IRC log for #asterisk on 20130612

00:03.04ChannelZOh lookie here, /etc/newt/palette linked to /etc/alternatives/newt-palette .. old systems don't even have /etc/newt
00:05.32ChannelZYAY, death to purple!
00:07.53Rhomberhaha, nice :)
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00:33.34carrarhrmm
00:36.03carrarHow does one influence the URI in the Diversion header that Asterisk adds if send_diversion=yes
00:36.17carraron a 302 redirect response from a sip phone
00:36.38carrar(aka forward)
00:37.03WIMPyForwarding is server based.
00:37.21carrarthat doesn't answer my question
00:38.09carrarmore specifically the number used in the diversion header, instead of a device name
00:39.38carrarthe user part of the URI
00:40.57igcewielingcarrar: remove the header and regenerate it on the outbound call leg
00:41.04carrarhaha
00:42.02carrarthe fix for this situation is to just set send_diversion=no, since the formatting of the Diversion URI is wrong in thie casae
00:42.21carraror the values I should say
00:43.00carrarMost people are not going to want Asterisk to use SIP Device names in the user section of a Diversion URI
00:43.09Ta^3I found that Asterisk 11.4 consumes more than 100% CPU resulting in choppy sound where 11.3 in the same production environment and same configuration not. I'm not able to run it inside a valgrind as it's on production but I would like to figure out what could be wrong.
00:45.33carrarIt would be nice if there was a way to specify what the user part of the Diversion URI should be in the event that asterisk adds that header.
00:45.48carrarthat would be usefull
00:45.55carrarperhaps there is, but I'm not seeing it
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00:59.37LokiScarletexit
00:59.39LokiScarlet...
00:59.43LokiScarletOops I'm not in bash
00:59.46LokiScarletXD
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01:31.59carrarok looks like 'REDIRECTING' solves this
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02:00.16igcewielingcarrar: does Asterisk send a sip: URI or a tel: URI?
02:04.08carrar11 uses sip:
02:04.19carrarI know they had some issues in the past with tel:
02:04.51carrarand switchvox was missed up recently too using both formats
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02:08.00igcewieling*nod*  I have to rewrite the tel: stuff in my AGI and then set the proper stuff
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04:25.03n8ideasQuick question... in dialplan, I read a string into a variable that itself contains a variable... for instance http://test.com/check.php?${NUM}
04:25.15n8ideascan I force Astersk to evaluate ${NUM} ?
04:25.48n8ideasI seem to recall having done this before, but am blanking on it now
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04:38.17igcewieling"core show function EVAL"
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04:51.45n8ideasEVAL is your friend
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08:31.58askinHi, i am using call transfer with voicexml (voiceglue). When transfer is failed, i can't execute _else_ condition in vxml. Call is holding and never hangup. Callfile + dialplan: http://pastebin.com/CXntJHup , VoiceXML: http://pastebin.com/rJQTEzaC , Asterisk Debug Log: http://pastebin.com/yV6n0y1Q
08:32.15askinWhat should i do
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08:39.35kaldemaraskin: see lines 107 and 108 in the CLI output.
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08:40.52kaldemaraskin: you usage of call files needs fixing. first write the call file in to a temporary file and then put it in the outgoing directory with an atomic file operation, such as mv from the same filesystem.
08:41.44askinkaldemar: cp /root/doruk.call /tmp; chown asterisk.asterisk /tmp/doruk.call; mv /tmp/doruk.call /var/spool/asterisk/outgoing; tail -f /var/log/asterisk/full
08:43.27kaldemarand then your dialplan does something...
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09:05.46askinkaldemar: New asterisk log, http://pastebin.com/XKxNR7Mi, new dial plan: http://pastebin.com/StfWX3cB. Problem is same. Call never hangup after call transfer.
09:08.45kaldemaraskin: you're getting "503 Service Unavailable" from the destination your AGI is trying to dial.
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09:18.30bulkorokhi
09:19.52askinkaldemar: provider is busy. This log is clean:
09:20.22askinkaldemar: http://pastebin.com/hRWjgiP4
09:21.59askinkaldemar: i want to play next vxml after i get response 480 but i get nothing.
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09:24.10kaldemaraskin: inspect the DIALSTATUS variable after you dial and decide what to do based on that.
09:30.28askinkaldemar: thanks
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10:17.42blodan-workwhere can i find the correct mysql table structure for queue_members? There is no sql file for that table in contrib/realtime/mysql/
10:19.51kaldemarblodan-work: check the postgresql directory.
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10:20.21fredericveWhat would be the best way to clear a dialgroup?
10:21.50fredericvewith the DIALGROUP function I can only add and remove devices
10:22.00fredericvebut there is no way to really clear it
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10:25.21fredericveI'm trying now with the DB and DB_DELETE functions but no luck so far
10:25.27fredericveasterisk 1.8.13.0 btw
10:31.00kaldemarfredericve: Set(DIALGROUP(group)=)
10:32.27kaldemarfredericve: you can also set the whole group to what you want on a single command if you omit the add/del. Set(DIALGROUP(group)=foo/bar)
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10:43.41CommaCrazyhi all
10:45.00CommaCrazyquestion, has anyone had any experience with Linksys SPA-942 IP phone?? I got one and it retries registration every 45 seconds (when it succeeds to register, when it fails registration it goes to 3500 sec retry period)
10:45.17CommaCrazyand after a few re registration it fails to register
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10:59.50fredericvekaldemar: thanks! that did it
11:00.21fredericvethe documentation is lacking that information
11:00.46fredericveah no
11:00.50fredericvemust have missed it
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11:04.09izbushkahi
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11:07.48izbushkai have a problem with ussd request by chan_dongle (USB 3g modem). It's imposible to send request with two "*" in it (dongle ussd dongle0  "*104*222#") Operator reports about incorrect request. What's wrong?
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11:27.48gavimobilehow can I change the refresh time listed for a registry in sip show registry? the default seems to be 105 but one of them is 45
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11:44.33kaldemargavimobile: set the expiry to what you want: register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
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11:45.16gavimobilekaldemar: I've changed that and reloaded sip but it doesn't seem to change when I run sip show registry from cli
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12:02.09kaldemargavimobile: show what you did and proof of that.
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12:03.40gavimobilewill do in just a moment
12:05.30gavimobilekaldemar: here is what I did
12:05.30gavimobilehttp://pastebin.com/hZNAiJ8P
12:05.43gavimobilewhat else can I prodive?
12:05.55gavimobileof course I restarted sip
12:05.59gavimobilesip reload*
12:07.01kaldemarthat should work. maybe their allowed maximum value is 45 seconds.
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12:13.41gavimobilekaldemar: don't know, but they do say that this is how one should configure their config files http://www.callcentric.com/support/device/asterisk/1_6
12:14.04gavimobileto save you some reading they mention to add session-timers=refuse session-expires=180 session-minse=90 session-refresher=uas. not sure what they do or if this is related
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12:16.49kaldemargavimobile: unrelated
12:17.29kaldemargavimobile: pastebin a sip debug of a registration to them.
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12:21.31LokiScarletGonna sound random as all hell but does the trunk happen to support dmix in chan_alsa?
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12:26.21din3shhelo
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12:36.34gavimobilekaldemar: ok, thanks
12:36.49gavimobileon a different note im having trouble receving incoming calls I need an extra eye here http://pastebin.com/zCcg4AUG
12:39.53gavimobileI have sip set debug on and its to much information to catch so I tried sip  sip set debug peer callcentric but I don't see anything wrong except for "Sending fake auth rejection for device <sip:13057714673@66.193.176.35>;tag=3580029189-597386" http://pastebin.com/xfwB5nk5
12:41.42gavimobilesip show peers && sip show registry http://pastebin.com/YQ398iwS
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12:52.23kaldemargavimobile: there are no REGISTER requests in your sip debug, so it's quite useless.
12:52.47gavimobilekaldemar: so I need to do a full sip set debug on
12:52.53gavimobileand try to output whatever I can?
12:54.42gavimobilekaldemar: ill try to get a better output
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12:58.40gavimobilekaldemar: this is what I got. please let me know if its helpfup http://pastebin.com/
12:58.43gavimobileoops
12:58.54gavimobilehttp://pastebin.com/YSbkX8LS
13:06.02kaldemargavimobile: useless. enable debug and do a "sip reload".
13:06.32Cuznergood morning fellow asteriskers
13:07.29gavimobileon centos how can I output it to a file?
13:09.05kaldemargavimobile: asterisk -vvvr | tee /tmp/ast_output.txt
13:09.06gavimobilekaldemar: this pastebin includes sip set debug on then a sip reload than a sip debug off pastebin.com/bvxWpcSE
13:09.19gavimobilekaldemar: that's goo to know for next time
13:09.37carrar*Y*A*W*N*
13:09.52carrarmust
13:09.53carrarget
13:09.56carrarespresso
13:11.38gavimobilehere is the full pastebin logged http://pastebin.com/0vB50crD
13:11.44LokiScarletgives carrar a can of NOS
13:11.45gavimobilewith the call
13:12.15carrarheh
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13:12.51msaraivaHey all
13:13.04kaldemargavimobile: you're not getting a reply back from callcentric to your register. with the ipcomms register you'll see that they answer you with a 200 OK with an Expires header.
13:13.05LokiScarletSometimes you need more than caffeine. NOS has stuff to optimize the process and make the caffeine much more effective
13:13.06msaraivamemory show summary -> 3696910 bytes allocated (7276 in caches) in 7827 allocations
13:13.19msaraivaactual consumption on top: 1180MB!
13:13.26msaraivaDoes that make any sense?
13:13.57gavimobilekaldemar: so is this NOT  a problem with my config?
13:14.03kaldemargavimobile: no
13:14.59gavimobileso what could be wrong with my config?
13:17.31kaldemardoes not compute. i just told you this is not a problem with your config.
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13:19.03gavimobileno + no == yes!
13:19.16gavimobilekaldemar: alright! thanks man
13:19.47gavimobileI see that iptables is blocking them, why might this be?
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13:22.48gavimobilekaldemar: could you give it another look, I unblocked them from iptables and did a new output http://pastebin.com/w3ADfcg5
13:24.51gavimobileI think it has something to do with "No matching peer for '13057714673' from '204.11.192.159:5060'"
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13:26.37kaldemargavimobile: i have no idea about your iptables.
13:26.55gavimobilekaldemar: did you give it another look and see anything different this time?
13:27.04gavimobilethere is some bad authorization somewhere
13:27.11kaldemargavimobile: i was looking at registrations, this is a new issue.
13:27.55gavimobileI assume your talking about the line I posted in quotes above
13:28.00kaldemarm: <sip:5169261796@81.218.196.30:5060>;expires=60 <-- line 432, they force your expiry to 60 seconds.
13:28.09kaldemargavimobile: no.
13:28.59kaldemargavimobile: your host on the callcentric peer is wrong.
13:29.24gavimobilekaldemar: in sip.conf?
13:31.04kaldemargavimobile: that's why it doesn't match. their calls come from 204.11.192.159 and you told asterisk that their address is callcentric.com which resolves to 9 different addresses.
13:31.23kaldemargavimobile: yes in sip.conf. peer definitions are there.
13:31.29gavimobilewow, amazning
13:32.13gavimobileso I need to change callcentric.com to the ip address you gave me, and make a new one for outgoing calls?
13:33.10kaldemargavimobile: you need to change the host to a host or an ip address that they use to send calls to you. are they using more than one address?
13:33.43gavimobilekaldemar: I wish I could tell you, but this is all the information I have right now for settings up their trunk http://www.callcentric.com/support/device/asterisk/1_6
13:33.44kaldemargavimobile: changing the ip might work for that call, but the next one may fail again if it comes from a different address.
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13:34.05gavimobiletried the ip change in host= but didn't work
13:34.13kaldemargavimobile: that's why you should ask them. their example configs won't help here.
13:34.29*** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net)
13:34.33gavimobileok, thanks for the lea kaldemar
13:34.39gavimobileyou got me pointed something...
13:36.06*** join/#asterisk cyborg-one (~cyborg-on@130-0-63-30.broadband.tenet.odessa.ua)
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13:51.51KattyGOOD MORNING ASTERISK.
13:52.04zoid_Hi, I'm writing an AGI script and the following came out: is it possible to know from the script when asterisk fails to play some audio (maybe because of it not existing or being in the wrong format) without stating the audio from the script previously?
13:52.36zoid_Doing so would be i/o intensive and useless 99% of the time
13:54.57zoid_In the tests I did asterisk prints the corresponding error on the console, but from the POV of the script, it's a success
13:54.58RufusGreetings
13:55.10*** join/#asterisk davlefouAMD (~david@41.225.172.115)
13:56.21RufusMight sound more of a windows question, and if it does I appologise. Windows xp, 2 NIC's. One in lan (via wifi), one secondary, with a SIP fone hooked up. Main nic las 192.168.1.36, second one is 192.168.0.1, an SIP phone is 192.168.0.2. the asterisks erver is 192.168.1.13 . How do I get SIP phone to connect to asterisk server?
13:57.56*** join/#asterisk Carlos_PHX_ (~Carlos@ip68-228-216-137.ph.ph.cox.net)
13:58.33Kattyyour subnet mask
13:58.39Katty255.255.0.0
13:58.47Kattythat way you can see 1.1 and 0.1
13:59.03Rufusoh that makes perfect sense
13:59.04Rufusthanks
13:59.36jmetroor a router..
13:59.40jmetroor a layer 3 switch
13:59.42Kattyor that.
13:59.50Kattywhatever suits your fancy pants.
13:59.52Kattyhugs jmetro
13:59.53Rufusthat's not possible, sadly
14:00.08jmetroKatty: My pants are quite fancy :3 {hugsback}
14:00.31Kattydare i ask....what are you wearing?!
14:01.48jmetroMeine work khakis :3
14:01.49bombevhi
14:02.08[TK]D-FenderYou don't change your mask
14:02.24[TK]D-FenderEach subnet is it's own thing.  Something has to act as a gateway
14:02.25Penguin[TK]D-Fender: its
14:02.33[TK]D-Fendero/
14:03.26kaldemarRufus: make the xp machine route. how to do that is a matter of another forum.
14:03.45bombevwhy i am getting this message
14:03.46bombevWARNING[7741] chan_sip.c: Retransmission timeout reached on transmission 52b99cd42cc15b313e79b69445e2c69c@212.71.129.36:5060_b2b-1 for seqno 10 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
14:03.46bombevPacket timed out after 6400ms with no response
14:04.53[TK]D-Fenderbombev: And you should be looking at the comms that are timing out.....
14:06.18*** join/#asterisk timahvo1 (~rogue@41.212.120.45)
14:12.22*** join/#asterisk geroz (~zerog@static-84-242-70-218.net.upcbroadband.cz)
14:12.36*** part/#asterisk geroz (~zerog@static-84-242-70-218.net.upcbroadband.cz)
14:13.05jmetroKatty: and what are you wearing that your pants arent fancy, anyway
14:14.47Kattythey're just dress pants from kohls.
14:16.11*** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger)
14:16.11*** mode/#asterisk [+o pabelanger] by ChanServ
14:16.29jmetroOh. so the same thing as me XD which makes us both COMMONERS
14:17.47*** join/#asterisk serafie1 (~erin@nat/digium/x-zooutepgpfzmulcs)
14:18.21Katty^_^
14:24.06Kobazugh
14:24.15Kobazstill having pickup problems
14:24.23Kobazbut sip debug wasnt going at the time
14:31.51*** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn)
14:33.33pzncan anyone help me to debug a hardware with ancient asterisk 1.2.13? I changed Dial(sip/123) to Dial(sip/123,30,m(musicclass)) but it is not playing music
14:34.09zoid_files a bug :)
14:34.16*** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell)
14:34.16*** mode/#asterisk [+o sruffell] by ChanServ
14:34.24[TK]D-Fenderbugs a file
14:34.27[TK]D-Fender:O
14:34.34fileswozzles
14:40.06igcewielingpzn: Apparently the answer is "no"
14:40.34pznigcewieling, yes... no problem... I think I'm on my own now :-)
14:41.11pznigcewieling, maybe throw the hardware on trashbin and buy a newer thing :-D
14:41.28igcewielingor install a newer version of Asterisk
14:43.43pznigcewieling, I'm afraid or rewriting the flash memory... maybe I lose the hardware.
14:44.24igcewielingpzn: next time get a real PC for your PBX.
14:44.44igcewielingwhich of the crap "appliances" do you have?  PIKA?
14:44.53*** join/#asterisk lodac (~lodac-h3@71-83-12-142.static.aldl.mi.charter.com)
14:46.26pznigcewieling, don't know it. I'm connected via network. it is a PBX that controls 8 extensions only...
14:46.54pznigcewieling, anyway... does not worth the time wasted. I'll recommend a new thing.
14:47.20WIMPycat /proc/cpuinfo;lspci
14:50.00pznWIMPy, "uname -a" shows powerpc architecture; cpuinfo says "msc71xx processor"; "lspci" does not exist.
14:50.40pznWIMPy, really... I'm convinced that this will be too much effort for lower results... I'll replace this for a PC.
14:51.01WIMPyOh, not even ARM. uncool.
14:51.37WIMPy(so would I)
14:54.24*** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl)
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14:54.50pzn:-)
14:56.18*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw)
15:01.57*** join/#asterisk vlad_starkov (~vlad_star@109.188.124.190)
15:08.49*** join/#asterisk b0ot (93b13ded@gateway/web/freenode/ip.147.177.61.237)
15:09.16b0otIs there a way you could setup asterisk to call a rtp audio stream (unicast or multicast)
15:10.35*** join/#asterisk vlad_starkov (~vlad_star@109.188.124.190)
15:12.01pabelangerAny debugs I can enable on DAHDI to see why a PRI's b-channels are restarting every 1 hour?
15:13.10Kobazthey are configured that way
15:13.10WIMPyYou should see the RESTART messages in the pri debug.
15:13.29Kobazif you dont want the restart, turn it off
15:13.39WIMPyOr is your side configured to do so?
15:14.15WIMPyYes, one of those strange defaults.
15:15.17pabelangerKobaz: turn what off? There is a setting in asterisk that does that?
15:15.32WIMPychan_dahdi.conf.
15:15.37pabelangerand ya, will have pri debugs going
15:15.44WIMPysearch for restart.
15:15.44Kobazyeah
15:15.52Kobazit's like restart=yes or something
15:15.53Kobazone of those
15:16.00pabelanger;resetinterval = 3600
15:16.01pabelangerodd
15:16.04Kobazyeah
15:16.04pabelangernever new about that
15:16.05Kobazthat one
15:16.09pabelangercool
15:16.13pabelangerchecking now
15:16.32Kobazsome broken pri implementations need the b channel restart
15:16.42Kobazdepends who you're talking to on the other end
15:17.34pabelangernope, not set
15:17.42pabelangerso, that is good asterisk is not doing it
15:17.49WIMPySet it to 'never'.
15:17.49pabelangerokay, enabling PRI debugs now
15:18.03Kobazyeah it might have a defaultvalue if its not set
15:18.22WIMPyI think it has.
15:18.51pabelangerYa, will check code too
15:22.21pabelangerokay, enabled. Will see what happens in 1 hour
15:23.04*** join/#asterisk blee (~blee@50-89-200-235.res.bhn.net)
15:23.15Kobazwhat's a ring cycle, 4 seconds?
15:23.32WIMPyWhich country?
15:23.39Kobazusa
15:23.52Kobaz4 on 2 off or something
15:23.54Kobaz6 seconds
15:24.10WIMPyhas a feeling the answer might be in indications.conf.
15:24.19ChainsawIt will be, yes.
15:24.28Kobazah r ight, yeah 6 seconds
15:24.30pabelangerYa, 6
15:24.32Kobaz2 on, 4 off
15:24.33Kobazi was close
15:25.19ChainsawAll I know is that it sounds wrong when it's not a double ring :)
15:25.52*** join/#asterisk vlad_sta_ (~vlad_star@nat.canmos.ru)
15:32.34*** join/#asterisk MrQuist (~rob@83.232.96.217)
15:32.39MrQuistHey guys
15:32.56MrQuistI need some help - i want to play a .mp3 file in an extension
15:33.20zoid_MrQuist: use mpg123
15:33.28MrQuisti put the .mp3 in /var/lib/asterisk/sounds/en/enrise
15:33.45MrQuistwhere enrise is my custom folder
15:34.08MrQuistzoid_, is that enough?
15:35.32pabelangerKobaz: WIMPy: Thanks, just seen the RESTART message in the PRI log, so far end is doing it
15:35.40pabelangeropening an issue with telco and finding out why
15:35.47zoid_MrQuist: see http://www.voip-info.org/wiki/view/Asterisk+cmd+MP3Player
15:37.07Kobazpabelanger: it's an option
15:38.25[TK]D-Fenderjsut install format_mp3.so
15:38.35[TK]D-Fenderand make sure it's non-VBR
15:38.42[TK]D-Fenderand no ID3
15:39.09ChainsawNon-VBR MP3, wow. They still make those? ;)
15:39.23WIMPyHopefully not.
15:39.44zoid_MrQuist: just curious, why mp3?
15:39.55MrQuistOh i had some mp3 files laying around
15:39.58MrQuistbut thats not a big deal
15:40.02MrQuisti've just changed them to .wav
15:40.05MrQuistlets see how that works
15:40.17MrQuistquality isn't really important
15:40.18WIMPyMight make sense to convert them once instead of on every call.
15:40.21ChainsawMrQuist: It'll like that better, wave is handled natively.
15:40.25WIMPyAnd might be more legal as well.
15:40.33MrQuistyep
15:40.46ChainsawWIMPy: More legal. That's in the "almost pregnant" category, isn't it.
15:41.04MrQuistok
15:41.06MrQuistanother question
15:41.20WIMPyWell if you convert the files at home, that's legal.
15:41.34MrQuistI have it as a .wav now.
15:41.42WIMPyAnd who will prove you did it at work? (if you did)
15:41.46MrQuistbut it looks like the system isn't cathing it?
15:42.02zoid_MrQuist: it's better to convert the files to the same codec the call will be using
15:42.10[TK]D-FenderMrQuist: Right now it looks like nothing....
15:42.17MrQuistExecuting playback(sip/0000FFFFAAAA, 'folder/my-file') in new stack
15:42.26MrQuistunder there is the same with tt-somethingwrong
15:42.39MrQuistand under there is "playingg tt-somethingwrong.gsm" (language en)
15:42.41ChainsawMrQuist: You answered the channel first right?
15:42.55[TK]D-FenderMrQuist: Show us complete actual details....
15:42.57MrQuistyes i can hear the something wrong. No problem
15:43.10Kobaz2013-06-12 11:42:39] NOTICE[3724] chan_sip.c: Correct auth, but based on stale nonce received from '"Stephen Kushner" <sip:6850@192.168.51.1>;tag=CBEFDBE9-A1A57004'
15:43.18WIMPyWhat Asterisk version?
15:43.25MrQuisthttp://pastebin.com/YU2zQ8vT
15:43.29MrQuist11.4.0
15:43.32Kobazgetting a lot of those
15:44.01[TK]D-FenderMrQuist: COMPLETE details...
15:44.05ChainsawKobaz: I have loads of clients that do this. Consider not running with SIP debug on.
15:44.11MrQuistcomplete..? what are you missing [TK]D-Fender
15:44.13ChainsawKobaz: As you won't see them then.
15:44.16Kobazhah
15:44.21Kobazi need sip debug today
15:44.31ChainsawKobaz: (core set debug 10 && core set verbose 10 is fine, but sip debug means "stale nonce" chattiness)
15:44.52[TK]D-FenderMrQuist: Folder dump with the files visible. "file-type dump to prove the contents match.  Some statement that the OTHER prompt played at all.
15:45.20MrQuistAnd me telling you isn't enough?
15:46.17zoid_you can set sip debug for one peer in particular
15:46.21[TK]D-FenderMrQuist: No.
15:46.54ChainsawMrQuist: Fender doesn't trust you enough to summarise Asterisk errors yet.
15:46.59ChainsawMrQuist: Please give him the raw data.
15:47.01MrQuistWhatever you want ->
15:47.38Kobazheh
15:47.46[TK]D-Fender"But I swear I did everything right" - First claim from the guy who did it wrong.
15:47.47Kobazfender doesnt trust anyone
15:47.49Kobazthat's the bottom line
15:47.55MrQuisthttp://pastebin.com/9R2R8Lz6
15:47.57[TK]D-FenderYou want an autopsy?  Give us a BODY
15:48.44MrQuistContents of extensions.conf -> http://pastebin.com/XqFs8ZJc
15:48.50[TK]D-FenderMrQuist: "ls-la /var/lib/asterisk/sound/WHATEVERSUBFOLDER", and "stat" on the file itself
15:49.12[TK]D-FenderMrQuist: I don't trust permissions, file sysized, that the contents match the file extension .... or anything else for that matter.
15:49.30[TK]D-FenderI find problems because I don't go around assuming things are just "right"
15:49.50[TK]D-FenderWhen things aren't working, trust is for chumps
15:50.02MrQuistCould it be the file ownership?
15:50.10MrQuistchown root:root myfolder/ -R
15:50.20MrQuistwhereas its currently "Vagrant"
15:50.32[TK]D-FenderAsterisk running as root is HIGHLY non-recommended
15:50.42MrQuistfor the time being
15:50.45MrQuistdevelopment stage
15:51.02MrQuistalso
15:51.02MrQuistls -al
15:51.02MrQuisttotal 580
15:51.02MrQuistdrwxrwxr-x 2 root root   4096 Jun 12 15:39 .
15:51.02MrQuistdrwxr-xr-x 9 root root  16384 Jun 12 15:28 ..
15:51.03MrQuist-rw-rw-r-- 1 root root 250028 Jun 12 15:39 kantoor-gesloten.wav
15:51.04[TK]D-Fenderand changing the ownership of the files without even confirming who * is running as ... is also dangerously complete.
15:51.06MrQuist-rw-rw-r-- 1 root root 317420 Jun 12 15:39 paasweekend-gesloten.wav
15:51.10MrQuistoops
15:51.26[TK]D-Fenderincomplete*
15:51.30MrQuistAlso, asterisk is running as user asterisk
15:51.44[TK]D-FenderSo why should the files be oewned by ROOT?
15:51.44MrQuistWhat about i'll just give you the whole VMDK?
15:51.54MrQuistthats not the issue right now
15:52.11[TK]D-Fendermake the basics sane, prove what's in the file.
15:52.15[TK]D-FenderThen you can move forward.
15:52.22MrQuistI would like to find out why asterisk isn't picking up my .wav files, but is picking up .gsm files.
15:52.28[TK]D-FenderAnd also didn't recall seeing the confirmation that the other file played
15:52.42[TK]D-FenderBelieve me ... I do not trust your wav file for one second
15:52.54MrQuist17:42:57 <MrQuist> yes i can hear the something wrong. No problem
15:52.58[TK]D-Fenderbecause the odds of you having given it the wrong SPECS is very high
15:53.21[TK]D-Fenderso go prove the exact format
15:54.11MrQuist[TK]D-Fender, RIFF�^C^@WAVEfmt ^P^@^@^@^A^@^A^@"V^@^@"V^@^@^A^@^H^@data<80>^C^@<80><81><80><80><80><80><81><80>~<80><80>
15:54.25MrQuistRaw data.
15:54.25vlad_sta_Question: Anyone know how to change the default shell command that Asterisk uses to combine In and Out audio streams for MixMonitor application?
15:54.32[TK]D-FenderNor with a readable format ...
15:54.35[TK]D-FenderNow*
15:54.44MrQuistHow do you want to read .wav?
15:58.19MrQuistso, i've compied vm-password.gsm to myfolder/vm-password.gsm
15:58.27MrQuisti mean
15:58.34MrQuist<PROTECTED>
15:58.49MrQuistsame => n,Playback(myfolder/kantoor-gesloten) doesn't work
15:58.56MrQuistsame => n,Playback("myfolder/kantoor-gesloten.gsm") doensn't work
15:59.50MrQuistHow can i tell what is going wrong with opening a file?
16:00.11MrQuistAll the debug info tells me is "EXECUTING PLAYBACK"
16:00.25GreenlightAre you sure the format is correct?
16:00.26[TK]D-Fendershow us the file along with the full path, and the * settings to confirm the path * start in for them "core show settings"
16:00.38*** join/#asterisk tech_travis (~Travis@174.46.237.96)
16:00.38newtonrMrQuist: what level of verbosity do you have VERBOSE and DEBUG set to?
16:00.46MrQuistnewtonr, 10 and 10
16:00.53MrQuistGreenlight, its a .wav
16:01.08[TK]D-FenderMrQuist: Go open it in some audio app that will confirm the precise encoding.
16:01.29[TK]D-FenderAnd for your other ussue show us precisely where the files are, what they look like, your core settings for path, etc
16:01.36MrQuistUncompressed 8-bit PCM audio
16:01.40MrQuistMono
16:01.46MrQuist22050 Hz Sample Rate
16:01.49MrQuistjust as i converted it
16:01.57GreenlightThat sample rate is too high
16:02.06GreenlightIsn't it ?
16:02.14MrQuist22050 is kinda low already
16:02.18newtonrMrQuist: has to be 8kHz IIRC
16:02.19MrQuist44100 = cd
16:02.26GreenlightI thought 8khz too
16:02.33GreenlightIf memory serves
16:02.42MrQuist11025 Hz ?
16:02.51MrQuistOr just 8000hz?
16:02.59newtonr8000
16:03.02Greenlight8000Hz
16:03.30[TK]D-Fender[11:52][TK]D-Fenderbecause the odds of you having given it the wrong SPECS is very high
16:03.31pabelangerWIMPy: Kobaz: So what happens if a PRI gets RESTART during an active call, I assume it waits for the channel to clear before dropping the call and restarting?
16:04.15GreenlightIf you want higher quality you can use other formats: https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Codecs+and+Audio+Formats
16:04.54WIMPypabelanger: I don't think so. The idea is to get it to a defined state if somethign went wrong. So it should release the call.
16:05.06MrQuisthow is 22050 hz too high?
16:05.08MrQuistah well
16:05.10MrQuistlets try 8000
16:05.41[TK]D-FenderMrQuist: 8000.  No substitutes
16:05.47Greenlight22050 is too high because 22050 > 8000
16:05.54MrQuistioooooh
16:05.56MrQuist-_-
16:06.21MrQuistnope, doesn't work either.
16:07.02GreenlightThen perhaps you've got something else wrong as well - follow [TK]Fenders earlier requests to diagnose that
16:07.18MrQuisti just don't know why this whole system works, except for telling me whats wrong
16:07.49igcewielingAsterisk should not drop calls, even when it wants to restart the channel.  It will wait for the channel to be idle for the restart time before restarting the channel.
16:07.59[TK]D-FenderMrQuist: Saying "it doesn't work" is not showing us all the fine details, for which I had asked you for several very specific things.
16:08.19*** join/#asterisk smkelly (~smkelly@mykonos.smkelly.org)
16:08.22igcewielingset bchannelreset=no or whatever the option is
16:09.23WIMPypabelanger: Confirmed. "A RESTART message is sent [...] in order to return channels or interfaces to the Null state."
16:09.41newtonrMrQuist: pastebin an asterisk log showing the playback with verbose and debug turned up to 10. also include a "core show channel <channelname>" of the channel running the Playback *while it's executing playback*
16:11.57GreenlightMy money is on wrong format still
16:12.01MrQuistGood. [TK]D-Fender, newtonr -> http://pastebin.com/C89kgsr6
16:12.15MrQuistGreenlight, i copied a working .gsm to the same folder, and even that didn't work.
16:12.37GreenlightWas there a .wav with same name ?
16:12.48MrQuistyes probably
16:13.07GreenlightSo that's not a great test if it tried to play the wav still
16:13.19*** join/#asterisk blee (~blee@50-89-200-235.res.bhn.net)
16:13.26newtonrMrQuist: doesn't look like you have debug messages enabled to the console.. see "logger show channels" to see if you do
16:13.53[TK]D-Fender[11:51]MrQuistAlso, asterisk is running as user asterisk <- your files should not be owned by "vagrant"
16:13.53newtonrMrQuist: then add it to the console line in logger.conf if you don't have it
16:14.19[TK]D-FenderMrQuist: And I'd like to see that the files are now in the proper format...
16:14.47newtonrMrQuist: and you didn't get the "core show channel <channelname>" , but you might still be working on that
16:14.57Kobaz[TK]D-Fender: now that i have sip debug on.  no one is reporting any problems
16:15.16MrQuistnewtonr, it did get the .gsm file in the same folder
16:15.25[TK]D-FenderMrQuist: Along with actual full call output...
16:15.36MrQuistthis _is_ the full call output
16:16.05[TK]D-Fenderwhich PB?
16:16.13GreenlightFirst, fix the permissions, that's obviously going to cause issues
16:16.23MrQuist[TK]D-Fender, the password.gsm file also works while the owner is vagrant.
16:16.39Kobazpabelanger: b channels will not restart while in use
16:16.49MrQuistnewtonr, how can i find out what channel i'm using?
16:17.06MrQuistnvmd
16:17.18newtonrnewtonr: when the channel is up and running you'll be able to tab complete at the end of "core show channel .." and see it
16:17.19Kobazpabelanger: it'll just get it on the next cycle if it's not in use by then
16:17.47Kobazpabelanger: unless you're having a specific problem, i really wouldnt worry about the b channel restarts, it's fairly standard
16:17.58*** join/#asterisk thehar (thehar@diddlebox.thehar.com)
16:18.09newtonrMrQuist: if you enable debug to the console in logger.conf and reload your logger it may help you see what is going on..
16:20.38MrQuistwtf
16:20.53MrQuistnewtonr, why is there no debug information when i say "core set debug 10"
16:21.00MrQuistbut is there when i enable debug in logger.conf?
16:21.29GreenlightMrQuist: Can you "sox -i <file>" please?
16:21.36MrQuisti have here;
16:21.41newtonrMrQuist: "core set debug X" is setting the verbosity level for debug. logger.conf is setting which messages get written where.
16:21.44MrQuist[Jun 12 16:20:30] WARNING[31355][C-00000003]: format_wav.c:126 check_header_fmt: Can only handle 16bits per sample: 1
16:21.45MrQuist[Jun 12 16:20:30] WARNING[31355][C-00000003]: file.c:398 fn_wrapper: Unable to open format wav
16:21.56MrQuistcongrats, [TK]D-Fender
16:21.59*** join/#asterisk fastDuane (~gt2013@mail.pserver.com.py)
16:22.50[TK]D-FenderMrQuist: I'm not sure what part of "go open the file in some audio program and prove its specs" was unclear before.  Do I trust your file?  NO.
16:23.02GreenlightKnew it was going to be the wrong format
16:23.14MrQuist[TK]D-Fender, i did. I showed you the stuff.
16:23.21GreenlightIf only the required format was listed online or something
16:23.29MrQuist-_-
16:24.08[TK]D-FenderGreenlight: Something like ... http://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk
16:24.10[TK]D-Fender?
16:24.25MrQuistthat was the joke [TK]D-Fender
16:24.26GreenlightOh gosh! That must be a new site is it?
16:24.40MrQuistalso, Bits per sample: 8
16:24.43MrQuistSample rate: 8000hz
16:24.49MrQuistCodec: PCM U8 (ARAW)
16:24.51[TK]D-FenderGreenlight: Decrepitly so!
16:25.10[TK]D-FenderMrQuist: 16bit <- go reconvert your original
16:26.25Greenlight8bit/8k would sound awful too I'd suspect
16:26.48[TK]D-FenderWe aim for "just awful enough"...
16:26.59GreenlightHeh
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16:31.59MrQuistGreenlight, thanks a bunch :)
16:32.14MrQuist[TK]D-Fender, thank you as well
16:32.22MrQuisti reconverted it with "sox"
16:32.23MrQuistit works now.
16:32.44GreenlightAsterisk is very picky about formats
16:32.45[TK]D-FenderMrQuist: Trust nothing, double-check everything, and compare to something that works.
16:33.17newtonrMrQuist: and make sure debug is turned on ;)
16:36.25MrQuistnewtonr -> thats what would have helped me out. I thought that "set debug on maximum level" would help. But it dind't
16:36.36MrQuist[TK]D-Fender, i know how that works.
16:36.45MrQuistGreenlight, indeed, kinda weird if you ask me.
16:39.45jalewisanyone know if the 1TE133F is actually out and shipping yet?
16:41.06drmessanoI saw the email.  That card is unreal.  I would imagine the next generation card will have one 8 pin DIP IC on it, and some RJ-45s.
16:45.15newtonrjalewis: drmessano: yeah they are being sold and shipping already
16:45.23jalewisalso, I see in the specs both the current 1 port T1 cards are half height, but all pics are with a full height bracket...anyone know if they include half a height bracket for use in 2U non-riser rackmount systems?
16:45.57igcewielingdrmessano: how exactly does it differ from previous cards?
16:46.27jalewisvery sparsely populated PC board
16:47.17jalewisI think he's saying the next gen will basically be an RJ jack and a single big chip
16:49.45newtonrjalewis: i'll ask one of the sales guys about the half-height bracket
16:49.57drmessanoExactly.. At some point they're going to have to adopt some magnetic DIP headers or something for the EC cards because the connection will be bigger than the entire sum of the chips
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16:50.31drmessanoInsanity
16:50.33sruffellheh…that picture of the new single spans *includes* the ec chip directly on the card.
16:50.51drmessanoOh nice
16:50.54WIMPyNext gen? But noone uses PSTN any more.
16:51.59jalewisWIMPy: unfortunately, that's not the case and won't be for some time
16:52.12igcewielingsounds more like the card is an upgrade for Digium's mfg process rather than anything which would affect end users
16:52.28WIMPyjalewis: I lost hope.
16:53.41jmetroI figured out my voicemail problem
16:53.46jmetroor rather my autodestruct problem
16:54.00jmetroapparently people are reaching voicemail and hanging up without leaving a message and every time they do that its generating a hung channel
16:54.33jalewisjmetro: on a PRI?
16:54.39jmetronegative, all sip
16:54.58GreenlightDamn those users how dare they hangup!
16:55.16jmetroyeah every time they hang up while listening to the VM message, i get autodstruct spam for 10 minutes
16:55.42drmessanoigcewieling, I don't think they're being promoted as "better", just newer hotness
16:55.48GreenlightWhich version you on ?
16:55.52jmetro11.4
16:55.58jmetrorecently upgraded to try and fix but no go
16:56.08GreenlightAs a "hack" I would try passing them via a Local channel
16:56.20GreenlightThat *may* help
16:56.30jmetroi just have it going same => n,Voicemail(${EXTEN},u)
16:56.32newtonrjalewis: no half-height bracket included btw. if you order from an e-tailer they should sell those as well. Digium can also order them. turns out we get very few requests for the half-height brackets.
16:56.40igcewielingjmetro: what do you have as the NEXT line?
16:56.44jmetroHangup
16:56.51igcewielingjmetro: you get a cookie
16:56.54igcewieling!
16:57.10igcewielingjmetro: took me over a year to get my boss to always end his extensions with a hangup.
16:57.50jmetroi end everything with a hangup
16:58.21jmetroreturns or hangups, except for my announcement macro which cant end in a hangup or it literally hangs up the call after announcing it.
16:58.26GreenlightLike if you did Dial(Local/${EXTEN}@voicemail/n)
16:58.41jmetroGreenlight: i didnt know you could dial exten@voicemail?
16:58.53GreenlightYou'll need to create the context ofc
16:58.54jmetroor is voicemail just a context that routes to voicemail ( exten _
16:58.56jmetro) *
16:58.59GreenlightExactly
16:59.00jmetroah
16:59.10GreenlightThat local chan as a "buffer" might help
16:59.12jmetrointeresting, but it seems like it would do exactly what its already doing.
16:59.39jmetroi made sure there were 0 dials in my plan until the point where an actual phone rings
16:59.39GreenlightAt a guess something is stuck waiting for an audio frame or trying to send one
16:59.46jmetroi have several drill down levels of subroutines
17:00.09jmetroso as a cleanliness procedure i only have 1 dial
17:00.21jmetroor rather 1 routine with dial
17:00.28GreenlightYup, good practice I'd say
17:00.34jalewisnewtonr: that's surprising.  Can you recommend any e-tailers who have product and are offering the lower intro price mentioned in my latest Digium spam?
17:01.54newtonrjalewis: Nope, just call up Digium sales and they'll send you to one.
17:02.17Greenlightjmetro: Or, use Playback() for the greeting and omit the "u"
17:03.44jmetroewww
17:04.03jmetrotoo complicated, my dialplan is huge and complicated enough already
17:04.21jmetroand my boss wants to expand on its complicatedness
17:04.30GreenlightI said it was a hack :) If it wasn't messy then it would be a "fix" :)
17:05.42*** join/#asterisk MLNoah (~chatzilla@noc.metalink.net)
17:06.02MLNoahis there a "recommended" way to share GROUP(var) across multiple servers working as a cluster?
17:06.42[TK]D-FenderMLNoah: not using a channel varaible at all.
17:06.53GreenlightMLNoah: Use a database
17:06.57[TK]D-Fender^
17:07.09MLNoahwhat i figured, just thought I'd ask the stupid question before I went plunging into yet another ODBC handler
17:07.23MLNoahat least hangup handlers in Ast 11 will mean I won't go completely insane trying to track it :)
17:22.55igcewielingMLNoah: doesn't help all of us who have already gone crazy from hangup handling
17:23.32MLNoahhe he.  yeah, I don't envy anyone trying to do hotdesking + queues + custom device states prior to hangup handlers
17:23.49MLNoahi tried it for about a month before 11 hit rc.  I immediately switched to 11-rc1 :P
17:24.41Kobazhangup handlers
17:24.42Kobazyay
17:24.45Kobazsomeone likes my feature
17:25.45igcewielingah, MLNoah you are already crazy
17:26.09MLNoahwell, i won't dispute the being crazy
17:27.10*** join/#asterisk picard276 (404f7f6e@gateway/web/freenode/ip.64.79.127.110)
17:27.19picard276hey guys i had a question about DTLS/SRTP encryption
17:27.32picard276i understand the basic concept in how it does a handshake
17:27.54picard276but is this after the SIP OK 200? and is there any information in the SDP to notify the endpoint that DTLS is going to be used?
17:41.08mjordanpicard276: no and yes. There's a good slide presentation of what is going on and what it happens here: http://www.ietf.org/proceedings/68/slides/rtpsec-0.pdf
17:50.00picard276so mjordan
17:50.07picard276is it just for sharing the keys
17:50.35picard276it appears that basically the SDP just has some line that defines wether it is a client or server
17:50.42picard276in the SDP you also must specif TLS/SRTP
17:51.00picard276and then the keys are exchanged in a three-way handshake over DTLS
17:51.12picard276and then the SRTP keys are exchanged and its business as usual?
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18:01.52glazis asterisk blocking 1-900 calls by default?
18:02.10MrQuistHey guys
18:02.26MrQuistWhats the extension command to make a sound loop until a group is picked up?
18:02.49MrQuiste.g. ->     same => n,Macro(setSDExtensions)
18:02.49MrQuist<PROTECTED>
18:03.20MrQuistwhile its ringing on that macro i want to play a sound
18:05.24jalewisglaz: most unlikely
18:06.59jmetroMrQuist: sounds like you want music on hold
18:07.15MrQuistwell, jmetro, kind of - we have this special number (emergency number)
18:07.19picard276? is that correct?
18:07.24MrQuistIf i call it, it rings the service desk for 10 seconds
18:07.29MrQuistafter that, it calls 5 cell phones
18:07.33igcewielingglaz: as there is no default dialplan, asterisk could not be blocking 1-900 calls be default.
18:07.43jmetroMrQuist: by music on hold i mean "A folder containing sound files that will play instead of ringing, on command"
18:07.47MrQuistbut, while its rining, its silent on the caller side. I need a default "calling beep"
18:08.03MrQuistBut i hear no ringing - thats the problem.
18:08.10jmetroadd an r
18:08.58[tpn]Cuzthe 5 cellphones aren't a ring group?
18:09.03MrQuistBut I hear no ringing - thats the problem.r
18:09.13MrQuist[tpn]Cuz, its a Macro
18:09.14MrQuistlook;
18:09.59MrQuist[tpn]Cuz, jmetro : http://pastebin.com/eStUD4yb
18:10.07jmetroMrQuist: so add music on hold with the sound file of what you want to hear. Your lowest level dials should have a Dial(exten,time,r)
18:10.10jmetrofor ringing
18:10.19jmetroor an exten,time,m(class)) for music
18:10.32jmetroi see no R's
18:10.40MrQuistthat kind of r. Aha.
18:12.27MrQuistjmetro, what do you mean an 'r' ?
18:12.37MrQuistjust a plain 'r' like this?
18:12.38MrQuistsame => n,Dial(${nooddienstExts},20,r)
18:12.48jmetroyep
18:12.52jmetrocore show application dial
18:15.25jmetrowhoat does nooddienstexts mean anyway
18:15.34glazI found a typo in my dialplan... sorry
18:17.01MrQuistNood = emergency
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18:17.03MrQuistDIenst = service
18:17.11MrQuistExts = extension
18:17.19MrQuistIts Dutch
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18:27.25MrQuistjmetro, its not working
18:27.36MrQuistif i take away the hello-world on top, it does work
18:27.59jmetrowell there you go =D
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18:28.17MrQuistbut thats not what i want, jmetro.
18:28.42jmetrowhy would you play weasels to an emergency caller?
18:28.46MrQuistWhat i want is; Dial emergencynumber "Welcome to our emergcy number. You're being put through" *beep until someone picks it up*
18:28.56jmetroahh
18:29.04MrQuistSo that people know that they are calling the emergency number
18:29.11jmetrobackground it rather than playback it
18:29.16jmetrotry that
18:29.19MrQuistIs that a simple function?
18:29.29jmetrobackground() rather than playback() yes
18:29.41*** join/#asterisk netit (~netit@66.118.84.2)
18:29.42jmetroand of course get rid of the same => n,ringing
18:29.46MrQuistexten => 6123,1,Background(prompt1&prompt2&prompt3)
18:29.50MrQuistyeah i notices
18:29.52MrQuistnoticed*
18:29.55MrQuisttook it away already.
18:30.12jmetromacro(setSDextensions) does that literally just set variables?
18:30.21netithello.. anyone willing to help debug a trixbox faxing setup?
18:30.50jmetronetit: upgrade to ast11 and ditch the gui [would be the response from most here]
18:31.16[tpn]Cuzt38modem fun whee
18:31.37netitjmetro: not really using the gui.
18:32.06jmetronetit: in that case you dont need trixbox at all, just update to ast11
18:33.01MrQuistjmetro, yes it does.
18:33.13MrQuistPerhaps its smarter to create a global one?
18:33.19jmetro=)
18:33.20MrQuistinstead of call it each time someone calls, right?
18:33.31jmetroyep
18:33.32MrQuistA collegue of mine made that part, hehehe
18:33.36MrQuistanyway
18:33.39MrQuisti tried the background thing
18:33.42jmetrosame with the noodiestexts
18:33.44MrQuiston beep / beepeter
18:33.50MrQuistnoodies hehehe
18:33.56mmlj4why can't you do exten => 911,1,Answer();   exten => 911,n,Playback(now-dialing-911);   exten => 5911,n,Dial(blah) ?
18:33.58MrQuistyeah thanks ill fix that :)
18:34.07mmlj4let Dial be your beep?
18:34.22MrQuistwell, mmlj4, Dial doesn't do anything
18:34.26MrQuistits just silent for as long as it rings
18:34.31MrQuistuntil the other side picks up
18:34.39MrQuistbackground() only plays it once
18:34.40mmlj4not on my box is isn't
18:34.47MrQuistperhaps thats Twinkle>
18:34.48MrQuist?
18:35.00mmlj4I have no idea what you're talking about
18:35.06jmetroi mean background for your hello-world
18:35.11jmetrokeep the dial (,,r)
18:35.18MrQuistTwinkle, the softphone software i'm using (im on ubuntu)
18:35.20mmlj4every time I dial on this box I get a dialing pulse, and have for years now
18:35.33netit-- Executing [913014626239@longdistance:1] Set("SIP/6291-000004bf", "CALLERID(number)=3014626291") in new stack
18:35.33netit<PROTECTED>
18:35.33netit<PROTECTED>
18:35.33netit<PROTECTED>
18:35.33netit<PROTECTED>
18:35.33netit<PROTECTED>
18:35.34netit<PROTECTED>
18:35.34netit<PROTECTED>
18:35.48MrQuistnetit, pastebin.com
18:35.54netitsorry
18:38.38jmetroMrQuist: Some softphones will supress ringing
18:38.45netithttp://pastebin.com/6jCWnZjp
18:38.45jmetroMrQuist: you could definitely look into that.
18:39.34MrQuistjmetro, my ringing sounds.. different
18:39.39MrQuistbut i can still hear it "ring"
18:39.40MrQuistalso -> http://pastebin.com/sW36c8aD
18:39.49MrQuistbackground(sound) makes me hear it once
18:40.25jmetroMrQuist: I mean get rid of the Playback(weasels) and do Background(weasels)
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18:43.05netitthis is a outbound fax call should the transfer capability be something other than SPEECH.. (3K1AUDIO) or is SPEECH correct?  http://pastebin.com/6jCWnZjp
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18:53.08zoid_[TK]D-Fender: bug reported, patch added, gotta love asterisk community
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19:01.37netitthis is a outbound fax call should the transfer capability be something other than SPEECH.. (3K1AUDIO) or is SPEECH correct?  http://pastebin.com/6jCWnZjp
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19:11.15MiccAnyone know the config file setting for a polycom to make it not accept calls from anything other than the proxy/registration server? I have a customer that says hes getting calls in the middle of the night from random servers.
19:12.46igcewielingMicc: Most settings are documented.
19:16.06jmetroor you know
19:16.12jmetrofix your server to stop sending it calls
19:20.28Miccno no, the calls aren't coming from my server.
19:23.23MiccI'm just not sure what this feature is called. Its not anonymous call rejection.
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19:28.13jmetrohow are calls not from your server even reaching it unless they are going through your server ?
19:28.40igcewielingjmetro: not everyone is behind nat
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19:38.37cb22Hi, I seem to have a problem using the jitterbuffer function in my dialplan - if I add a line "s,1,Set(JITTERBUFFER(adaptive)=200,,60);" the Asterisk console is flooded with messages saying "func_jitterbuffer.c:223 hook_event_cb: Failed to acknowledge timer in jitter buffer".
19:43.59MiccI would hope he is behind nat, but possible he is not. I see his user agent shows his external ip, so that is possible he's not nat.
19:44.24Micconly one sip device on the network. I bet his router is just letting everything to port 5060 through to his phone.
19:44.32Miccand someone is fucking with him.
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19:45.58jmetrowell all my phones have their external ip if they arent on our lan
19:46.07jmetrothey still dont accept calls from anything but their registered pbx
19:47.02Miccthis is a new polycom vvx 600
19:47.23[tpn]CuzHey we just started playing with a few of those
19:47.26[tpn]Cuznice screens.
19:49.02[tpn]Cuzthough some of the guys around here prefer the larger screen on the 1500 to the higher resolution of the 600
19:50.07MiccI find the touch is a bit laggy, but not too bad.
19:50.17MiccI like the new yealink T46 the best though.
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20:08.49gavimobilemy google voice is set to forwards to my ipcomms did, does anyone have google voice configured for the call to be answered directly from a peer without forwarding through another itsp?
20:09.14drmessanoYou need chan_motif
20:09.28drmessanoBut XMPP may not be available much longer
20:12.02gavimobiledrmessano: hrm, than I better leave it the way it is
20:12.04gavimobilethanks
20:12.33drmessanoI actually ported my numbers out..
20:13.01gavimobileon a different note, I need an addition free did. it seems like ipkall and callcentric has this problem that if I call right after another call to my did, then the call won't go trough
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20:13.54gavimobileit seems that I need to wait a while before I can be called again.. its like an intermittant answer
20:13.56drmessanoYou mean if you call, answer, and then try another inbound call?
20:14.01drmessanooh
20:14.47gavimobileI was messing with callcentric all day and just remembered I had the same issue with ipkall. ipcomms seems to be the most stable free itsp
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21:32.08[404]has anyone ever received this error message before when dealing with realtime voicemails:  WARNING[3488] res_odbc.c: Unable to retrieve database handle for table description
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22:42.45navaismoMaybe people in here want to check this https://code.google.com/p/telepresence/
22:51.14gavimobileok I pinpointed my issue. it seems that allowguest must be set to yes for my free did to work. is there a way around this, I don't want to allow guest calls
22:51.55WIMPyCreate one peer for any possible IP they send calls from.
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23:13.20picard276in the sdp
23:13.24picard276for a SDES connection
23:13.30picard276setting up the SRTP connection with SDES
23:13.44picard276in the SIP 200 OK should there be two crypto lines? (one with a / in them?)
23:15.24picard276http://pastebin.com/diV5Xsc9'
23:15.37picard276http://pastebin.com/diV5Xsc9
23:16.43picard276im getting a=sendrecv failed to parse video codes correctly (which makes me want to believe that the crytpo is not working correctly?)
23:27.49*** join/#asterisk TimeRider (~steve@timerider.plus.com)
23:29.00*** join/#asterisk pontis (~pontis@83.137.2.145)
23:32.27pontishi, is the huawei k3772 compatible with asterisk?
23:33.26WIMPyhttp://wiki.e1550.mobi/
23:34.32*** join/#asterisk vlad_starkov (~vlad_star@109.188.126.212)
23:35.09pontisthanks
23:36.31*** join/#asterisk Carlos_PHX1 (~Carlos@ip72-223-31-187.ph.ph.cox.net)
23:41.08*** join/#asterisk DEMNVT (~Adium@rmsaus7.lnk.telstra.net)
23:42.17pontishow can i see if asterisk recognizes usb huwaei dongle?
23:43.04WIMPyHave you installed chan_dongle?
23:44.09pontisyes
23:45.30pontisi ask that because the k3772 is not in the table test, i supose nobody tested!
23:48.40WIMPy'dongle show devices'
23:48.52*** join/#asterisk italorossi (~italoross@187.61.168.117)
23:49.14WIMPyYou need to configure it, off course.
23:51.52pontisit don´t appear
23:52.30WIMPyi think it should appear, even if it won;t work.
23:52.43WIMPyCheck your config.
23:54.03pontisi add the [dongle1] in dongle.conf
23:54.08pontiswith imei
23:54.33WIMPyThat's optional.
23:54.42pontisand just appear dongle 0
23:54.45WIMPyJust specify the devices.
23:55.34pontishow?
23:56.50WIMPyIf you don't know the devices, try to use minicom or so to check what is what.
23:56.52*** part/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell)
23:59.47filehehe... dongle

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