35.02tompaw"In general, if an application exits with an error it will return -1, if successful it returns a 0. A return value of -1 means that Asterisk will hang up the channel without passing it along the dialplan."
22:35.06tompawDamn.
22:35.15tompawI need to be able to react for that "-1" code.
22:35.18tompawHow do I do it?
22:38.29saxanight ppl
22:39.39tompawcome on people, it must be a very common situation!
22:43.01tompawis the only way to mess around with SendDTMF's source code?
22:45.03russellbuse the 'h' exten
22:45.27tompawrussellb: it's ignored when the call is hung up during either SendDTMF() or Dial's D()
22:45.36russellbwhat is
22:45.43tompawthe "h" extension
22:45.51russellbthat seems wrong ...
22:45.57russellbit shouldn't ...
22:46.01tompawit only works when I wait for D() or SendDTMF() to finish and hang up then
22:46.20tompawrussellb: what about this quote "In general, if an application exits with an error it will return -1, if successful it returns a 0. A return value of -1 means that Asterisk will hang up the channel without passing it along the dialplan."
22:46.30russellbwhat about it
22:46.35tompawWhen a call is hang up during SendDTMF, it returns -1.
22:46.39russellb-1 just  means hangup, but from there 'h' should run
22:46.46tompawit doesn't
22:47.23*** join/#asterisk buttons840 (n=buttons8@c-24-10-149-58.hsd1.ut.comcast.net)
22:47.45tompawstatic int senddtmf_exec(struct ast_channel *chan, void *vdata) << what if I modify this manually and re-compile?
22:48.02tompawand make it return 0?
22:48.22russellbyou have the source :-)
22:48.26russellbif you want to do it, go for it
22:48.28tompawinstead of: res = ast_dtmf_stream(chan, NULL, args.digits, timeout <= 0 ? 250 : timeout, duration); return res;
22:48.36tompawbut do you think that would help in any way?
22:48.38russellbreturn 0 instead
22:48.54russellbit will ensure the dialplan continues to the next step
22:49.07tompawsounds scary to me, I never touched a sourcecode ever.
22:49.11tompaw*'s sourcecode ;-)
22:49.17leevrussellb: do you have experience with asterisk in freebsd?
22:49.26russellbleev: nope.
22:49.32russellbtompaw: i have.  ;-)  it's safe
22:49.32leevdamn.
22:49.44tompawok, return 0 it is then.
22:49.48leevfresh install and its coring when trying to run
22:49.51tompawdo I have to make uninstall first?
22:49.54leevof asterisk1.6
22:49.56russellbnope
22:49.59tompawcool.
22:50.00russellbyou'll just install over it
22:50.00tompawok.
22:51.41tompawif it works, fucking hell...
22:52.25buttons840as we learn in the book "few things are harder to put up with than the annoyance of a good example"
22:56.51tompawit compiled just fine, but it still ignores "h"
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23:01.00tompawdamn. there must be a "bulletproof" way to do it. simply register an action to be done when the call ends. no matter what the reason for that is.
23:01.28*** join/#asterisk kamanashisroy (n=kamanash@119.30.36.70)
23:02.06buttons840I'm very new, but isn't that was h is for?   And it's ignoring h?   Sounds frustrating.
23:02.41tompawit is. now I commented out all the dtmf actions and it still happens
23:03.03tompawit looks like it always happens when the ooh323 is not in normal, "connected'ish" state
23:03.30tompawbecause when I dial it and quickly hang up, like in 1 second even before far end responds with something
23:03.34tompawit also ignores 'h'.
23:04.19tompawlet me try with sip, maybe I'm simply very stupid.
23:04.44buttons840Again, I'm new, but i'm sure you've put a simple verbose statement in the h extension to ensure that is actually is being ignored?
23:05.05tompawyes, it's a mysql query.
23:05.52buttons840well, i don't know about those, so i can only simpathise :\
23:07.22tompawhmmm
23:07.24tompawsame with sip.
23:07.28tompawvery, very strange.
23:08.26buttons840Question: I'm still learning, and I want to experiment with the AMI.  How would a program running on the same computer as asterisk best interface to the AMI?  Just loop back a telnet session and go from there?  Just looking for someone to point me in the right direction and I'll research it further on my own.
23:08.26Nuggettelnet is eeeeeeevil!
23:08.30ruben23hi, can hear cient side on my calls with this logs http://pastebin.com/m56d9bcb7
23:08.48ruben23hi, cant hear cient side on my calls with this logs http://pastebin.com/m56d9bcb7
23:12.06*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler)
23:13.52jksbuttons840, look up tcp sockets...
23:14.21buttons840ami works through tcp correct?
23:15.12jksyes
23:15.44tompawis there a way
23:15.46tompawto somehow debug
23:15.54tompawwhy asterisk ignores the "h" part?
23:17.17buttons840jks, so the only way to interface with the AMI is through TCP?  What if I have a local running script I want to control asterisk (that is, a running process on the same server as asterisk)?  The only way to do this is through tcp, even though it's all the same computer?
23:17.28jksbuttons840, yes
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23:19.55buttons840.call files would work very well for me, but if i understand correctly from voip.org large quantities of .call files do not work reliably?
23:20.04jksI have no idea
23:20.36jksI always use AMI... that works very well
23:20.44Corydon76-digbuttons840: The issue is that the call files fire whether or not you have outbound channels or even bandwidth available
23:21.08freenoseis there something wrong with this lines: exten => _X.,1,NoOp() and exten => _X.,n,Dial(SIP/lesnet_peer/${EXTEN}), I can't get a call out through les.net, console say, rejected extension not found
23:21.13Corydon76-digSo it's very possible for a call file to fire and to fail to generate a call, because you don't have capacity
23:21.32leevbuttons840: maybe http://www.perl.com/doc/FAQs/FAQ/oldfaq-html/Q5.14.html
23:21.53freenoseusing 1.6.1
23:22.25freenoseI just copy those lines from the book, I'm in chapter four
23:22.25jksleev, won't work for AMI
23:22.39*** join/#asterisk simNIX (n=simNIX@156-60.bbned.dsl.internl.net)
23:23.33buttons840Corydon76-dig, so say i have 10 lines and i dump 1000 call files in at once, will it process them one at a time, or will it try and delete most of them?   or will it try some, and then just ignore them from then on, what would happen to the files themselves?
23:23.46tompawgives up and upgrades 1.6.0 -> 1.6.1 praying it will fix the problem.
23:23.51jksbuttons840, it will process them all in one go
23:24.11buttons840so most will end up being deleted without the calls having actually been made?
23:24.22jksbuttons840, depends on what your system is capable of
23:24.29Corydon76-digbuttons840: correct, they will all fire at once
23:24.34jksbuttons840, you can time the files so they are not processed at the same time though
23:25.06buttons840yes
23:25.37jkswhy on earth do you want to create 1000 calls anyways...
23:25.52jayteespam
23:26.05Corydon76-digjks: there's a number of legitimate applications, such as early warning systems
23:26.17jayteethat's true
23:26.20Corydon76-digBut yes, there's also phone spam
23:26.24buttons840actually, 8000 in 4 hours; probably over my head, but a phone admin friend of mine will use my code if i can figure it out
23:26.32jksCorydon76-dig, early warning systems that fire up 1000 calls at once?
23:26.50Corydon76-digjks: Hence the issue with the Asterisk spooler
23:27.08jksCorydon76-dig, what issue?
23:27.17Corydon76-digjks: generating more calls than capacity
23:27.28jksCorydon76-dig, I fail to see why it's a big problem
23:27.34jksit's easy to schedule calls in a number of ways
23:27.49Corydon76-digjks: Eh, it's undesireable that it would fail in that way
23:28.04Corydon76-digI'm always looking for ways to improve Asterisk
23:28.04jkswho says it will fail?
23:28.28Corydon76-digand specifying a maximum number of concurrent call files might be one such helpful option
23:28.34jkswell, you could always introduce a throttler
23:29.15buttons840the call file exists until the call completes?  so if i have 10 lines and i instruct a script to pass no files in unless there are fewer than 10 files it will work?
23:29.17tompawGuys, it looks like it's more general problem. For example: when I Dial() a trunk that responds with an error (no matter what error) after 4 seconds, the "h" part is executed.
23:29.20Corydon76-digSame concept
23:29.30tompawBUT If I hang up after 1 second, "h" is not touched.
23:29.42tompawWhy is it so?
23:30.02Corydon76-digtompaw: No answer
23:30.23Corydon76-digtompaw: it's the difference between caller ending the call and callee ending the call
23:31.02tompawCorydon76-dig: not really, if a caller ends the call __AFTER_ITS_BEEN_ESTABLISHED__, "h" works too.
23:31.20tompawbut if I hang up during the calling process, it's ignored.
23:32.03Corydon76-digtompaw: yep, no answer
23:32.21tompawCorydon76-dig: what does "no answer" mean? That there is no answer to that problem?
23:32.33Corydon76-digbuttons840: as long as your call files are done synchronously, yes
23:32.45Corydon76-digtompaw: the call was not answered
23:33.18Corydon76-dig"h" is short for hangup, and you cannot hangup a call which hasn't been answered
23:33.50tompawCorydon76-dig: hm... but it also happens when I use M(dtmf_macro), and in that dtmf_macro I SendDTMF(), and when I hang up during that process, it also ignores "h"
23:34.01Corydon76-digbuttons840: if you specified Async: yes, then the call file disappears immediately
23:34.17buttons840where is this specified?  a config file?
23:34.25tompawCorydon76-dig: does it mean that using SendDTMF "marks" the call as "not-answered", too?
23:34.26Corydon76-digbuttons840: in the spool file itself
23:34.38buttons840ok
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23:35.40Corydon76-digtompaw: M(foo) is run on the slave channel, not the master channel, so the master channel is where the answer matters
23:36.00tompawI think I know what to do then. The whole thing is about using MySQL backend for virtual "channels" and keeping information which are busy, which are not.
23:36.14Corydon76-digtompaw: the intent is to allow logic in the slave channel to control whether an answer is propagated (and the channels bridged)
23:36.20buttons840the spool file is the .call file?  where can i get more information on this, i hate to rely on the irc, so is there any good source of documentation on it?
23:36.36Corydon76-digbuttons840: sample.call
23:36.43tompawI will simply add extra "middle" state and run a cron every minute to "clear" channels that have been processed for longer than X seconds.
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23:37.06tompawCorydon76-dig: is there a way to capture that "no answer" state?
23:37.28Corydon76-digtompaw: ${CDR(disposition)}
23:38.24tompawwhoa, I didn't expect that
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23:40.25*** join/#asterisk LemensTS (n=customgt@adsl-70-238-131-23.dsl.stlsmo.sbcglobal.net)
23:40.37LemensTSwhat is the default password to the asterisk user?
23:41.56tompawCorydon76-dig: so does this business logic make sense then: http://pastebin.com/d3a0c93f3 ?
23:42.41tompawNo, it's not, it's stupid. Because that Goto will never be executed, will it?
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23:43.20Corydon76-digtompaw: Yes, it can be
23:43.37Corydon76-digtompaw: when there's no bridge, Dial falls through to the next priority
23:43.52tompawCorydon76-dig: right! so actually the goto's NOT required.
23:44.03tompawI just need to free the channel in BOTH the next priority AND the h,1.
23:44.15tompawAnd that should cover all the cases, shouldn't it?
23:45.29Corydon76-digtompaw: well, unless the caller hangs up
23:45.43buttons840covering all cases is a myth
23:45.51tompawif the caller hangs up, it falls into "h", doesn't it?
23:46.05Corydon76-digonly if the call was answered
23:46.18tompawbut if it wasn't, it goes into s,n, doesn't it?
23:46.31Corydon76-dignot if the caller hung up
23:46.49tompawhm... so in that case... what happens?
23:46.50Corydon76-digYou might want to look at func_odbc
23:47.04Corydon76-digIn that case, you have resources that aren't cleaned up
23:47.09tompawbecause that exactly is my case, the caller hangs up before the call is answered.
23:47.17Corydon76-digOr you could just Answer the call before you Dial
23:47.30Corydon76-digAlthough that could screw up your CDRs
23:47.33tompawCorydon76-dig: I actually clean the resources before I Dial, so it's not a problem.
23:47.39tompawCDRs ar not an issue, either.
23:47.56tompawThe problem is - I need to know that call has ended, and mark it in the database.
23:48.05tompawFor no matter which reason.
23:48.34Corydon76-digI think that's the best answer, then
23:48.39Corydon76-digAnswer before Dial
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23:50.36*** part/#asterisk alerios (n=alerios@190.144.75.22)
23:50.56tompawdidn't help :-(
23:51.52buttons840Corydon76-dig, the sample.call was very helpful, 2 questions now:  1) if i set archive to yes, can i set create my own status indicators, perhaps manually adding in a comment?   2) sample.call mentions nothing of the sync vs async you mentioned, what is a spool file?  is it the same as a call file?
23:52.32Corydon76-dig1) I have no idea, and 2) Yes
23:53.59tompawCorydon76-dig: I have now Answer -> Dial -> s,n -> h,1. But everything after Dial is ignored if caller hangs up during the call set up.
23:54.11Corydon76-digOh, it appears Async is only available in AMI, not spool files
23:54.25buttons840ok
23:54.33Corydon76-digtompaw: I don't have an answer for you, then
23:55.11tompawCorydon76-dig: ok, thanks for trying, at least now I know why it doesn't work.
23:56.09tompawMaybe I'll try and add another extension "e" :-)
23:56.39tompawe - Extension called when the call ends.
23:56.55Corydon76-dig"e" is already taken for "Exception"
23:57.10tompawhm...
23:57.30tompawto be honest, I am really surprised noone has asked for this before, it seems like a very useful tool.
23:57.53tompawI mean, there must a part of the code responsible for doing "something" when the call ends.
23:58.04tompawLike updating CDR for example.
23:58.21buttons840I'll end by asking your advice if i may.  I'm attempting to create an appointment reminder that will send thousands of automated calls.  I'll have a t2 line.  Shall I spool call files or create or use the AMI?
23:58.36tompawMaybe it would be possible to hack into that part and somehow register one final handler to be parsed.

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