| 00:07.48 | ldiamond | why is it so hard to find a decent voip provider? :( |
| 00:11.02 | r0d3nt | its not. |
| 00:11.04 | *** join/#asterisk Fritz09 (~Adium@pop1-357.catv.wtnet.de) |
| 00:11.20 | r0d3nt | vitelity |
| 00:12.17 | ldiamond | I'm in Canada. They specify "US" only. |
| 00:12.17 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 00:13.58 | ldiamond | their rates are high for canada too |
| 00:14.10 | ldiamond | they specify "us" only for SMS (forgot to specify) |
| 00:15.35 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 00:16.42 | *** join/#asterisk Mhaddog_Mac (~anonymous@z65-50-116-231.ips.direcpath.com) |
| 00:20.42 | [TK]D-Fender | les.net |
| 00:22.37 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 00:23.05 | *** join/#asterisk ChrisInSydney (~Chris@120.155.225.52) |
| 00:24.17 | SeRi|afk | [TK]D-Fender, to see the debug of extentions underr asterisk cli I set core debug 3? |
| 00:25.12 | SeRi|afk | got it |
| 00:25.14 | SeRi|afk | :) |
| 00:48.10 | *** join/#asterisk kerx (~kerx@76-240-161-60.lightspeed.irvnca.sbcglobal.net) |
| 00:54.21 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 01:07.40 | *** part/#asterisk bminish (~bminish@pdpc/supporter/professional/bminish) |
| 01:09.43 | ldiamond | les.net doesn't have incoming sms :( |
| 01:11.20 | [TK]D-Fender | * doesn't really support SMS anyway. Go setup some other vendor's API |
| 01:12.31 | ldiamond | well, you can always have SMS to SIP Message in the SMS gateway |
| 01:12.39 | ldiamond | s/always/sometimes |
| 01:12.49 | SeRi|afk | [TK]D-Fender, can you give me a quick hand on a dial plan context that I am building for a gsm gateway I bought? |
| 01:13.42 | LiENUS | can asterisk interface with a multi function print to fax style print/fax machine to take t.38 input and fax it out over that? |
| 01:13.47 | SeRi|afk | basically I am calling in and using DISA to dial out a new number |
| 01:14.00 | SeRi|afk | but its not working |
| 01:15.17 | [TK]D-Fender | SeRi|afk: Show what you've done and what you get and we'll ssee |
| 01:15.39 | SeRi|afk | [TK]D-Fender, Thank You. I will include in the pb some notes ok. |
| 01:15.45 | SeRi|afk | Thank you thank you! |
| 01:15.49 | SeRi|afk | brb going to PB |
| 01:21.06 | SeRi|afk | [TK]D-Fender, http://pastebin.com/vZTfFF2a |
| 01:21.31 | SeRi|afk | [TK]D-Fender, here is the gsm gateway I bought |
| 01:22.09 | SeRi|afk | http://www.portech.com.tw/p3-product1_1.asp?Pid=13 |
| 01:23.36 | SeRi|afk | when I call the gateway it goes to vmail and it does not route the call to ext 1002 "DISA" |
| 01:23.51 | SeRi|afk | cell phone provider vmail |
| 01:23.57 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:25.02 | [TK]D-Fender | "what you get" = failure @ CLI |
| 01:26.13 | SeRi|afk | nothing at all. no failures because the gateway is not going to ext 1002 |
| 01:26.28 | SeRi|afk | so when I have debug on I see no activity from 1002 |
| 01:27.02 | [TK]D-Fender | SIP DEBUG? |
| 01:27.08 | SeRi|afk | nada |
| 01:27.14 | [TK]D-Fender | Doesn't help then |
| 01:27.19 | [TK]D-Fender | See what packets actually come in |
| 01:27.26 | [TK]D-Fender | Who knows if your peers are bad.... |
| 01:27.52 | SeRi|afk | well everything works fine. it just that ext. and I think I might now my own answer |
| 01:28.33 | SeRi|afk | I think the issue is the gateway |
| 01:30.12 | SeRi|afk | the gateway is not doing what is told too. |
| 01:35.54 | SeRi|afk | I think I got it. |
| 01:35.56 | SeRi|afk | Now to test |
| 01:42.16 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 01:46.43 | SeRi|afk | [TK]D-Fender, Thanks for the help. I got it. #1 The context was wrong for the gateway. Fixed, #2 The extensions where colliding in the gateway. Fixed, #3 | has been replaced by , . Fixed :) |
| 01:47.01 | SeRi|afk | all working now. |
| 01:47.04 | SeRi|afk | Thanks! |
| 01:47.06 | [TK]D-Fender | Glad to hear |
| 02:05.43 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 02:07.23 | *** join/#asterisk twitchnln (~Adium@74-60-202-243.gar.clearwire-wmx.net) |
| 02:22.07 | *** join/#asterisk x86 (x86@i.am.leet.org) |
| 02:31.14 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 02:33.26 | *** join/#asterisk twitchnln (~Adium@74-60-202-243.gar.clearwire-wmx.net) |
| 02:42.00 | *** join/#asterisk carrar (tim@osburn.com) |
| 02:42.07 | carrar | moo |
| 02:44.09 | WIMPy | C M cows? |
| 02:54.25 | carrar | Hai! |
| 02:55.42 | *** join/#asterisk jblack (~jblack@pool-71-173-1-251.sctnpa.east.verizon.net) |
| 02:56.21 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:59.02 | Kobaz | teh cows |
| 03:08.44 | *** part/#asterisk twitchnln (~Adium@74-60-202-243.gar.clearwire-wmx.net) |
| 03:11.47 | *** join/#asterisk depressed (~depressed@ca16.v6.us.gnics.net) |
| 03:19.43 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 03:26.35 | *** join/#asterisk Elijah_ (~elijah@71-209-232-44.phnx.qwest.net) |
| 03:28.00 | Elijah_ | Hi all, having a problem with getting a cisco 7940 to register... it's like asterisk doesn't reply to the REGISTER |
| 03:28.02 | Elijah_ | http://pastebin.com/p4mbY4uN |
| 03:28.26 | Elijah_ | ^^ output of the sip debug |
| 03:28.41 | Elijah_ | The latest version of asterisk 1.8 |
| 03:28.57 | Elijah_ | 1.8.7 |
| 03:30.16 | Elijah_ | Also for testing purposes... the asterisk box and the phone are on the same local network |
| 03:31.11 | [TK]D-Fender | Did you enable global SIP debug and not a specific IP? |
| 03:31.37 | Elijah_ | No, only an IP... wasn't aware there was a difference |
| 03:31.48 | [TK]D-Fender | Now try looking at the whole thing |
| 03:32.57 | Elijah_ | should I also enable the specific IP debugging, or is that a subset of 'sip set debug on' |
| 03:34.00 | Elijah_ | Hmm, still seeing the same thing though |
| 03:34.22 | [TK]D-Fender | off, then on (global) |
| 03:35.02 | Elijah_ | ok good, did that but still seeing the same thing |
| 03:38.02 | Elijah_ | http://pastebin.com/k8cws6aa |
| 03:39.01 | carrar | Were is that sip.conf file? |
| 03:40.09 | Elijah_ | http://pastebin.com/MCiZLD1V |
| 03:40.35 | carrar | try add: pedantic=no |
| 03:40.53 | carrar | in general |
| 03:41.17 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 03:42.12 | Elijah_ | Yeah, no luck |
| 03:42.18 | Elijah_ | I think it defaults to no anyways |
| 03:42.30 | carrar | you reload sip.conf? |
| 03:44.05 | Elijah_ | yeah.... lemme reboot the phone to get it to try some more |
| 03:45.03 | Elijah_ | whoa... it actually started talking back now and registered |
| 03:45.26 | Elijah_ | cool! My question is, that does default to no according to voip-info, so wonder why that changed anything.... |
| 03:50.29 | *** join/#asterisk cerberus_za (~coert@8ta-151-8-22.telkomadsl.co.za) |
| 03:54.20 | Elijah_ | Well that's great, these haven't worked since I upgraded to 1.8. Really appreciate it |
| 04:03.47 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 04:20.27 | PhoenixMage | Anyone know what the significance of the user= in a dialplan.xml is? |
| 04:21.17 | [TK]D-Fender | "dialplan.xml"? I think you took a wrong tunr .... |
| 04:21.21 | [TK]D-Fender | turn* |
| 04:22.42 | PhoenixMage | [TK]D-Fender: No I am configuring a cisco 7975G |
| 04:22.47 | PhoenixMage | for sip |
| 04:24.16 | PhoenixMage | I understand the rest of the contents of the file just not what the User means |
| 04:36.58 | Elijah_ | While I'm here, anyone have problems with the gtalk interface in the past few weeks? |
| 04:37.12 | Elijah_ | I haven't been able to make outgoing calls |
| 04:37.32 | carrar | heh |
| 04:38.26 | carrar | Hi TK! |
| 04:38.42 | Elijah_ | http://pastebin.com/M02QhMBd <--- log of the jabber debug, in case anyone's interested |
| 04:39.23 | carrar | I'm not interested |
| 04:39.54 | carrar | I just ate lunch |
| 04:40.01 | Elijah_ | lol |
| 04:42.01 | carrar | cold somen noodles, takoyaki, and some Korokke & Tempura |
| 04:42.52 | carrar | and a iced latte |
| 04:43.49 | Elijah_ | Can't say that sounds too appetizing, but so long as you like it :P |
| 04:44.00 | carrar | I liked it |
| 04:44.37 | Elijah_ | Where do you live, japan or something? |
| 04:44.54 | carrar | Seattle, just staying in Japan for a while |
| 04:45.14 | carrar | come here for a few months every year or so |
| 05:08.30 | Elijah_ | Aah, cool |
| 05:18.47 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 06:02.11 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 06:08.13 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 06:08.42 | jsjc | I wonder if anybody could show me an example of an if condition on dialplan such us if this dahdi DIAL is not succesful for anyreason then do this other DIAL. |
| 06:09.26 | jsjc | ut has been said that the onbly reason why a dahdi might fail is the not availability of a channel and it will follow on the plan but If the call gets not answered then it will follow and call again with next dial right? |
| 06:41.04 | kaldemar | put a timeout for the first dial so it moves on in the dialplan in case of no answer too. then do what you want in the next priority. |
| 06:55.27 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 06:56.08 | *** join/#asterisk cerberus_za (~coert@8ta-151-75-96.telkomadsl.co.za) |
| 07:08.29 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 07:08.46 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 07:15.44 | *** join/#asterisk donnib (~donnib@0x555281d0.adsl.cybercity.dk) |
| 07:45.56 | *** join/#asterisk jkroon (~jkroon@dsl-241-250-161.telkomadsl.co.za) |
| 07:58.02 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 08:10.35 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:24.31 | *** join/#asterisk jblack (~jblack@75-149-160-4-Washington.hfc.comcastbusiness.net) |
| 08:40.33 | *** join/#asterisk irroot (~irroot@41.51.95.167) |
| 09:03.23 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 09:08.30 | *** join/#asterisk Fritz09 (~Adium@pop1-357.catv.wtnet.de) |
| 09:08.54 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 09:31.57 | *** join/#asterisk cerberus_za (~coert@8ta-151-21-123.telkomadsl.co.za) |
| 09:37.58 | *** join/#asterisk irroot (~irroot@41.53.18.73) |
| 09:38.07 | *** join/#asterisk jblack (~jblack@pool-71-173-1-251.sctnpa.east.verizon.net) |
| 09:42.14 | *** join/#asterisk irroot (~irroot@41.52.146.19) |
| 10:03.03 | *** join/#asterisk cerberus_za (~coert@8ta-151-28-67.telkomadsl.co.za) |
| 10:14.22 | EmleyMoor | Are there any freely-available means to connect to tollfree numbers in the Netherlands or Norway now? What about other countries? (got means for France, Germany, Luxembourg, UK, US) |
| 10:15.07 | WIMPy | Get an ITSP in that country? |
| 10:16.26 | EmleyMoor | WIMPy: Well, yes, that's the other way. FWD offered Netherlands and Norway... |
| 10:18.27 | WIMPy | Sipgate might be an option. |
| 10:18.38 | dym | WIMPy is never an option |
| 10:18.43 | dym | (just saying) |
| 10:19.22 | WIMPy | might be |
| 10:19.41 | EmleyMoor | sipgate allow Germany, openly... |
| 10:20.27 | WIMPy | Well, they don't allow guests, but a free account without balance is enough. |
| 10:20.44 | WIMPy | It probably works for the UK as well. |
| 10:21.01 | EmleyMoor | WIMPy: Well, never tried it but it is on my "public" list |
| 10:21.08 | EmleyMoor | Magrathea allow UK |
| 10:23.05 | EmleyMoor | There's several for most of the countries I've got means for |
| 10:23.59 | EmleyMoor | I need to update my user guide soon... |
| 10:29.34 | dym | EmleyMoor: But Magrathea isnt free tho, is it? |
| 10:30.33 | EmleyMoor | There is a public server available at no charge. It often gets used if I dial a UK toll free - before it gets do my VoIPtalk account |
| 10:31.49 | dym | Oh, can you PM me the details to this server? |
| 10:38.58 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 10:39.30 | EmleyMoor | Discovered the loss of fwd while working on my ratefiles... looks like I haven't actually needed it anyway for a while - not had cause to call any number I needed it for |
| 10:46.41 | jsjc | I wonder if anybody could show me an example of an if condition on dialplan such us if this dahdi DIAL is not succesful for anyreason then do this other DIAL. |
| 10:46.41 | jsjc | has been said that the only reason why a dahdi might fail is the not availability of a channel and it will follow on the plan but If the call gets not answered then it will follow and call again with next dial right? |
| 10:46.58 | jsjc | No I will stay so I can give replies hehe because I was off yesterday and didnt got replies. |
| 10:49.01 | EmleyMoor | jsjc: What actually happens when a call times out over DAHDI has a lot to do with the line provider. Over here we eventually get SIT and Digital Dorothy saying "Sorry, there is no reply." - not sure the DAHDI connection ever hangs up from the far end on that - never actually tried though. |
| 10:56.45 | EmleyMoor | If you can simulate the occurrence, you can always inspect the DIALSTATUS variable... |
| 10:57.54 | jsjc | emeleymooor how could I inspect the dialstatus? I do not know how to do an if condition inspecting the dialstatus |
| 10:59.24 | EmleyMoor | Just NoOp(${DIALSTATUS}) to inspect it - provided you have verbose set to at least 3 |
| 11:00.54 | EmleyMoor | Once you know all the results in which you either do or don't want a specific action followed (you don't need to know them for both) then you can write your "if" |
| 11:00.56 | WIMPy | jsjc: What kind of dahdi interface? |
| 11:01.25 | jsjc | WIMPy: what you mean by what kind of interface? it is just a FXS/FXO card. |
| 11:01.49 | WIMPy | That was what I wanted to know. |
| 11:06.31 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 11:07.34 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 11:07.34 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 11:10.56 | jsjc | I think I am getting to something close ;) |
| 11:11.02 | jsjc | GotoIf(${DIALSTATUS}=CHANUNAVAIL?fromsip) something like this shoudl work let me test. |
| 11:12.27 | EmleyMoor | is still not clear of the precise purpose of this... |
| 11:18.28 | *** join/#asterisk timahvo1 (~rogue@41.223.56.17) |
| 11:21.04 | kaldemar | jsjc: your GotoIf is lacking $[] |
| 11:21.48 | jsjc | lacking of? I just did a test and it is working fine... |
| 11:21.57 | jsjc | I was now going to implement it on live asterisk. |
| 11:22.35 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 11:22.36 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 11:22.45 | kaldemar | it will not work properly, you'll see. |
| 11:24.53 | jsjc | so how it should be? I am not sure then where is lacking of something. |
| 11:25.46 | EmleyMoor | Have you tested both the true and false cases? It actually needs to be GotoIf($[${DIALSTATUS}=CHANUNAVAIL}?... |
| 11:25.52 | EmleyMoor | Oops |
| 11:25.55 | EmleyMoor | Have you tested both the true and false cases? It actually needs to be GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?... |
| 11:27.11 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 11:32.54 | kaldemar | even better if GotoIf($["${DIALSTATUS}" = "CHAUNAVAIL"]?fromsip) when comparing strings. |
| 11:36.39 | jsjc | ok I see. |
| 11:36.49 | jsjc | I will do it that way so I lear the right way Thanks kaldemar! |
| 11:39.29 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 11:39.29 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 11:49.23 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 12:38.44 | *** join/#asterisk niekie (quasselcor@CAcert/Assurer/niekie) |
| 12:49.09 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 12:59.40 | *** join/#asterisk irroot (~irroot@197.108.73.199) |
| 13:03.20 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 13:08.19 | *** join/#asterisk cerberus_za (~coert@41.151.106.165) |
| 13:11.10 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:14.48 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 13:57.18 | *** join/#asterisk irroot (~irroot@41.51.36.17) |
| 14:01.56 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 14:07.15 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 14:07.59 | jsjc | what this thing mean ?? heck_auth: username mismatch, have <voiprovider>, digest has <username> |
| 14:08.00 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 14:08.08 | *** join/#asterisk master_of_master (~master_of@p57B552AB.dip.t-dialin.net) |
| 14:08.26 | jsjc | handle_request_invite: Failed to authenticate device "phonenumber" |
| 14:08.39 | jsjc | I just changed voip provider and I am having a hell of a time ;) |
| 14:09.48 | *** join/#asterisk af_ (~getsmart@78.134.22.161) |
| 14:30.41 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 14:32.25 | *** join/#asterisk jblack (~jblack@pool-71-173-1-251.sctnpa.east.verizon.net) |
| 14:51.56 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 14:52.33 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 15:09.24 | *** join/#asterisk radic (~radic@dslb-094-216-242-102.pools.arcor-ip.net) |
| 15:32.58 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 15:45.26 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 16:02.51 | *** join/#asterisk Fritz09 (~Adium@pop1-357.catv.wtnet.de) |
| 16:08.48 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 16:14.04 | p3nguin | jsjc: Pastebin your peer entry for that provider. |
| 16:20.36 | *** join/#asterisk Tim_Toady (~fuzzy@77.49.252.192) |
| 16:39.52 | *** join/#asterisk luckman212 (~irc@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 16:40.11 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
| 16:43.38 | SeRi | p3nguin, you should have an ATT sim coming your way next week |
| 16:51.36 | EmleyMoor | I am using the ancient LCDial.sh script, and trying to call announce.sh from it... however, the charge announcement does not happen. Is there any way I can trace these AGI scripts as they run to see why there is no announcement? |
| 17:00.42 | *** join/#asterisk irroot (~irroot@197.107.189.18) |
| 17:05.05 | Korolev | EmleyMoor, I bet there is an easier way to trace a .sh |
| 17:05.23 | Korolev | but since I have no idea, how about just adding verbose throughout the script |
| 17:05.48 | Korolev | or echo and run it from bash |
| 17:11.17 | EmleyMoor | Korolev: It needs to be run as "normal" - running it from bash will not help. As for Verbose, I will have a look... |
| 17:19.44 | EmleyMoor | Hmmm... at a loss to see where Verbose will help... |
| 17:28.47 | EmleyMoor | I added a line to the top of the announce.sh script to indicate in the asterisk console that it had been called - but it wasn't |
| 17:32.00 | Korolev | is announce.sh called from another script? |
| 17:32.04 | Korolev | or from the dialplan? |
| 17:32.04 | *** join/#asterisk TJNII (~TJNII@207.189.199.62) |
| 17:32.56 | SeRi | p3nguin, you avail? |
| 17:32.57 | Korolev | does calling announce.sh depend on the result of other scripts called earlier? |
| 17:33.04 | EmleyMoor | From LCDial.sh - it shows being called, but it doesn't get called |
| 17:33.54 | Korolev | dumb question but just to make sure: is it executable? |
| 17:34.19 | EmleyMoor | Korolev: It has the same permissions as LCDial.sh |
| 17:35.17 | EmleyMoor | (yes, it is executable...) |
| 17:35.30 | Korolev | Im really guessing blindly, Ive never seen lcdial before, or any agi using sh |
| 17:35.35 | Korolev | but anyways |
| 17:36.13 | Korolev | so the program flow gets to the line in lcdial.sh right before calling announce.sh |
| 17:37.04 | Korolev | then calls it, announce doesnt seem to have run, and lcdial.sh continues as if it was all good? |
| 17:37.09 | *** join/#asterisk adolfomaltez (~taro@190.62.179.253) |
| 17:37.17 | EmleyMoor | Yes, that's right |
| 17:37.53 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
| 17:39.08 | Korolev | wow, thats odd, is lcdial.sh short enough to paste the code so I can read it? |
| 17:40.07 | EmleyMoor | It's 16KiB |
| 17:40.50 | Korolev | nevermind, I think I found it in voipinfo |
| 17:40.53 | Korolev | let me see |
| 17:41.29 | EmleyMoor | Mine has a minor local mod... as does announce.sh... and worked perfectly under asterisk 1.4 |
| 17:42.46 | Korolev | what are the mods? |
| 17:43.02 | p3nguin | seri: Yes. |
| 17:44.38 | EmleyMoor | Korolev: A "try next channel" under certain dialstatuses that would normally stop the script, in LCDial.sh, and one new sound file played before the number, and use of penny/pence instead of cents sound files |
| 17:44.46 | EmleyMoor | (in announce.sh) |
| 17:48.23 | SeRi | p3nguin, one sec. I want to show you something |
| 17:53.01 | SeRi | p3nguin, http://pastebin.com/xQ8D6U7L |
| 17:53.43 | SeRi | what you are looking at is a line that came back with a status of unavailable and the system was executed it the back up line |
| 17:53.48 | SeRi | my theory worked :) |
| 17:54.17 | SeRi | now I have to fine tune it to ignore busy signals and congestions |
| 17:54.34 | SeRi | all I am looking for the chan to be down for it to kick over |
| 17:54.40 | SeRi | with a backup carrier |
| 17:54.57 | SeRi | right now I am testing it with the same provider |
| 18:03.20 | EmleyMoor | Asterisk shows that announce.sh is being called, with the right price, but there is no evidence that it actually does anything |
| 18:09.12 | Korolev | doesnt complain about not finding the sound files or anything? |
| 18:10.03 | EmleyMoor | No - nothing at all |
| 18:10.52 | EmleyMoor | Just proceeds straight to Dial - which is back in LCDial.sh |
| 18:12.13 | Korolev | and sh_agi_cmd EXEC Verbose blah |
| 18:12.18 | Korolev | doesnt do anything either.. |
| 18:13.02 | EmleyMoor | I added that to announce.sh - nothing happened |
| 18:14.14 | EmleyMoor | Hmmm... got that doing something - so it is being called... |
| 18:15.48 | Korolev | did you get it to say blah? |
| 18:15.58 | EmleyMoor | Yes... |
| 18:16.11 | EmleyMoor | It looks like the cost is getting hosed! |
| 18:17.34 | EmleyMoor | Hmmm... it's not the cost that seems to be getting passed to it... as when I Verbose $cost at the beginning I get bt, not 11900 |
| 18:18.01 | Korolev | is bt one of the parameters? |
| 18:18.07 | Korolev | british telecom? |
| 18:18.31 | Korolev | you said it worked in 1.4 and breaks in 1.6 |
| 18:18.52 | Korolev | have you changed the parameter delimiters from | to , ? |
| 18:18.55 | EmleyMoor | bt is one of the providers - actually the most expensive one - it's not the one it's going to use in this case |
| 18:19.18 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-212.ks.ks.cox.net) |
| 18:19.25 | EmleyMoor | Korolev: Is that in the script? If so, no - will look |
| 18:19.33 | Korolev | in the dialplan |
| 18:19.38 | EmleyMoor | This has given me some clues :-) |
| 18:19.39 | p3nguin | seri: unknown paste ID |
| 18:20.02 | EmleyMoor | Do they need to be | or ,? |
| 18:20.14 | Korolev | agis used to be called like AGI(script|parm0,parm1|parmn) |
| 18:20.25 | EmleyMoor | I've never used | |
| 18:20.25 | Korolev | they need to be , in 1.6 |
| 18:20.30 | Korolev | ah ok |
| 18:21.08 | Korolev | maybe the order is messed up or you have a null variable in the dialplan |
| 18:21.50 | EmleyMoor | It appears to be the LCDial.sh script at fault... |
| 18:22.40 | Korolev | verbose all the variables in lcdial.sh before calling announce |
| 18:22.50 | Korolev | see what the value of each one is |
| 18:32.47 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 18:33.46 | *** join/#asterisk pyite_mac (~dschreibe@unaffiliated/pyite) |
| 18:34.20 | *** part/#asterisk pyite_mac (~dschreibe@unaffiliated/pyite) |
| 18:35.43 | EmleyMoor | Oddly, it shows the value of $cost in the console, and passes $cost in the very next line... so where it's getting BT from, I have no idea |
| 18:35.47 | EmleyMoor | bt, even |
| 18:36.58 | Korolev | try inside announce.sh |
| 18:37.15 | Korolev | verbose $0, verbose $1, verbose $2 |
| 18:37.42 | Korolev | Im not very fluent in sh, so I dont know whether the first parameter should be 0 or 1 |
| 18:38.08 | Korolev | but just to see what is being passed to announce.sh |
| 18:38.16 | p3nguin | arg0 is the actual command (in this case, the script). |
| 18:39.00 | Korolev | ok, so verbose $0 should give you announce.sh |
| 18:39.20 | EmleyMoor | Hmmm.. none correct at all! $0 = bt, $1 = rates/bt.live, $2 = 0... so where has 11900 gone |
| 18:39.24 | EmleyMoor | ? |
| 18:40.27 | Korolev | paste the line where announce.sh is called from lcdial.sh |
| 18:40.50 | EmleyMoor | Hmmm... result is varying! |
| 18:41.24 | EmleyMoor | . $ANNOUNCE "$cost" |
| 18:41.56 | Korolev | and if you put verbose $cost right before that line |
| 18:42.03 | Korolev | you get the correct value for cost? |
| 18:42.06 | SeRi | p3nguin, looks like it expired. |
| 18:42.10 | SeRi | You want to see it? |
| 18:42.18 | p3nguin | Only if you need me to see it. |
| 18:42.26 | EmleyMoor | Oh, hang on |
| 18:42.38 | EmleyMoor | I misread... |
| 18:42.40 | SeRi | I dont need you too. Just wanted you to see that my "theory" worked :) |
| 18:43.07 | EmleyMoor | $0 is /usr/share/asterisk/agi-bin/LCDial.sh (in announce.sh!) |
| 18:43.44 | EmleyMoor | In fact, the parameters appear to be those of LCDial.sh, not announce.sh |
| 18:44.09 | Korolev | and parameter $1 for lcdial is bt? |
| 18:44.27 | EmleyMoor | Er, no... actually that's even stranger... |
| 18:44.55 | EmleyMoor | There is this line above: |
| 18:45.22 | EmleyMoor | sh_agi_log "==\> calling: . $ANNOUNCE $cost" |
| 18:45.43 | EmleyMoor | That shows the correct value in the console. |
| 18:47.12 | EmleyMoor | Whatever is going on, it's addling announce.sh's parameters |
| 18:49.10 | EmleyMoor | $1 appears to be the name of the most expensive provider, $2 the name of its ratefile... |
| 18:50.28 | EmleyMoor | $3 the editing expression for the number and $4+ empty |
| 18:50.29 | Korolev | is $ANNOUNCE $cost called in only one place in lcdial? |
| 18:51.14 | EmleyMoor | Yes, just one |
| 18:53.49 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 18:56.09 | p3nguin | Is there a full list of built-in asterisk variables and the purpose of each somewhere? |
| 18:57.07 | Korolev | there is one in voipinfo, Im not sure how complete it is though |
| 18:57.41 | p3nguin | I'm looking at it, but I didn't see some variables I was looking for. Either the ones I was looking for aren't standard, or the list isn't updated. |
| 18:58.31 | Korolev | it probably isnt updated, I remember not finding some of the CHANNEL variables there, CHANNEL(recvip) I think it was |
| 18:58.42 | WIMPy | Isn't there one on the wiki? |
| 19:00.06 | *** join/#asterisk fleixius (~fleixius@unaffiliated/fleixius) |
| 19:00.23 | Korolev | EmleyMoor, again just to rule out insanity |
| 19:00.36 | Korolev | comment out the sh_agi_log line |
| 19:00.37 | fleixius | Is there a SIP Configuration which would enable peer validations? Say that a pear is still alive and 'listening' |
| 19:01.04 | Korolev | fleixius, qualify? |
| 19:01.04 | EmleyMoor | Will try that in a bit... |
| 19:01.05 | fleixius | I'm having issues where my clients are not sending SIP BYE's on force closes and Asterisk continues to transmit RTP data flooding the clients. |
| 19:01.40 | WIMPy | fleixius: Session timers and rtptimeout. |
| 19:06.04 | fleixius | WIMPy, rtptime is when the client receives no RTP or when Asterisk receives no RTP data? |
| 19:06.12 | fleixius | rtptimeout * |
| 19:06.59 | WIMPy | When Asterisk doesn't receive rtp. That's what it knows about. |
| 19:07.58 | fleixius | That's what I figured, one problem is my clients are in a listen only state where no audio is being transmitted from them. THerefore no RTP traffic would be sent. |
| 19:08.26 | WIMPy | No silence? |
| 19:08.43 | fleixius | I'm looking for a SIP ALIVE or something that would validate the clients are in fact listening. |
| 19:08.52 | WIMPy | Usually you get all sorts of issues when Asterisk doesn't receive rtp. |
| 19:09.21 | WIMPy | sessin timers, but IIRC you can't set them to less than 5 minutes. |
| 19:09.27 | fleixius | WIMPy, This happens with my MeetMe clients with the listen only flag sent. |
| 19:09.57 | WIMPy | That wouldn;t prevent the clients from sending rtp. |
| 19:10.05 | *** join/#asterisk kerx (~kerx@76-240-161-60.lightspeed.irvnca.sbcglobal.net) |
| 19:10.21 | WIMPy | And unless something has changed recently, they wouldn't receive rtp if they didn't send any. |
| 19:10.37 | irroot | WIMPy hi there having some problems with the b410 and mISDNv2 the system seems to hang on a call any ideas off top of head |
| 19:11.12 | WIMPy | irroot: No. What exactely hangs? |
| 19:11.18 | fleixius | WIMPy, on my clients end, I see all inbound RTP traffic but no outbound. |
| 19:11.27 | fleixius | Not sure if they are sending it or not, let me run tcpdump on my * box. |
| 19:12.08 | irroot | WIMPy im home so cant say for sure lost all access to the system at office dont have problem with USB any options i need to pass hfcmulti ? |
| 19:13.20 | WIMPy | irroot: I don't use any options. TE or NT? Coming or going? |
| 19:13.46 | irroot | TE originating from asterisk chan_console |
| 19:14.49 | WIMPy | irroot: Hmm. let me try. I have come across an issue, where 'channel originate' produced crashes. |
| 19:15.20 | irroot | im using the git standalone looking at diff to in kernel ATM |
| 19:16.37 | WIMPy | Interesting. |
| 19:16.49 | *** join/#asterisk Fritz09 (~Adium@pop1-357.catv.wtnet.de) |
| 19:17.44 | WIMPy | If I do 'console dial' to a local phone, I hear a reing back sample in the phone. The phones audio goes to the speakers, however. |
| 19:18.05 | WIMPy | But I have to note that I disables audio in, so that might be the cause. |
| 19:18.41 | irroot | mmm cant see anything obvious jump me what version you use ?? |
| 19:18.51 | WIMPy | And the console doesn't seem to support early media :-( |
| 19:18.53 | irroot | doubt it will be mISDNuser t |
| 19:19.30 | WIMPy | Using mISDN from 3.1.0-rc4 and mISDNuser from git. |
| 19:20.12 | irroot | ok im using git all round ill try kernel version of hfcmulti im on 3.0.6 |
| 19:20.48 | irroot | will wait till moring till i can lab it have 2 servers i can abuse properly |
| 19:21.00 | irroot | thx for the input appreciated |
| 19:23.24 | WIMPy | Just tried 'channel originate'. That seems to be ok (again?) now. |
| 19:23.42 | fleixius | WIMPy, it appears rtptimeout works for the time being, I'll need to validate my clients are effectively shutting down, using BYE's etc |
| 19:23.54 | fleixius | Thanks though, the timeout value appears to be working perfectly for the time being. |
| 19:24.18 | irroot | fleixius i patched rtptimeout recently to honour per peer settings |
| 19:24.44 | irroot | so only global might be used in sip unless you on 1.8.8.0-rc1 |
| 19:24.49 | WIMPy | It seems to have an off-by-one as well. |
| 19:25.43 | WIMPy | If you set rtptimeout to n it kills th call after n+1 seconds. At least according to the message. I haven't used a stop watch yet. |
| 19:25.55 | irroot | WIMPy ill break it till its fixed in the morning let you know .... it got to this state by fixing it till it was broken :P |
| 19:27.08 | irroot | WIMPy its possible the rtptimeout check is not 100% reliable if the channel is locked for example it will skip till next check but this should be ms not s |
| 19:27.19 | WIMPy | I probaly won't be here tomorrow, or not until late night, so feel free to PM or Mail. |
| 19:27.42 | irroot | no problem have a good day then hope it counts |
| 19:27.45 | WIMPy | It has always been one second more than configured for me. |
| 19:31.06 | irroot | i guess its a > not >= in the code .... |
| 19:31.31 | WIMPy | The usual suspects :-) |
| 19:32.09 | WIMPy | But not really important in that case. |
| 19:33.27 | irroot | yeah but then should be documented as such |
| 19:35.02 | WIMPy | Well, you could argue it's correct. n seconds of no rtp would be over at the beginning of the n+1th second. |
| 19:35.28 | WIMPy | I think it's time to count seconds. |
| 19:37.00 | *** join/#asterisk pyite_mac (~dschreibe@unaffiliated/pyite) |
| 19:37.06 | WIMPy | Nope. Seems to be one more. |
| 19:38.51 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 19:50.49 | irroot | ps 3.1-rc8 is out kernel.org is bit out of date |
| 19:51.33 | WIMPy | My version was from before they went down. |
| 19:54.30 | WIMPy | But I usually only reboot when doing hardware stuff. |
| 19:54.35 | irroot | ah it still shows think the scrape to web is broke |
| 19:54.43 | irroot | links are also broke |
| 19:54.51 | *** join/#asterisk _Raptor_ (raptorblue@andariel.informatik.uni-erlangen.de) |
| 19:55.13 | irroot | im a 3.1 virgin so pulled rc8 |
| 19:55.14 | WIMPy | git works but not all projects are back, yet. |
| 19:55.38 | irroot | yeah some are renamed -stable for one |
| 19:55.50 | WIMPy | Yes :-( |
| 19:56.11 | WIMPy | Although 2.6-stable wasn't the right name for 3.0 anyway. |
| 19:56.21 | irroot | its there but not the .y gregkh trees |
| 19:56.38 | irroot | there was a 3.0.y stable |
| 19:57.47 | WIMPy | I used linux-2.6-stable. That took me to the current 3... version. |
| 19:58.16 | WIMPy | I did so before 3 came, off course. |
| 19:58.54 | irroot | yeah but then it was going to be 2.6.40 :P |
| 19:59.23 | WIMPy | I didn't get there. |
| 20:13.08 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 20:19.40 | *** join/#asterisk evanderv (~evanderv@74.85.207.3) |
| 20:22.24 | evanderv | Hi everyone just setup my first asterisk server with Ubuntu and got my sip account registered but I see this coming up when I try calling it WARNING[19346]: chan_sip.c:13288 check_auth: username mismatch |
| 20:23.37 | p3nguin | ~pb |
| 20:23.37 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 20:23.44 | p3nguin | Pastebin your peer entry. |
| 20:23.57 | ChrisInSydney | evanderv: Check that the name of the trunk in the [trunk-name] is the same as the username=trunk-name |
| 20:24.35 | p3nguin | Actually, pastebin your entire sip.conf. |
| 20:24.58 | ChrisInSydney | fix that and reload. If that doesn't work, do what p3nguin suggests. Hell help ypu. I have to go |
| 20:29.37 | *** join/#asterisk evanderv (~evanderv@74.85.207.3) |
| 20:29.41 | evanderv | http://pastebin.com/HSDUTaJu |
| 20:29.48 | evanderv | There is my sip.conf |
| 20:30.47 | evanderv | Now I am using the asterisk-gui and I tried looking in my sip.conf and didn't see the account in there |
| 20:31.00 | evanderv | But yet I see it registered on the gui |
| 20:32.24 | p3nguin | We don't support GUIs here. |
| 20:32.56 | evanderv | Ok then let me know what to do without the GUI |
| 20:33.09 | evanderv | I can edit text |
| 20:33.36 | p3nguin | You've shown me the sample sip.conf. sip.conf is the normal place to configure your phones. |
| 20:33.48 | p3nguin | GUIs configure phones elsewhere. |
| 20:34.23 | *** join/#asterisk Fritz09 (~Adium@pop1-357.catv.wtnet.de) |
| 20:34.34 | evanderv | To my knowledge it parses the conf files in the asterisk directory |
| 20:35.27 | p3nguin | Your knowledge is slightly skewed from actuality. |
| 20:36.45 | evanderv | Well how come asterisk shows the sip account in the debug when I call the phone |
| 20:37.32 | p3nguin | (1533.47) <p3nguin> GUIs configure phones elsewhere. |
| 20:37.51 | p3nguin | Elsewhere = not sip.conf, where we normally configure phones. |
| 20:38.45 | *** join/#asterisk Ashutto (~Ashetic@2001:b00:7c00:600:6ef0:49ff:fe7d:3d35) |
| 20:38.47 | Ashutto | Hello |
| 20:39.11 | evanderv | Ok so you wont help me then |
| 20:39.31 | Ashutto | i'm behind nat, i have no control over the nat. is asterisk able to use a stun server ? |
| 20:39.33 | evanderv | GUI asside what do I need to look at on my sip.conf |
| 20:39.50 | p3nguin | Configure a peer definition for your phone. |
| 20:42.01 | p3nguin | The problem is that you can't simply say "GUI aside" and make it go away. The GUI has still already configured things in at least one file which is not sip.conf. |
| 20:43.27 | evanderv | Ok at least tell me what that error means and what I should be looking for |
| 20:44.16 | *** join/#asterisk scubes13 (~scubes13@cpe-024-168-196-000.sc.res.rr.com) |
| 20:44.21 | p3nguin | It tells me there is a misconfiguration. But since I don't have your configuration in a supported format, I'm not sure if I could tell you where you've messed up. |
| 20:45.34 | evanderv | Ok thanks |
| 21:04.52 | [TK]D-Fender | evanderv: users.conf |
| 21:07.27 | EmleyMoor | I tried commenting out the line that logged the call of announce.sh - no difference |
| 21:13.18 | EmleyMoor | I'll just summarise my latest trouble... I'm using the LCDial.sh script, with its associated announce.sh. Despite the call of announce.sh being correct, the parameter (the call cost) is not available in announce.sh, with the fields from a line of the LCDial.sh.conf being present instead. This strikes me as bizarre. The end result: no call cost announcement. Why is this happening? |
| 21:16.26 | ChannelZ | that has more to do with whatever scripts you're using than Asterisk |
| 21:17.38 | EmleyMoor | ChannelZ: They worked fine on my old setup, 32-bit Debian Lenny, Asterisk 1.4. They have never worked properly on the new one, 64-bit Debian squeeze, Asterisk 1.6 |
| 21:17.57 | ChannelZ | Like I said |
| 21:18.16 | ChannelZ | You're talking about 3rd party scripts |
| 21:26.06 | *** join/#asterisk cerberus_za (~coert@8ta-151-80-128.telkomadsl.co.za) |
| 21:35.15 | *** join/#asterisk OneNarrowWay (~OneNarrow@ip4da1344b.direct-adsl.nl) |
| 21:40.34 | *** join/#asterisk PickledParrot (~PickledPa@host86-150-3-99.range86-150.btcentralplus.com) |
| 21:43.12 | *** join/#asterisk devil_evoxxx (~d3v1l@host79-37-dynamic.4-87-r.retail.telecomitalia.it) |
| 21:44.16 | CrossWired | at what point is a call handed to agi? After the connection is established right? |
| 21:44.40 | [TK]D-Fender | CroWhen you call AGI |
| 21:44.42 | *** part/#asterisk CrossWired (~chatzilla@65.210.186.34) |
| 21:44.46 | *** join/#asterisk CrossWired (~chatzilla@65.210.186.34) |
| 21:44.50 | CrossWired | at what point is a call handed to agi? After the connection is established right? |
| 21:45.00 | [TK]D-Fender | CrossWired: When you call AGI |
| 21:45.21 | devil_evoxxx | hi all |
| 21:45.49 | CrossWired | [TK]D-Fender: i'm using originate, and pointing it to a context, that context has one entry that points to my agi, is that a correct setup? |
| 21:46.02 | [TK]D-Fender | CrossWired: If that's what you want to do |
| 21:46.38 | CrossWired | [TK]D-Fender: for a busy line, how do i capture that status as that doesn't seem to ever hit my agi |
| 21:46.52 | CrossWired | [TK]D-Fender: same for disconnected numbers |
| 21:47.11 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 21:47.58 | [TK]D-Fender | CrossWired: Go look at the channel you are originating. |
| 21:48.12 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 21:48.48 | [TK]D-Fender | CrossWired: Because where you are dumping it only matters when the call is answered |
| 21:49.17 | CrossWired | [TK]D-Fender: ok, so AGI is only for answered calls, i need to monitor the created channel for the non answered status, got it |
| 21:49.30 | devil_evoxxx | if i use cdr_radius module, the cdr is sent to radius when call hangup, or is sent when call start with interim-update when call going on? |
| 21:50.39 | CrossWired | [TK]D-Fender: is there a secret for getting different statuses? i set up a timer to check my channel status and it seems I only get Down or Ringing, never a busy |
| 21:51.54 | *** join/#asterisk Greenlight (~Wullie@cpc2-dund11-2-0-cust994.sgyl.cable.virginmedia.com) |
| 21:52.08 | [TK]D-Fender | CrossWired: Because when it dies with busy that's it. Game over. It doesn't stick around. |
| 21:52.35 | [TK]D-Fender | CrossWired: You should be thinking about what type of channel you should be creating that would let you use your own logic around the attempt(s) |
| 21:52.48 | CrossWired | [TK]D-Fender: but that should show as the hangup cause right? My hangup cause comes back as normal clearning |
| 21:53.34 | CrossWired | [TK]D-Fender: what do you mean what type of channel? are there different types of channels for the AMI originate? |
| 21:54.23 | Greenlight | You could use a local channel and then use whatever logic you needed? |
| 21:54.59 | CrossWired | use a local channel, have it run a dial, and grab the status off of that? |
| 21:55.14 | [TK]D-Fender | Correct |
| 21:55.16 | Greenlight | Indeed |
| 21:55.28 | [TK]D-Fender | Greenlight: Way to just HAND it to him :p |
| 21:55.40 | Greenlight | :P |
| 21:56.03 | Greenlight | I'll be sure to keep him guessing some more in future :) |
| 21:56.19 | CrossWired | I thought I was doing fairly well not coming in pestering you guys with retarded questions today, but when I realized i had been here 7 hours, i knew i needed a bit of help |
| 21:56.31 | CrossWired | and i appreciate it |
| 21:57.12 | Greenlight | :P |
| 21:58.03 | devil_evoxxx | guys there is some way to "rate" the cdr during a call? i've to make a prepayed-system and integrate it wity our crm.. |
| 21:58.24 | Greenlight | Rate - like an account code? |
| 21:58.27 | devil_evoxxx | but billing multiple number associated with single account |
| 21:58.40 | devil_evoxxx | is not easy |
| 21:59.00 | devil_evoxxx | rate..something like " decrasing $ during call" |
| 21:59.54 | Greenlight | I've not done anything specifically like that before but think you might need to go down the AGI route |
| 22:00.05 | *** part/#asterisk pyite_mac (~dschreibe@unaffiliated/pyite) |
| 22:00.43 | devil_evoxxx | yes, but agi is launched when i call it in dialplan |
| 22:01.21 | Greenlight | I seem to recall a method of launching it in the background, I may be wrong tho |
| 22:01.26 | [TK]D-Fender | AMI monitoring process |
| 22:01.33 | Greenlight | Yea - or that |
| 22:01.53 | devil_evoxxx | with ami can i read the billsec for every channel? |
| 22:02.16 | Greenlight | You can do anything pretty much with AMI |
| 22:02.20 | devil_evoxxx | because i'm thinking to make a dialplan for prepayed account and use "L" options in dial application |
| 22:03.10 | Greenlight | Would that not just end the call after x secs tho? |
| 22:03.28 | devil_evoxxx | yes, but in case of two cuncurrent call |
| 22:03.32 | devil_evoxxx | associated with same account |
| 22:03.41 | devil_evoxxx | here is the problem |
| 22:04.20 | devil_evoxxx | call 1 start, and i set L limit calculated on destination number... |
| 22:04.36 | devil_evoxxx | call 2 start, and now how can i calulcate the right limit? |
| 22:04.38 | Greenlight | With concurrent calls you won't know what to set L to tho surely? |
| 22:04.41 | Greenlight | Yea |
| 22:04.56 | Greenlight | You need a method of tracking and controlling it in realtime |
| 22:05.06 | devil_evoxxx | and so, i need something like a periodical update of channel status |
| 22:05.18 | devil_evoxxx | yea, i need to control it in realtime |
| 22:05.44 | Greenlight | AMI would give you the best result - eg you could orignate a chanspy whisper to play a message to the caller "You have X seconds remaining" so the callee doesn't hear it etc |
| 22:06.44 | Greenlight | And then of course cut of calls once prepaid credit is used |
| 22:07.49 | devil_evoxxx | ok..but i'm little bit confused how to change the L limit when a call is already started. Or i obmit the L parameter and cut the call via ami..or.. |
| 22:07.53 | devil_evoxxx | or..i don't know |
| 22:08.12 | Greenlight | If you went down the AMI route, you'd have to control it all yourself from there |
| 22:09.41 | devil_evoxxx | i think that if i check the billsec status with amy every 30 second |
| 22:10.17 | devil_evoxxx | when i recalculate the credit amount, if the call end after a ami-script udpate |
| 22:10.21 | devil_evoxxx | i loose that billsec |
| 22:11.01 | Greenlight | Billsec is saved in the cdr tho isn't it |
| 22:12.24 | devil_evoxxx | yes, so you say that i've to use ami only to check channel status / billsec for calculating the effective max duration of call |
| 22:13.03 | Greenlight | And you'd need to use it to hangup channels that are over the limit |
| 22:13.35 | Greenlight | As I said before AGI is perhaps also an option, but I've not a lot of experiance with it - I've almost exclusively used AMI in the past |
| 22:15.01 | devil_evoxxx | i've used ami for something |
| 22:15.14 | devil_evoxxx | so..i'm searching for a good solutions |
| 22:15.23 | Greenlight | Just did a quick google - http://www.dynx.net/ASTERISK/AGI/ccard/agi-ccard.agi is that the sort of thing you're meaning? |
| 22:16.02 | Greenlight | Actually reading though it i'm not sure it deals with hanging up and checking ongoing credit |
| 22:16.22 | Greenlight | But you get the idea |
| 22:16.49 | devil_evoxxx | yes, something like that |
| 22:16.58 | devil_evoxxx | there is the concept |
| 22:19.45 | Greenlight | Ultimately AMI will give you more control over things, and if you've used it before then i'd go with that |
| 22:20.45 | devil_evoxxx | Greenlight: thankyou :) |
| 22:20.57 | Greenlight | np |
| 22:20.58 | devil_evoxxx | now i've got some idea.. |
| 22:21.04 | devil_evoxxx | tonight i sleep on this idea |
| 22:21.36 | devil_evoxxx | i'm thinking in a entry on a mysql table |
| 22:21.40 | devil_evoxxx | and chec periodically |
| 22:21.52 | devil_evoxxx | ..mm |
| 22:22.00 | Greenlight | mmhmm more control again |
| 22:22.06 | Greenlight | Could tie in your payment system too |
| 22:22.22 | Greenlight | Even have a web portal to check balances etc |
| 22:22.39 | devil_evoxxx | yea, and whe a secon call has been generated i can recalculate max amount of "second per account" |
| 22:22.46 | devil_evoxxx | in realtime and after that |
| 22:22.50 | devil_evoxxx | close the call |
| 22:23.01 | Greenlight | Yea - both calls |
| 22:23.06 | devil_evoxxx | fine :) |
| 22:23.14 | Greenlight | Good luck! |
| 22:23.22 | devil_evoxxx | thankyou |
| 22:23.30 | devil_evoxxx | now i try to start my tv with MythBuntu :P |
| 22:23.39 | Greenlight | ;P |
| 22:23.47 | devil_evoxxx | Good night and thankyou again |
| 22:23.53 | Greenlight | np nn |
| 22:23.59 | devil_evoxxx | bye |
| 22:32.37 | Greenlight | hmm - I'm still getting issues with jitterbuffering between two boxes with an IAX trunk. Anyone know if there could have been some changes between 1.8 and 10 which have introduced incompatibilities with trunking and jitterbuffering ? |
| 22:33.21 | Greenlight | eg one box is 1.8 and it's trunked to a box running 10 |
| 22:35.28 | p3nguin | If you disable trunking, do you still have a problem with the jitter buffer? |
| 22:36.05 | Greenlight | Yea, come to think of it I do |
| 22:36.18 | p3nguin | So then it's probably not related to the trunking. |
| 22:36.37 | Greenlight | Yup good point |
| 22:36.42 | p3nguin | Sounds more like plain jitterbuffer problems in IAX2. |
| 22:36.58 | Greenlight | That still a common thing these days? |
| 22:37.16 | p3nguin | I couldn't say. I don't have my jitterbuffer enabled. |
| 22:39.19 | Greenlight | From my tests it seems that I need to have jb enabled on the channel in order to use say the dialplan JITTERBUFFER function |
| 22:39.28 | Greenlight | But it's hard to tell |
| 22:40.24 | Greenlight | Using a Local channel with "/nj" did seem to help at first but thing I might have just got a couple of lucky test calls as it seemed to get jittery again after |
| 22:42.23 | Greenlight | I'd hoped it was just something silly I'd not configured correctly, but think I've tried just about every possible combination of config options to no avail. All that's left is to try using a SIP trunk between the boxes which I'd been loathed to try so far |
| 22:46.31 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 22:54.13 | *** join/#asterisk DanFromUK (~DanFromUK@2.27.26.114) |
| 22:55.20 | DanFromUK | i just noticed that asterisk hadnt closed some old channels, and although the sip trunk provider ended the call a few days ago, asterisk still thought it was running. just rebooted the server and the cdr was updated with billsec=16004412. |
| 22:55.51 | DanFromUK | is there any way to detect this kind of thing and "restart when convenient" automatically? |
| 22:56.09 | DanFromUK | actually r.w.c wouldnt work if asterisk thinks a call is still active |
| 23:00.03 | [TK]D-Fender | tell * to end the channel |
| 23:01.02 | DanFromUK | how can you get the call length on the cli? |
| 23:01.39 | [TK]D-Fender | "core show channel [thechannel}" |
| 23:02.13 | DanFromUK | i tried that but it didnt show call length |
| 23:02.49 | [TK]D-Fender | PB it |
| 23:08.30 | DanFromUK | http://pastebin.com/UEh0jmQP |
| 23:08.52 | DanFromUK | actually, thats from sip show channel {ddddd} |
| 23:08.59 | DanFromUK | let me try core..... |
| 23:10.22 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 23:12.22 | DanFromUK | strange. even during an active call billsec=0 |
| 23:12.51 | DanFromUK | ah, i see it. elapsed time. |
| 23:13.12 | DanFromUK | ok, thanks for that. i'll create something that will monitor for large elapsed times. |
| 23:13.17 | DanFromUK | thanks for your help. |
| 23:15.50 | *** join/#asterisk tzanger (tzanger@174.142.65.89) |
| 23:23.45 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 23:28.34 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |