| 00:00.11 | Dovid[Laptop] | mmartinn: i got bumped. i am back. did u see my last ? |
| 00:00.25 | mmartinn | Dovid[Laptop]: (repeating what I said before) It appears so. If you're running this on the same machine, that second page shows you how to do it without that class. The LJ page is barely a class at all. |
| 00:01.17 | Dovid[Laptop] | thnaks |
| 00:01.27 | Dovid[Laptop] | didnt get it the first time caus emy internet crapped out |
| 00:01.31 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:02.25 | *** join/#asterisk outlier (n=tyoung@70.141.147.180) |
| 00:02.32 | *** join/#asterisk Soulbane (n=Sunforge@202.3.110.33) |
| 00:02.43 | JT | anyone here from sydney? |
| 00:04.55 | *** join/#asterisk qdk (n=qdk@80.243.125.204) |
| 00:06.31 | Dr-Linux|home | i need an idea, is it possible, if a new voicemail come a mailbox, i dial a pstn number using .call file and deliver the message? |
| 00:06.40 | outlier | Sorry for the stupid question, but can you connect to a POTS line in the states with an ATA, or do I need one of the PC cards to do it? |
| 00:06.54 | JerJer | outlier: you can do it with like a SPA-300X |
| 00:07.11 | outlier | Thanks, JerJer |
| 00:07.52 | Dr-Linux|home | does my quesiton make sense? :S |
| 00:08.35 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 00:08.50 | JerJer | Dr-Linux|home: its possible |
| 00:09.06 | JerJer | not very friendly to pull off, but certianly possible |
| 00:09.22 | deeperror | why not just route the call to the other number instead of vm? |
| 00:10.37 | Dr-Linux|home | deeperror: imagine you are my customer and i call your cell phone and tell you, "you have a new message" |
| 00:11.05 | deeperror | ahh |
| 00:11.27 | Dr-Linux|home | JerJer: then looks difficult for a guy like me :) |
| 00:12.27 | Dr-Linux|home | but there is an other thing as well, i want if my customer calls my IVR, he should have get auto prompt, you have 2 new messages |
| 00:13.26 | *** join/#asterisk Juggie (i=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 00:16.04 | outlier | Could someone recommend a good reference for someone who needs to do a moderately complicated looking project but has never touched Asterisk before? |
| 00:16.20 | JT | ~thebook |
| 00:16.32 | jbot | thebook is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 00:16.32 | outlier | ...? |
| 00:16.42 | JT | patience, jbot is lagged |
| 00:16.43 | sbingner | Dr-Linux, that should be easy assuming they log in somehow |
| 00:17.52 | Dr-Linux|home | sbingner: yes, but thinking about a good idea to do that |
| 00:18.23 | outlier | Thanks all. |
| 00:18.26 | sbingner | Dr-Linux, there are apps or functions that tell if there is new vm for a box |
| 00:19.41 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 00:19.46 | Dr-Linux|home | sbingner: like? |
| 00:19.59 | *** join/#asterisk designdream (n=felipe@rrcs-71-40-49-30.sw.biz.rr.com) |
| 00:28.43 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 00:30.46 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 00:35.19 | *** join/#asterisk monstertruck (n=monstert@74.167.124.204) |
| 00:36.38 | `p4r14h | ok, i have a x100p on IRQ 5 all by itself, if i add another x100p should i try to get it all on its own IRQ or would sharing 5 between the two work fine, I know these cards cause alot of interupts thats why I am asking. |
| 00:37.13 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 00:38.31 | *** join/#asterisk Mercestes (n=Merceste@cpe-24-175-82-3.houston.res.rr.com) |
| 00:41.17 | elriah | `p4r14h: If your BIOS is assigning them both 5, it should work fine. |
| 00:42.33 | monstertruck | haha, aydiosmio |
| 00:44.28 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 00:45.27 | DrukenLPY | son of a bitch..... |
| 00:45.29 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 00:45.43 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 00:47.09 | DrukenLPY | moving in two weeks, gonna be double digits for this week and next, then on the saturday (the day i'm moving) it's going to plunge like 20 fucken degrees into the negative |
| 00:47.23 | deeperror | im' moving too |
| 00:47.25 | deeperror | to detroit mi |
| 00:47.26 | hphinc | I don't know how to authenticate PASSWORD to 64.61.93.90 mean? |
| 00:47.28 | deeperror | hahaha |
| 00:47.36 | DrukenLPY | from ? |
| 00:47.38 | hphinc | what does I don't know how to authenticate PASSWORD to 64.61.93.90 mean? |
| 00:47.48 | deeperror | columbus |
| 00:47.59 | DrukenLPY | ohio |
| 00:48.01 | DrukenLPY | ? |
| 00:48.06 | deeperror | yea |
| 00:48.17 | DrukenLPY | so not a huge weather change |
| 00:48.25 | deeperror | its different |
| 00:48.30 | *** join/#asterisk kiwoneka (n=kiwi@KTNRON06-1168103823.sdsl.bell.ca) |
| 00:48.32 | deeperror | mi is iffy at best |
| 00:48.39 | deeperror | the weather changes every day |
| 00:48.42 | deeperror | cols is more steady |
| 00:48.44 | kiwoneka | hello to all |
| 00:48.45 | DrukenLPY | hahahaha |
| 00:48.58 | DrukenLPY | try crossing the border and driving a while :) |
| 00:49.03 | DrukenLPY | that's where i am |
| 00:49.06 | deeperror | ca |
| 00:49.14 | deeperror | yea its right there on the other side of the water |
| 00:49.22 | kiwoneka | today i would like to get hinting working with my polycom 601s |
| 00:49.24 | hphinc | Does anyone know what " I don't know how to authenticate PASSWORD to 64.61.93.90" means? |
| 00:49.28 | kiwoneka | i will need some hel |
| 00:49.33 | kiwoneka | help |
| 00:49.56 | hphinc | kiwoneka: Try this: http://www.voip-info.org/wiki/view/Asterisk+presence |
| 00:50.24 | kiwoneka | let me get started |
| 00:50.26 | kiwoneka | thanks |
| 00:50.28 | DrukenLPY | oh the wiki is back up is it? |
| 00:50.33 | hphinc | yep |
| 00:50.37 | hphinc | came up a couple of days ago. |
| 00:50.39 | Qwell | DrukenLPY: for like a week now |
| 00:51.04 | DrukenLPY | Qwell: well, i do work ya know... i know it was questionable a while ago... but it's a fact... :) |
| 00:53.18 | Fieldy | any suggestions on DID providers? just need two, one in the philippines, the other in the usa |
| 00:53.22 | hphinc | Trying to do an IAX authorization to place a call is giving me a warning: I don't know how to authenticate (USERNAME) to 64.61.93.90 |
| 00:53.33 | hphinc | anyone have any ideas? |
| 00:53.50 | hphinc | I have auth=md5.... |
| 00:54.18 | hphinc | Fieldy: If you find some good ones, I'd like to know about them. |
| 00:54.54 | Fieldy | :P |
| 00:55.34 | kiwoneka | hphinc, i have already done that |
| 00:55.47 | kiwoneka | do i need to setup buddies? |
| 00:56.48 | *** join/#asterisk chrisknight (n=explodin@cpe-71-79-81-174.columbus.res.rr.com) |
| 00:57.29 | ezway` | i a mdisapointed of asterlink ;( |
| 00:58.38 | *** join/#asterisk Shaun2222 (n=shaun@ip68-4-212-221.oc.oc.cox.net) |
| 00:58.46 | Shaun2222 | whats a good sip softphone? |
| 00:58.56 | Qwell | hmm, speaking of softphone |
| 01:00.42 | ezway` | x lite ? well ... |
| 01:00.55 | Shaun2222 | for windows.. |
| 01:01.03 | ezway` | x lite ? well ... |
| 01:01.28 | Shaun2222 | there was one i tryed way back that was nice... |
| 01:01.34 | Shaun2222 | cant remember the name though. |
| 01:01.45 | Shaun2222 | all gui'd up... |
| 01:01.48 | JT | did it cost money? |
| 01:01.52 | Shaun2222 | no dont think so. |
| 01:02.05 | JT | xlite, ekiga, idefisk |
| 01:02.11 | Shaun2222 | idefisk |
| 01:02.14 | Shaun2222 | that was it i think |
| 01:02.20 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.bp-1031.rubicom.hu) |
| 01:02.22 | JT | idefisk does iax2 |
| 01:02.31 | Shaun2222 | hmm... |
| 01:03.31 | chrisknight | I've got a problem... maybe someone can tell me what Im missing. Ive got an asterisk pbx & 2 phones. Ext 200 & ext. 110. 1 phone (ext 200) and the asterisk box are at lan 1... a vpn tunnel links the 2nd network back here. From the far end, he can call me, and he can dial 9 to get an outside line & it works, but I cant call him. It says that hes on the hone when he is really not. does that make sense? |
| 01:03.48 | Shaun2222 | does iax2 have the same issues with nat that sip does? |
| 01:03.56 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 01:03.58 | JT | not really |
| 01:04.01 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 01:04.04 | JT | it punches nat easier |
| 01:04.16 | JT | it is less supported however |
| 01:04.29 | JT | and doesn't seem to have much pbx functionality |
| 01:04.43 | *** join/#asterisk mog (n=mog@71.207.200.130) |
| 01:04.43 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:04.48 | Shaun2222 | i just need somthing that will work through a laptop for now... |
| 01:05.01 | Shaun2222 | probably only use it for a day or two.. |
| 01:05.07 | chrisknight | x lite works for me... |
| 01:05.19 | JT | well if you have iax2 setup on your server, it's as good as anything else for that |
| 01:06.09 | Shaun2222 | i'm using iax2 right now to connect to the voip provider.. |
| 01:07.02 | Shaun2222 | x lite does video, thats kinda cool, i havnt gotten into that yet. |
| 01:07.39 | Shaun2222 | not that i realy want too... guess i could start the first 900 video phone sex service ;) |
| 01:08.27 | JT | that'd be expensive to run |
| 01:08.40 | JT | you'd need empolyees with a hot face AND voice |
| 01:08.47 | Shaun2222 | lol |
| 01:08.50 | Shaun2222 | true. |
| 01:09.46 | Shaun2222 | could just keep it focused on there chest or.... |
| 01:09.59 | JT | those sms services, often have guys working for them |
| 01:10.01 | Shaun2222 | haha, ok, i think i'm going to install this xlite and see if it works well. |
| 01:10.08 | *** join/#asterisk jjshoe (n=jjshoe@adsl-75-14-241-209.dsl.irvnca.sbcglobal.net) |
| 01:10.09 | Shaun2222 | JT: ya i dont doubt it |
| 01:10.20 | jjshoe | what's the syntax for specifying multi variables with set in a call file? |
| 01:10.24 | Shaun2222 | those places must make bank... people are idiots. |
| 01:10.38 | jjshoe | set a=b b=c c=d ? |
| 01:10.50 | flenders | is there a way to store voicemail passwords on a database? |
| 01:11.06 | Shaun2222 | flenders: probably realtime+voicemail |
| 01:11.27 | Shaun2222 | not sure though, i just know when i played with realtime everything was in a db.. |
| 01:12.11 | flenders | Shaun2222: I'll have a look into that, thanks |
| 01:12.11 | Shaun2222 | get ready to have phun! :) |
| 01:12.36 | *** part/#asterisk mog (n=mog@71.207.200.130) |
| 01:13.03 | *** part/#asterisk deeperror (n=deeperro@adsl-69-209-151-167.dsl.sfldmi.ameritech.net) |
| 01:14.27 | *** join/#asterisk nog (n=evan@c-69-180-239-206.hsd1.tn.comcast.net) |
| 01:14.41 | *** part/#asterisk nog (n=evan@c-69-180-239-206.hsd1.tn.comcast.net) |
| 01:15.03 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 01:15.03 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:15.36 | *** join/#asterisk nog (n=evan@c-69-180-239-206.hsd1.tn.comcast.net) |
| 01:15.37 | nog | hello |
| 01:15.47 | Qwell | erm |
| 01:17.07 | nog | i am having troubles getting asterisk to work right. so far google has failed to help me with my problem.. |
| 01:19.11 | hphinc | me too, nog, but what's wrong, maybe I can help... |
| 01:20.07 | nog | i have 2 softphones and one budgetone sip phone hooked up here at my house, i added the appropriate configs. They seem to work, as i can connect to voicemail and other such things... |
| 01:20.13 | Fieldy | anyone using nuphone for a DID in the philippines? tried querying their page but their rate calculator neither says the rate or if it's supported |
| 01:20.25 | nog | the problem is that when i try to call another, it never rings and goes straight to voicemail |
| 01:20.37 | hphinc | nog, pastebin your extensions.conf |
| 01:20.58 | nog | is pastebin running ok now? ;) |
| 01:21.07 | Qwell | is it ever? |
| 01:21.24 | Fieldy | rafb.net/paste is a lot more reliable |
| 01:21.41 | Qwell | uuoc.com |
| 01:21.47 | Fieldy | you can also install nopaste and pipe output into it and it gives you a url |
| 01:21.55 | Qwell | uuoc has one of those too :D |
| 01:22.15 | Qwell | uuocpipe...very fitting name |
| 01:22.36 | Qwell | Don't tell Idle` I said that...he'll get a big head about it ;) |
| 01:22.39 | Fieldy | any suggestions for a DID provider? looking for two, one in the Philippines, one in the USA |
| 01:23.01 | Fieldy | pointers of where else i could ask are welcome too, google searches aren't really returning useful results |
| 01:24.24 | *** join/#asterisk LakeSolon (n=blake@64-83-205-22.dhcp.stcd.mn.charter.com) |
| 01:25.34 | MACscr | has anyone tried the voiceone gui with asterisk? |
| 01:25.58 | JT | err what's wrong with pastebin.ca? |
| 01:26.21 | Fieldy | these days it's suffering from the same issues as pastebin.com: overuse |
| 01:27.02 | JT | it seems fine to me |
| 01:27.17 | JT | it also is done a lot better than most other paste sites |
| 01:27.37 | JT | pastebin.com... i have no idea how you can stuff something so simple up so much |
| 01:27.43 | MACscr | i use phpfi.com for simple stuff |
| 01:29.01 | JT | phpfi is not as functional as pastebin.ca |
| 01:29.37 | MACscr | correct, hence simple stuff |
| 01:31.16 | JT | why is why it's easier to use pastebin.ca every day |
| 01:31.35 | nog | ok.... |
| 01:31.36 | nog | http://rafb.net/p/lp9iaY69.html |
| 01:31.41 | MACscr | right, but if pastebin.ca is having issues, then you need something else, duh |
| 01:31.47 | MACscr | your such a troll, sheesh |
| 01:32.10 | JT | i'm not a troll |
| 01:32.15 | JT | you're just illogical |
| 01:32.16 | JT | that is all |
| 01:33.04 | Fieldy | cough |
| 01:33.59 | nog | i should also preface it all by saying the configs were generated using freepbx |
| 01:34.01 | JT | Fieldy: feeling unwell? |
| 01:34.07 | JT | nog: :( |
| 01:35.33 | nog | freepbx is usefull for an environment where the person adding extensions and such is not extremely familiar with asterisk configs... |
| 01:36.02 | JT | that is not an optimal environment |
| 01:36.18 | JT | an interface for users to change names on extensions, etc, is ok |
| 01:36.28 | JT | but they should not be configuring a pbx from scratch |
| 01:36.37 | flenders | freepbx is very messy, isn't it? |
| 01:36.52 | JT | have you seen the diaplans from it? |
| 01:36.56 | JT | they're pretty awful |
| 01:37.08 | JT | 40+ lines to dial a simple phone number |
| 01:37.13 | flenders | I could setup asterisk here, with most of the features I wanted, and my extensions.conf file is A LOT better and easier to read than that |
| 01:37.16 | nog | of course its not an optimal environment. optimal would mean everyone knows all about everything they use... |
| 01:37.21 | nog | you will never see that :P |
| 01:37.33 | flenders | nog: have a quick read at the book |
| 01:37.39 | JT | nog: nono, as in the person who created the pbx knew what they were doing |
| 01:37.42 | flenders | nog: it's a lot easier than you might think |
| 01:37.47 | JT | the users need not know how to setup asterisk |
| 01:37.54 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:38.17 | MACscr | if a book is required for basic setup and functionality of a system, then it was poorly made |
| 01:38.18 | nog | so, you are going to burden the person who setup the pbx's time to make a simple change to it? |
| 01:38.32 | JT | MACscr: what a crock of crap |
| 01:38.54 | flenders | wow, can't believe someone would say that |
| 01:38.55 | JT | MACscr: well designed systems are not made by idiots |
| 01:39.07 | JT | what would the point of an IT department be then |
| 01:39.11 | MACscr | i said basic functionality, i didnt say extended |
| 01:39.24 | flenders | macTijn: have you read it? |
| 01:39.31 | flenders | MACscr: have you read it? |
| 01:39.39 | JT | even if you want a basic propretary pbx, someone who costs money comes in and installs it |
| 01:39.53 | MACscr | no, why would i? i have setup an asterisk system before without it too |
| 01:39.54 | JT | that's the parallel |
| 01:40.41 | flenders | thing is, you don't need to read the whole thing, but all the questions a beginner might have ARE there |
| 01:40.41 | MACscr | JT: when i did that for a living, i didnt read a single book about it |
| 01:40.52 | JT | nog: depends if "simple" extends to reconfiguring the extensions or just a very simple change |
| 01:40.53 | r0d3nt | aydiosmio: hi |
| 01:41.13 | JT | MACscr: you don't HAVE to read the book, it simply has the potential to save you a bit of time |
| 01:42.06 | flenders | JT: I thought you HAD to read it... damn it! what I waste of time! |
| 01:42.10 | JT | :P |
| 01:42.11 | MACscr | to me, a good installer uses a wizard for simple setup, then of course something else for advanced setup |
| 01:42.11 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-171-251.ny325.east.verizon.net) |
| 01:42.12 | flenders | :o) |
| 01:42.20 | flenders | WIZARD???? |
| 01:42.24 | JT | no, that's simply a windows mentality |
| 01:42.35 | JT | wizards are notorious for screwing things up |
| 01:42.51 | nog | haha... (apart from the actual conversation) so are typos |
| 01:43.20 | MACscr | JT: all you like to do is argue, i give up |
| 01:43.33 | flenders | actually, are there any wizards for asterisk? |
| 01:43.34 | JT | MACscr: you like giving up, seems like a common theme |
| 01:44.01 | MACscr | JT: im only giving up on talking to you |
| 01:44.02 | JT | MACscr: i like to discuss things based on facts and actual merits, not rhetoric like "a good installer..." |
| 01:44.18 | JT | MACscr: an sipX, and freeswitch, and openser, and opbx |
| 01:44.30 | r0d3nt | ~nick aydiosmio |
| 01:44.42 | JT | <PROTECTED> |
| 01:44.43 | r0d3nt | hmm. i forget what it was.... |
| 01:45.06 | r0d3nt | did you guys have a lame nick checker ?? |
| 01:45.07 | JT | flenders: MACscr went through every other open source telephony channel, and gave up on them one by one |
| 01:45.15 | JT | flenders: whinging about how hard they were |
| 01:45.29 | MACscr | lol, i didnt whine about how hard they are |
| 01:45.30 | JT | flenders: and how the developers were stupid |
| 01:45.39 | MACscr | lol, i didnt say that either |
| 01:45.45 | JT | you didn't add me to ignore either, it's funny how people usually lie about that |
| 01:45.54 | flenders | hahahah |
| 01:46.08 | MACscr | actually i just got miranda and havent figured out how to get /ignore to work on it yet |
| 01:46.16 | JT | i know, no wizard? |
| 01:46.26 | *** join/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no) |
| 01:46.35 | nog | one observation... if you know how people usually dont add to ignore, either you are around a lot of people that need to be ignored or are the one... (not saying you are the one) |
| 01:46.56 | nog | or |
| 01:46.59 | MACscr | lol, nice one |
| 01:47.01 | nog | the person who isnt adding to ignore |
| 01:47.07 | JT | nog: being in channels like #freenode-social and #wikipedia, you see a lot of people reporting others as /ignored |
| 01:47.43 | MACscr | lol, thats funny that you were even in those channels |
| 01:47.47 | nog | hence, when you whois me.. you dont see me in any channels like that :P |
| 01:48.09 | JT | MACscr: why is that funny? |
| 01:48.33 | MACscr | JT: i dont feel like explaining humor |
| 01:49.05 | JT | alright, i guess i'll just accept the fact that it's hilarious being in #freenode-social and #wikipedia |
| 01:49.12 | JT | must be an in joke :) |
| 01:49.39 | masked | being in #wikipedia, why woulh you need to have humor explained to you? |
| 01:49.47 | *** join/#asterisk Opperior (n=chatzill@75.69.241.84) |
| 01:49.49 | nog | so, i hate it when i click on a download link.. and it opens up a new blank window and then the download dialog... i cant even begin to explain how stupid that is |
| 01:49.58 | JT | masked: eh? |
| 01:50.01 | *** join/#asterisk Strom_M (n=strom@12-189-87-2.att-inc.com) |
| 01:50.13 | nog | ahh.. masked, good one |
| 01:50.39 | *** part/#asterisk jjshoe (n=jjshoe@adsl-75-14-241-209.dsl.irvnca.sbcglobal.net) |
| 01:50.44 | masked | i'm not saying either of you are stupid |
| 01:50.44 | masked | hehe |
| 01:51.26 | JT | so does anyone actually have a problem with ASTERISK? |
| 01:51.40 | masked | why? so you can diss them? |
| 01:51.40 | nog | yes, trying to figure out why my phones dont ring :P |
| 01:51.46 | JT | which doesn't involve reading books being considered unnecessary |
| 01:51.57 | JT | nog: unfortunately you have a freepbx problem |
| 01:52.00 | JT | ~freepbx |
| 01:52.10 | jbot | freepbx is probably unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 01:52.10 | masked | lol JT |
| 01:52.20 | JT | masked: no, i help them :) |
| 01:52.39 | masked | hmm speaking of problems |
| 01:53.14 | masked | i call my ata on it's did or itsp number, and it just progresses for ~20s then puts me back to itsp voicemail |
| 01:53.29 | masked | about 5% of the time it actually rings |
| 01:53.39 | masked | spa-3000 btw. |
| 01:53.59 | JT | i am not very familiar with the spa-3000 |
| 01:54.00 | flenders | masked: is the ata registering with asterisk? |
| 01:54.34 | masked | yes. all outgoing calls work fine |
| 01:54.41 | masked | ports are open and forwarded |
| 01:54.51 | *** join/#asterisk poppo (n=adas@S0106004063d8e527.ed.shawcable.net) |
| 01:55.23 | flenders | do you see the call coming in on asterisk? |
| 01:55.25 | poppo | I need some help i am compiling asterisk 1.4 with cesptral and i am getting this error app_cepstral.c:48: warning: type defaults to 'int' in declaration of 'STANDARD_LOCAL_USER' |
| 01:55.32 | poppo | can somebody point me in the right direction |
| 01:55.34 | masked | flenders: yes |
| 01:55.43 | JT | poppo: does it still compile? |
| 01:55.46 | poppo | noop |
| 01:55.53 | JunK-Y | poppo: remove that. |
| 01:55.53 | JT | must be another error? |
| 01:56.03 | JT | WARNINGs are not fatal |
| 01:56.03 | poppo | well there is a bunch but the first one is that |
| 01:56.05 | JT | ERRORs are |
| 01:56.32 | flenders | masked: and is the ata registered to asterisk at the time of the call? |
| 01:56.37 | poppo | <PROTECTED> |
| 01:56.47 | poppo | thats the error |
| 01:56.47 | *** part/#asterisk nog (n=evan@c-69-180-239-206.hsd1.tn.comcast.net) |
| 01:56.56 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:57.09 | JunK-Y | poppo: where did ya get ur app from? |
| 01:57.15 | flenders | masked: wanna pastebin your sip.conf and dialplan? |
| 01:57.37 | poppo | the app_cepstrail.c from the http://www.automated.it/asterisk/ |
| 01:57.41 | masked | flenders: can't it's not my asterisk |
| 01:57.52 | JT | :o |
| 01:57.54 | masked | it's my itsp |
| 01:58.04 | JT | maybe your spa? |
| 01:58.09 | masked | so it should be fine, i think it's the spa |
| 01:58.10 | masked | yeah. |
| 01:58.41 | flenders | masked: I just asked you if you could see the call coming in on asterisk, and you said yes |
| 01:59.28 | sbingner | Dr-Linux, VMCOUNT |
| 01:59.37 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 02:00.07 | anonymouz666 | In a Macro can i concat things like.. ${ARG2}-blah |
| 02:00.12 | masked | flenders: ok. i just wanted to see where you were going |
| 02:00.57 | flenders | masked: hm, sorry then. can't help you |
| 02:01.01 | masked | np |
| 02:01.23 | flenders | masked: each ITSP have its own settings, and we have no idea what they are |
| 02:01.32 | masked | i'll do a firmware upgrade on the spa and see how it goes |
| 02:01.35 | flenders | masked: you'd be better off ringing their tech support |
| 02:01.47 | masked | flenders: argeed |
| 02:03.08 | *** join/#asterisk AJaymn (n=Me@66-188-80-40.dhcp.mdsn.wi.charter.com) |
| 02:03.19 | *** part/#asterisk AJaymn (n=Me@66-188-80-40.dhcp.mdsn.wi.charter.com) |
| 02:10.45 | *** join/#asterisk gammah (n=gammah@cpe-66-69-224-62.austin.res.rr.com) |
| 02:12.17 | anonymouz666 | when I got a call to some Macro inside let's say [from-did] context, and inside this macro I have something Set(blah=duh) - inside the context [from-did] the value of 'blah' will be 'duh'? |
| 02:13.06 | JT | i don't believe so |
| 02:13.22 | JT | you usually need to pass variables you want as args |
| 02:13.24 | anonymouz666 | then all my logic is wrong :S |
| 02:15.02 | anonymouz666 | the macro call is macro(macroname, name1, name2, name3) in macro arg1, arg2, arg3 ... but I am calling this macro inside from-did context |
| 02:15.13 | anonymouz666 | at some point name2 inside the macro got a value |
| 02:15.30 | anonymouz666 | I wanna know if I can use gotoif to compare name2 with something |
| 02:15.33 | JT | you can use NoOp or Verbose to help debug |
| 02:15.41 | JT | yes you can |
| 02:17.29 | poppo | i am getting error: 'STANDARD_HANGUP_LOCALUSERS' undeclared (first use in this function) |
| 02:17.29 | poppo | <PROTECTED> |
| 02:19.30 | anonymouz666 | JT but not inside the macro, but inside from-did context |
| 02:19.48 | JunK-Y | delete that line, ya dont need it in 1.4 |
| 02:20.04 | anonymouz666 | well I am doing the dialplan first, after done I will run everything to see tons of erros |
| 02:20.22 | JT | anonymouz666: so you haven't tried it yet? |
| 02:21.39 | anonymouz666 | I didn't try anything I am doing right now |
| 02:21.58 | anonymouz666 | I am just doing |
| 02:22.16 | JT | hrm it may be an idea to try |
| 02:22.24 | JT | stops you from going down the wrong path too far |
| 02:22.40 | anonymouz666 | yeap |
| 02:22.45 | anonymouz666 | you are right |
| 02:25.10 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:29.32 | anonymouz666 | when using labels |
| 02:29.47 | anonymouz666 | the correct is s,label) or s,(label)) |
| 02:30.56 | Strom_M | no |
| 02:31.12 | Strom_M | exten => s,n(label),Application() |
| 02:31.14 | Qwell | no , |
| 02:31.16 | Qwell | erm |
| 02:31.19 | Qwell | nm |
| 02:31.46 | Strom_M | was Re: Re: Re: Re: re: Re: FW: Re: Re: RE: No |
| 02:32.08 | Qwell | What, no fwd? |
| 02:32.11 | anonymouz666 | exten => s,n,GotoIf($[${passnumber_reply} = 0]?from-did-stage3,s,1:s,loop) |
| 02:32.22 | anonymouz666 | or s,(loop) |
| 02:32.47 | Strom_M | at the end of a gotoif, you'd write s,loop |
| 02:32.48 | *** join/#asterisk topping (n=topping@adsl-68-122-42-65.dsl.pltn13.pacbell.net) |
| 02:32.54 | Qwell | Why not just use While? |
| 02:32.57 | Strom_M | Qwell: there's a FW: in there |
| 02:33.04 | Qwell | Strom_M: fwd != fw :p |
| 02:33.19 | Strom_M | depends on the client |
| 02:33.23 | anonymouz666 | Qwell: there a label loop that calls a while |
| 02:33.52 | anonymouz666 | something like s,n(loop),While .... |
| 02:34.00 | *** join/#asterisk __freedom__lover (n=__freedo@200-206-253-206.dsl.telesp.net.br) |
| 02:34.21 | *** part/#asterisk __freedom__lover (n=__freedo@200-206-253-206.dsl.telesp.net.br) |
| 02:34.34 | Qwell | haha, I'm gonna write Duffs Device in dialplan logic |
| 02:35.09 | Qwell | http://en.wikipedia.org/wiki/Duff's_device |
| 02:36.51 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 02:36.52 | Qwell | where's Corydon-w ..? |
| 02:37.08 | Qwell | or codefreeze ... |
| 02:37.12 | Strom_M | i didn't hide them |
| 02:37.26 | Shaun2222 | hmm xlite doesnt really look to have NAT support... |
| 02:37.56 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 02:37.59 | Strom_M | the whole damn SIP protocol doesn't really have NAT support |
| 02:38.05 | Strom_M | much less xlite |
| 02:40.33 | anonymouz666 | http://rafb.net/p/s7Mt8y74.html |
| 02:40.44 | JT | Shaun2222: the sip server and nat device is much more important than the softphone, for getting throug NAT |
| 02:40.44 | anonymouz666 | Qwell would be better using execif? |
| 02:40.53 | anonymouz666 | Strom_M? |
| 02:41.13 | Strom_M | catsex? |
| 02:41.38 | Strom_M | um |
| 02:41.50 | Strom_M | you cant have two lines with priority 1 |
| 02:42.05 | anonymouz666 | oh that was typo |
| 02:42.17 | anonymouz666 | line 6 is wrong too... should jump to stage4 |
| 02:42.37 | anonymouz666 | what about the logic? there is a better way to do that ? |
| 02:42.52 | Strom_M | i cant debug logic by looking at a single piece of code |
| 02:43.01 | Strom_M | you've got to show me the whole dead hooker |
| 02:43.30 | bkruse_home | Strom_M: you still in town? |
| 02:43.37 | Strom_M | yup |
| 02:43.38 | mihinomenest | well, I've never heard that phrase applied to debugging before. |
| 02:43.55 | Strom_M | haha |
| 02:43.58 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 02:44.00 | Strom_M | well, it does fit, right? |
| 02:44.06 | anonymouz666 | Strom_M: do you wanna see the macro ? |
| 02:44.14 | Strom_M | i mean, a dead hooker is just as useful as a broken macro :) |
| 02:44.20 | JT | i think he wants to see extensions.conf |
| 02:44.28 | Strom_M | exactly |
| 02:45.21 | anonymouz666 | ok |
| 02:45.47 | Strom_M | it's like giving me a single bolt and asking "is this a good choice for repairing my car?" |
| 02:46.39 | mog | lol |
| 02:47.50 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 02:48.35 | infinity1 | in ael, how do you append variables? |
| 02:50.33 | infinity1 | e.g. PHONES="${EXT205}"; |
| 02:51.50 | anonymouz666 | Strom_M http://rafb.net/p/MeRJTh19.html |
| 02:51.56 | *** join/#asterisk jjshoe (n=jjshoe@adsl-75-14-241-209.dsl.irvnca.sbcglobal.net) |
| 02:52.26 | jjshoe | has anyone ran into call files scheduled into the future not being sent? |
| 02:52.31 | Strom_M | anonymouz666: holy cocks man, what are you trying to actually accomplish? |
| 02:52.47 | jjshoe | I've verified with stat that the time is in the future Access: 2007-03-20 19:47:35.000000000 -0700 |
| 02:52.47 | jjshoe | Modify: 2007-03-20 19:47:35.000000000 -0700 |
| 02:53.07 | jjshoe | now that the time is past, it's not placing these calls, is there any way I can find out why? |
| 02:53.18 | anonymouz666 | Strom_M: what's so wrong? :) |
| 02:53.41 | Strom_M | anonymouz666: it looks elephantine |
| 02:53.46 | Strom_M | what are you trying to do? |
| 02:54.29 | anonymouz666 | read number 1 and number 2 and compare, is it ok? go to the next stage |
| 02:54.43 | Strom_M | no no no no |
| 02:54.47 | anonymouz666 | if not enter while loop for N times and give up |
| 02:54.53 | Strom_M | what are you trying to ACCOMPLISH |
| 02:54.57 | Strom_M | what's the point of all of this |
| 02:55.12 | Strom_M | not "how do you get there" but "where are you trying to go?" |
| 02:56.04 | anonymouz666 | compare two numbers and if its ok go to the next stage.. thats what i am trying to do |
| 02:56.10 | Strom_M | GAH NO |
| 02:56.13 | Strom_M | i give up |
| 02:56.43 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 02:56.45 | anonymouz666 | ok, thanks anyway. |
| 02:56.48 | JT | anonymouz666: he's asking a very simple question |
| 02:56.51 | Strom_M | this is like me asking you which floor you want to get to in the elevator, and you responding "I want to push buttons" |
| 02:57.05 | Strom_M | it's not the means I'm after, it's the end |
| 02:57.22 | JT | anonymouz666: what is the task that will be accomplished? |
| 02:57.30 | JT | in a 1 line summary |
| 02:57.38 | infinity1 | argh ..is it possible to concatinate variables in AEL? |
| 02:57.57 | anonymouz666 | read data and post data to another server |
| 02:58.02 | Strom_M | yes, complete this sentence, anonymouz666: "The problem I am trying to solve by using this macro is: _________________" |
| 02:59.44 | anonymouz666 | Strom_M: you don't need to help if you don't want. what you are doing is not necessary. |
| 03:00.02 | Strom_M | anonymouz666: but I want to help you |
| 03:00.17 | Strom_M | it's you who isn't helping me help you |
| 03:00.24 | JT | anonymouz666: it is, it makes it easier for use to helpo you if we can determine if there is a mechanism to do what you want much easier |
| 03:01.03 | JT | either calling card or recharging a prepai account is my guess |
| 03:01.07 | Strom_M | the whole reason I'm asking these questions is because I think you've made a huge mistake somewhere along the line and therefore want to help you build a more elegant solution |
| 03:01.10 | JT | either way being secrative won't help |
| 03:01.22 | Strom_M | JT: I think it's hopeless. |
| 03:01.38 | Strom_M | it's a whole screenfull of 9 point text already |
| 03:02.54 | anonymouz666 | I just need to compare two values and send it to another server through POST... What will be done in the server with the data I don't know. It's a black box. |
| 03:04.18 | anonymouz666 | it's a IVR system that read data and send data through a post using curl. |
| 03:05.25 | anonymouz666 | english it's not my native language maybe I don't know how to explain correct - it must be that |
| 03:05.48 | JT | anonymouz666: what sort of data does the blakbox need? |
| 03:05.52 | JT | what does it return? |
| 03:06.41 | anonymouz666 | it returns integers that will be stored on ${arg3} |
| 03:07.22 | anonymouz666 | the data that blackbox need is ${arg1} that will be read from IVR |
| 03:07.53 | JT | what sort of data is it, apart from an integer? is it a phone number? |
| 03:08.01 | anonymouz666 | 0 or 1 |
| 03:08.13 | anonymouz666 | thats why you see a gotoif comparing with 0 |
| 03:08.40 | mmartinn | He's creating a serial connection over an ivr...? |
| 03:08.47 | JT | but what does the box actually do? |
| 03:08.59 | JT | mmartinn: http connection to a black box apparently |
| 03:09.12 | mmartinn | JT: it sounds like it's emulating a null modem with 0s and 1s |
| 03:09.12 | anonymouz666 | yes it is http connection |
| 03:09.31 | anonymouz666 | lol no |
| 03:09.31 | mmartinn | literally transferring binary data over an ivr into http |
| 03:09.50 | mmartinn | that's hard core 8-) |
| 03:10.21 | JT | i think it's only one thing where it's asking for a 1 or 0 result |
| 03:10.29 | JT | maybe it's a secret government machine |
| 03:10.40 | anonymouz666 | yeah CIA stuff. |
| 03:10.44 | mmartinn | oh (I didn't look at the post)... someone slap me before speaking without thinking first |
| 03:10.47 | mmartinn | =) |
| 03:11.47 | anonymouz666 | thanks anyway guys. |
| 03:12.32 | anonymouz666 | the intention is always valid :) |
| 03:12.50 | mmartinn | that's a crazy dialplan |
| 03:12.54 | Strom_M | cocks |
| 03:12.57 | Strom_M | on a stick |
| 03:13.07 | anonymouz666 | crazy dialplan haha |
| 03:13.11 | codefreeze | infinity1: question answered about AEL? |
| 03:13.32 | anonymouz666 | that's a beautiful and clear dialplan |
| 03:13.38 | JT | :o |
| 03:14.13 | anonymouz666 | the problem is that nobody understand :D |
| 03:14.20 | mmartinn | I would resort to AGI; for some reason I find the dialplan format disagreeable. |
| 03:14.31 | codefreeze | infinity1: Set(x=${var1}${var2}); |
| 03:14.39 | *** join/#asterisk oej_ (n=olle@p5485f9f9.dip.t-dialin.net) |
| 03:14.43 | mmartinn | I don't mind it for simple things, but it gets crufty quickly. |
| 03:18.14 | mmartinn | it's making me sleepy...zzzzzz |
| 03:18.16 | mmartinn | nite ;) |
| 03:19.51 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 03:20.57 | *** part/#asterisk jjshoe (n=jjshoe@adsl-75-14-241-209.dsl.irvnca.sbcglobal.net) |
| 03:21.28 | *** join/#asterisk shodan- (n=shodan@ip124.96-113-216.pppoe1.joliette.intermonde.net) |
| 03:22.44 | delmar | Hi everyone. I just experienced something a little strange. I called a business, and was placed on hold (they say they use *7) but the hold was actually initiated on my own Asterisk server, and I heard my own hold music. They have a purely digital phone system to the telco, and that telco has an interconnect to my telco/VISP etc. It would seem that their 'hold' command was passed all the way to my server. It all worked fine.. |
| 03:22.44 | delmar | . their hold worked as normal as far as they were concerned... but how weird is that. Shouldnt there be some sort of block in place somewhere to prevent this? |
| 03:22.54 | delmar | is there something i need to do in Asterisk to stop this or what? |
| 03:24.07 | JT | features.conf |
| 03:24.25 | JT | you should never let external parties activate your features or trasfers |
| 03:24.29 | Strom_M | delmar: what's so wrong about that? |
| 03:24.38 | Strom_M | don't you like listening to your own hold music? |
| 03:24.51 | delmar | Strom_M, because the correct method would be that they put me on hold and I hear their hold queue not my own |
| 03:25.04 | delmar | JT, there is nothing obvious in my features.conf that I can see |
| 03:25.15 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 03:25.24 | Strom_M | delmar: the whole point of ISDN and SS7 is to not transfer unnecessary audio |
| 03:25.27 | delmar | JT, all i ahve activatesd in there is the parking stuff. |
| 03:25.45 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 03:25.45 | Strom_M | you're trying to apply analog thought to a digital signaling world |
| 03:25.48 | Strom_M | doesnt work |
| 03:25.52 | JT | umm |
| 03:25.58 | delmar | Strom_M, u are right. i dont really have a problem with it.. i think its kinda cool actually... but its still not really the correct way things are done. |
| 03:26.01 | aptura | Strom that is for call setup right |
| 03:26.11 | JT | Strom_M: surely it shouldn't be activting his on hold music |
| 03:26.41 | Strom_M | "not the correct way things are done"? |
| 03:26.42 | delmar | what Strom_M is suggesting is that its better to have locally generated hold music than pipe it all over the digital paths from someone else... |
| 03:26.42 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:26.48 | Strom_M | did you work for Bell Labs for 30 years? |
| 03:26.56 | delmar | and I do agree from an efficiency perspective..... |
| 03:27.32 | delmar | Strom_M, but the expected way things operate .. for me.. and for customers... is that when a company puts me on hold.. I hear THEIR hold music.. and sometimes.. sadly.. hear their advertising... |
| 03:27.40 | delmar | Strom_M, is that not the industry standard ? |
| 03:29.01 | JT | what q.931 messages would they be sending to cause hold music? |
| 03:29.19 | JT | and yes, i have never heard of another system putting your own hold music on, digital or not |
| 03:29.22 | delmar | JT, I guess I would need to debug the thing eh? |
| 03:29.34 | JT | delmar: does anything come up in console? |
| 03:29.37 | Strom_M | yeah, i'm curious which isdn message you're getting |
| 03:29.47 | delmar | JT, yeah.. thats waht I expect... they put me on hold on their system. not put me on hold on my own system. |
| 03:29.53 | Strom_M | guh no |
| 03:29.56 | Strom_M | you're not listening :) |
| 03:30.01 | Strom_M | it's /one call/ |
| 03:30.16 | delmar | heh. |
| 03:30.28 | Strom_M | where you get put on hold is not important; the only important bit is tht you get put on hold oh fuck it, I'm not going to be able to get you to listen to me |
| 03:30.47 | delmar | Strom_M, no .. im interested.. give it a try. |
| 03:31.11 | noli_OF | <PROTECTED> |
| 03:31.24 | Strom_M | ~101 |
| 03:31.35 | jbot | methinks 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 03:31.36 | Strom_M | go read that |
| 03:31.37 | delmar | bot not working? |
| 03:31.39 | delmar | oh there we go |
| 03:32.41 | delmar | JT, yeah .. on my Asterisk servers console I see the message such as Started music on hold, class 'default', on IAX2/blah-1 etc |
| 03:33.40 | delmar | I have a SIP connection from my Asterisk to my VSP / DID provider... i cant comment accurately what they have from there but the business I called also has a digital connection to their Telco. |
| 03:34.17 | delmar | ISDN the guy reckons |
| 03:34.27 | Strom_M | duh, ISDN |
| 03:34.30 | delmar | and they have some kinda alcatel digital phone system |
| 03:34.44 | delmar | and their command for hold is *7 .. |
| 03:34.49 | JT | delmar: if you watch full log, with debug on, you can see what triggers it |
| 03:35.17 | delmar | yeah. let me go see about that. bbs |
| 03:35.25 | Strom_M | it's an isdn message |
| 03:35.29 | Strom_M | it's not the feature code |
| 03:35.38 | JT | Strom_M: he has an itsp though |
| 03:35.46 | JT | so it won't be in pri intense debug |
| 03:35.48 | Strom_M | it's all ISUP on the back end anyway |
| 03:35.51 | JT | maybe sip debug might have it |
| 03:39.50 | delmar | ok.. i captured a sip debug of the hold event.... scrolling back.... what am i looking for? |
| 03:41.11 | Strom_M | pastebin the sip debug |
| 03:41.26 | Strom_M | well actually |
| 03:41.31 | Strom_M | it'll be iax2 debug, wont it? |
| 03:41.37 | Strom_M | since the provider is iax2, not sip |
| 03:41.45 | delmar | they are sip |
| 03:42.31 | flenders | is there a way to filter sip debug messages to a single channel? |
| 03:42.35 | Strom_M | plz2pastebin |
| 03:42.46 | JT | flenders: by ip, yes |
| 03:43.32 | flenders | JT: oh, just saw that |
| 03:45.29 | delmar | ok done. can i priv message u the link? |
| 03:45.42 | JT | can't it go in channel? |
| 03:45.54 | delmar | kine like my privacy. |
| 03:45.57 | delmar | kinda* |
| 03:46.11 | delmar | but then.. |
| 03:46.18 | flenders | delmar: I hope you had all passwords and DIDs removed |
| 03:46.18 | delmar | pastebin is googleable anyway |
| 03:46.19 | delmar | http://www.pastebin.ca/404624 |
| 03:46.26 | delmar | pfft |
| 03:46.30 | JT | you can make pastbins expire |
| 03:46.30 | delmar | who cares. |
| 03:46.46 | delmar | there are no passwords in there. just numbers. |
| 03:46.51 | JT | that's right |
| 03:47.01 | CunningPike | 1.4.2, eh? |
| 03:47.24 | delmar | me? no. |
| 03:48.55 | *** join/#asterisk `p4r14h (n=j0sh@69.92.145.178) |
| 03:49.39 | *** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net) |
| 03:49.40 | infinity1 | argh ..is it possible to concatinate variables in AEL? can i get an example? |
| 03:49.47 | delmar | so the question is... how to stop the 'hold' action occurring when its received .. from that provider etc. |
| 03:52.08 | codefreeze | infinity1: Set(x=${var1}${var2}); |
| 03:52.25 | kiwoneka | good evening |
| 03:52.40 | kiwoneka | i am hoing i can get some help |
| 03:52.47 | kiwoneka | with hinting |
| 03:52.48 | *** join/#asterisk BigCanOfTuna (n=arustad@dsl-mac-66-18-226-119-cgy.nucleus.com) |
| 03:53.16 | delmar | CunningPike, should i be using 1.4.x? |
| 03:53.17 | BigCanOfTuna | Where can I find some up to date info about running Asterisk on OS X? voip-info.org seems to be very out of date. |
| 03:53.28 | kiwoneka | http://www.voip-info.org/wiki/view/Asterisk+presence |
| 03:53.34 | *** join/#asterisk kgx0 (n=karuna@60.234.196.160) |
| 03:53.35 | kiwoneka | i have tried that |
| 03:53.48 | CunningPike | delmar: I was merely commenting that 1.4.2 is out |
| 03:53.58 | kiwoneka | and i am not getting the result i had hoped |
| 03:54.00 | delmar | ah right'o. |
| 03:54.06 | delmar | been thinking of trying it out sometime myself. |
| 03:54.12 | codefreeze | infinity1: does that help? |
| 03:55.03 | kiwoneka | i have polycom 601s |
| 03:55.29 | kiwoneka | i would like the the other extensions to be indicated when they are on the phone |
| 03:55.36 | delmar | JT Strom_M , waiting for your thoughts on that. I gotta go get my gurl in the bath. back in a few. |
| 03:55.36 | [TK]D-Fender | kiwoneka, You need to enable presence support in your provisioning |
| 03:55.54 | JT | delmar: nothing obvious so far |
| 03:56.06 | [TK]D-Fender | kiwoneka, Do you have a "buddies" soft-key on under the 4th soft-key while on idle? |
| 03:56.07 | JT | might be something obvious that a sip expert will catch :) |
| 03:56.56 | kiwoneka | yes i do |
| 03:57.09 | kiwoneka | http://www.voip-info.org/wiki/view/Asterisk+presence that is what i ahve done |
| 03:57.14 | kiwoneka | followed |
| 03:57.28 | kiwoneka | but, i dont fully understand |
| 03:58.14 | [TK]D-Fender | kiwoneka, and you enabled "watch buddy" in your contact directory entry that you want to watch? |
| 03:58.48 | [TK]D-Fender | kiwoneka, You should see a "head" like IM icon next to their speed-dial entry for status |
| 03:58.53 | kiwoneka | that is in the phone provisioning |
| 03:59.05 | [TK]D-Fender | kiwoneka, no, that is in the CONTACT directory |
| 03:59.29 | kiwoneka | so i have to add the extension to my contact list |
| 04:00.14 | [TK]D-Fender | kiwoneka, Yes. It doesn't magicall choose who to watch, you have to tell it. |
| 04:01.17 | kiwoneka | so this is not enough |
| 04:01.18 | kiwoneka | exten => 771,hint,SIP/771 |
| 04:01.18 | kiwoneka | exten => 772,hint,SIP/772 |
| 04:01.18 | kiwoneka | exten => 773,hint,SIP/773 |
| 04:01.40 | [TK]D-Fender | kiwoneka, That only tells * what a phone CAN ask to know about. the phone has to CARE <- |
| 04:01.45 | JT | delmar: |
| 04:01.47 | JT | <PROTECTED> |
| 04:01.51 | JT | c=IN IP4 203.184.16.2 |
| 04:02.09 | JT | i think the reinvite to 0.0.0.0 tells asterisk to hold |
| 04:02.10 | kiwoneka | ok |
| 04:02.12 | JT | is my guess |
| 04:02.20 | file | that's one of the ways to signal hold. |
| 04:02.27 | kiwoneka | that is a good explanation |
| 04:02.31 | kiwoneka | thank you |
| 04:02.31 | JT | file: can asterisk ignore it? |
| 04:02.43 | kiwoneka | let me see if i can make the changes you suggested |
| 04:02.50 | file | ignore it? and do what? |
| 04:03.08 | [TK]D-Fender | kiwoneka, So go add a contact to your contact directory, . contact = exten to watch, and scroll further down to enable "watch buddy". |
| 04:03.09 | JT | file: ignore the reinvite request and continue to pass rtp as it was previously doing |
| 04:03.15 | [TK]D-Fender | kiwoneka, You can.... |
| 04:03.27 | file | JT: no, that wouldn't be proper... and the phone would just ignore it anyway |
| 04:03.50 | JT | file: delmar is hearing his own on hold music instead of the other party's (on PSTN), and does not want to |
| 04:03.50 | file | and it wouldn't send a stream of audio in to send to the other person either |
| 04:04.00 | JT | hmm |
| 04:04.36 | JT | surely there must be a way around it, as it doesn't sound very normal |
| 04:04.56 | JT | on hold signalling passed all the way from far end telco lines to itsp to him |
| 04:05.11 | file | never heard of it before, how amusing |
| 04:05.24 | JT | maybe his itsp is misconfigured? |
| 04:05.39 | JT | file: you haven't heard of it before? |
| 04:06.05 | JT | i must say i've never come across it |
| 04:06.08 | file | I've never heard of someone having the issue |
| 04:06.17 | JT | yeah it's a weird one |
| 04:06.52 | Strom_M | its happened to me |
| 04:07.09 | Strom_M | but only in situations where I'm talking to another asterisk box directly with IAX2 |
| 04:07.26 | file | yeah... 1.4+ will pass those through |
| 04:07.37 | Strom_M | this was in 1.2 and maybe even 1.9 |
| 04:07.38 | Strom_M | er |
| 04:07.40 | Strom_M | 1.0 |
| 04:08.05 | JT | would setting sip reinvite to no help him? |
| 04:08.19 | Strom_M | maybe |
| 04:08.24 | Strom_M | it looks like an INVITE which sets it off |
| 04:08.44 | file | it won't, canreinvite controls Asterisk sending out reinvites |
| 04:08.53 | JT | damn |
| 04:08.56 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 04:09.23 | JT | he can't tell his itsp's sip server to "get stuffed" when it tried to reinvite to on hold? ;) |
| 04:09.28 | JT | tries |
| 04:09.28 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:09.53 | kiwoneka | is it possible to send a reboot command to polycom phones from cli? |
| 04:10.26 | SwK | kiwoneka: sip notify polycom cfg-check or something like that |
| 04:10.39 | SwK | but you have to have modified the polycoms configs for that to work |
| 04:10.58 | SwK | (simplest way to ensure they reboot is add 1 space to their mac.cfg file) |
| 04:11.10 | file | it's a Silik0n! |
| 04:11.15 | [TK]D-Fender | kiwoneka, yes, there is a check-config packet you can send out that if your provisioning allows will trigger a reboot |
| 04:11.17 | SwK | its a file |
| 04:11.24 | [TK]D-Fender | kiwoneka, Though you shouldn't HAVE to... |
| 04:11.39 | file | SwK: how are you? |
| 04:12.02 | kiwoneka | i made the changes you sugguested the global contact list |
| 04:12.37 | kiwoneka | in this file located in the contact dir 000000000000-directory.xml |
| 04:12.52 | SwK | 'sip notify polycom' is the command from the asterisk cli |
| 04:13.13 | SwK | file: fine getting annoyed by dumb people again |
| 04:13.36 | [TK]D-Fender | kiwoneka, No, taht file is now officially USELESS to you. |
| 04:13.59 | [TK]D-Fender | kiwoneka, It only gets copied over the very first time you provision your phone |
| 04:13.59 | kiwoneka | ? |
| 04:14.24 | [TK]D-Fender | kiwoneka, your phone now uses <mac>-directory.xml for its contacts |
| 04:14.34 | kiwoneka | yes i know that |
| 04:14.39 | kiwoneka | i erased it |
| 04:14.51 | kiwoneka | so that it copies the 0000 |
| 04:14.54 | kiwoneka | file |
| 04:14.55 | [TK]D-Fender | kiwoneka, It won't recopy.... jsut enter it direct on the phone itself. |
| 04:15.04 | kiwoneka | ? |
| 04:15.12 | SwK | so any telemarketers around tonight? PM me |
| 04:15.20 | kiwoneka | ok |
| 04:15.41 | infinity1 | codefreeze: nope. that didn't work |
| 04:16.16 | JT | SwK: "any telemarketers around tonight? call me, call me now" ;) |
| 04:16.16 | kiwoneka | does that mean that i have to go to buddies to see the active state of the other extensions |
| 04:16.28 | kiwoneka | can't i get one of the six lines to light up |
| 04:16.45 | kiwoneka | to identify that an extension is in use |
| 04:21.05 | [TK]D-Fender | kiwoneka, If you have a free line key and specify a "speed dial index" for the contact they will appear in index order on each available line-key. They overflow similarly on the 601 Attendant Modules |
| 04:22.26 | kiwoneka | i have enabled it the watch buddy on all the extensions |
| 04:22.44 | infinity1 | did anyone see sanjia on american idol ...all i gotta say is ..WTF |
| 04:22.53 | kiwoneka | but they are still no indicating that they are busy |
| 04:22.53 | SwK | JT: something like that |
| 04:22.55 | wunderkin | heh |
| 04:23.22 | SwK | file: and I dont wanna cookie... but I'll gladly accept cash or paypal |
| 04:24.06 | file | SwK: Canadian cash? |
| 04:24.11 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 04:25.04 | [TK]D-Fender | kiwoneka, Well lets actually see you * side of things. patebin your extensions.conf releven contexts, and the sip.conf entires for the, masking only passwords |
| 04:25.06 | [TK]D-Fender | ~pb |
| 04:25.13 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 04:25.13 | Strom_M | loonie money |
| 04:30.13 | infinity1 | how can i do something like this in the global section of extensions.ael: Set(PHONES=${EXT205}); |
| 04:30.31 | kiwoneka | http://www.pastebin.ca/404675 |
| 04:32.25 | *** join/#asterisk CrashHD (n=crashhd@67.166.155.233) |
| 04:32.32 | [TK]D-Fender | kiwoneka, take a look at the CLI output of "show hints". |
| 04:35.56 | kiwoneka | -= Registered Asterisk Dial Plan Hints =- |
| 04:35.56 | kiwoneka | <PROTECTED> |
| 04:35.56 | kiwoneka | <PROTECTED> |
| 04:35.56 | kiwoneka | <PROTECTED> |
| 04:35.56 | kiwoneka | ---------------- |
| 04:35.57 | kiwoneka | - 3 hints registered |
| 04:36.01 | kiwoneka | sorry |
| 04:37.16 | [TK]D-Fender | kiwoneka, Place a call, look at it again and see if it changes |
| 04:37.49 | [TK]D-Fender | kiwoneka, And when you look directly in "buddies" you don't see the entry listed as being watched? |
| 04:38.19 | aptura | TK I am looking for message and debug and cannot locate them. Did asterisk change the directory where the sip debug info is stored? |
| 04:38.30 | [TK]D-Fender | kiwoneka, the "watchers" count says how many phones are looking at a given hint. |
| 04:38.50 | [TK]D-Fender | aptura, I never used output log files.... |
| 04:39.09 | aptura | okay where is sip debug stored then |
| 04:39.26 | kiwoneka | the buddies are there, but all they all say 'online' |
| 04:39.42 | aptura | need to get this NEW problem resolved with my incomming DID no audio issue. |
| 04:39.46 | kiwoneka | the cli while on a call they all say idle |
| 04:39.50 | kiwoneka | the state |
| 04:40.06 | [TK]D-Fender | kiwoneka, as in you take one of those phones, call up Voicmailmain and just sit on an active call. the "show hints" says NOTHING? |
| 04:40.31 | kiwoneka | i just did that on two extensions |
| 04:40.41 | kiwoneka | they did nothing |
| 04:40.42 | [TK]D-Fender | kiwoneka, pastebin the entire CLI output of that |
| 04:40.49 | kiwoneka | ok |
| 04:40.55 | kiwoneka | let me do it again |
| 04:41.05 | [TK]D-Fender | kiwoneka, You should be able to scroll back for it |
| 04:43.11 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 04:47.05 | kiwoneka | http://www.pastebin.ca/404687 |
| 04:47.09 | kiwoneka | there |
| 04:47.29 | Qwell | <rant> |
| 04:47.47 | Qwell | Whoever the freaking idiot over at Microsoft was that decided that Vista was release ready... |
| 04:47.57 | Qwell | Should be shot. Repeatedly. |
| 04:48.12 | kiwoneka | where is the petition |
| 04:48.18 | kiwoneka | where do i sign |
| 04:48.29 | Qwell | How can you release an OS that *KILLS* hardware? |
| 04:48.39 | kiwoneka | i have a huge headace supporting the fool that have upgraded |
| 04:48.40 | Qwell | I mean, it's par for the course, but come on |
| 04:48.46 | [TK]D-Fender | Qwell[], Whoever believed that Vista was anywhere near ready and actually bought it is already living out their sentence :) |
| 04:48.53 | Qwell | [TK]D-Fender: I didn't buy it ;) |
| 04:49.02 | kiwoneka | i have been uninstalling and installing ubuntu |
| 04:49.07 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:49.14 | [TK]D-Fender | Qwell[], Enjoy your position in proxy :D |
| 04:49.50 | [TK]D-Fender | kiwoneka, I don't see you calling "show hints" in the middle there. also, please attempt again starting with "verbose 10" |
| 04:50.05 | kiwoneka | ok |
| 04:53.30 | kiwoneka | http://www.pastebin.ca/404695 |
| 04:53.46 | Mercestes | Hm. I had no problems with Vista. What hardware did it kill?? |
| 04:53.50 | kiwoneka | i think my sip.conf may be missing something |
| 04:55.03 | [TK]D-Fender | kiwoneka, Why am I still not seeing you typing "show hints" in the MIDDLE of that call like I've been asking? |
| 04:55.08 | denon | Qwell: kills hardware? or just doesn't support it? |
| 04:55.22 | [TK]D-Fender | kiwoneka, if those are legit phone entries, they sshould be fine |
| 04:55.35 | [TK]D-Fender | kiwoneka, 773 should light up... |
| 04:55.49 | Qwell | denon: kills it |
| 04:57.01 | [TK]D-Fender | Qwell, Don't think "complete loss of hardware", think "upgrade opportunity!" |
| 04:57.41 | denon | Qwell: how so? |
| 04:57.47 | Qwell | got me |
| 04:57.50 | infinity1 | do nested includes in AEL2 actually work as described in the example included with asterisk? |
| 04:57.50 | denon | well .. |
| 04:57.52 | aptura | who here works for xorcom rapid |
| 04:57.53 | denon | I mean how is it dead |
| 04:57.55 | Qwell | upgrade a driver, and bam, dead |
| 04:58.00 | denon | oh .. heh |
| 04:58.00 | Qwell | it's dead...it ceased to be |
| 04:58.10 | denon | that's the driver code then, not the OS |
| 04:58.14 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:58.28 | denon | what kind of device? you do know that a lot of hardware "drivers" are both drivers and a firmware flash, I assume.. |
| 04:59.07 | [TK]D-Fender | infinity1, type "show dialplan" at CLI and find out. |
| 04:59.36 | [TK]D-Fender | aptura, That'd be tzafrir I believe |
| 04:59.52 | aptura | k |
| 05:00.04 | kiwoneka | Extension Changed 773 new state Ringing for Notify User 772 |
| 05:00.12 | kiwoneka | is that what your looking for |
| 05:00.21 | infinity1 | [TK]D-Fender: oh neeto! :) ..thanks |
| 05:00.35 | [TK]D-Fender | kiwoneka, then 722 should see that 733 is "busy" |
| 05:00.49 | kiwoneka | it does not |
| 05:00.58 | kiwoneka | the buddies just say online |
| 05:01.01 | kiwoneka | that is it |
| 05:01.08 | [TK]D-Fender | get rid of the "subscribecontext |
| 05:01.09 | infinity1 | [TK]D-Fender: i don't suppose you know how to concatinate variables in AEL ? |
| 05:01.21 | [TK]D-Fender | in your sip.conf entries, and do a SIP reload. |
| 05:01.55 | [TK]D-Fender | infinity1, No... AEL is a nearly complete waste of dev time better spent improving chan_sip or something that we really NEED.... |
| 05:01.56 | aptura | btw this is a little odd but ever see a case of outgoing DID audio does not work but incomming DID two way audio is good? I wonder if it could be my wholesaler. |
| 05:01.59 | *** join/#asterisk Mercestes (n=Merceste@cpe-24-175-82-3.houston.res.rr.com) |
| 05:02.08 | [TK]D-Fender | </bile></venom> |
| 05:02.42 | [TK]D-Fender | aptura, Bad peer entry. |
| 05:03.01 | infinity1 | [TK]D-Fender: should i be using agi instead of ael? or old-stype extensions? |
| 05:03.10 | infinity1 | er s/stype/style/ |
| 05:03.12 | aptura | good point |
| 05:03.16 | [TK]D-Fender | infinity1, Good 'ole extensions.conf |
| 05:03.55 | [TK]D-Fender | <- Zen master of the blatantly obvious |
| 05:05.55 | aptura | TK both are friend. |
| 05:06.30 | aptura | Even though thay normally should not be . When you say bad peer what do you mean? |
| 05:07.17 | *** join/#asterisk hardwire (n=bip@rdbck-1277.palmer.mtaonline.net) |
| 05:07.26 | kiwoneka | [TK]D-Fender, i thank yo for your patience, my little one up |
| 05:07.33 | [TK]D-Fender | aptura, friend is both peer and user. it could be however that they are send calls TO you un-authed and thats why they are coming through. |
| 05:07.37 | kiwoneka | i gotta get her back to sleep |
| 05:07.45 | kiwoneka | i will try again tomorrow |
| 05:07.49 | [TK]D-Fender | kiwoneka, k |
| 05:08.00 | kiwoneka | thank you |
| 05:08.07 | kiwoneka | good night to all |
| 05:09.03 | aptura | i think i see the issue. |
| 05:14.37 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 05:17.34 | [TK]D-Fender | ok, checkout time... later all... |
| 05:23.39 | *** join/#asterisk Splat (n=splat@eth112.tas.adsl.internode.on.net) |
| 05:41.23 | *** join/#asterisk rrrobert (n=rrobert@58-65-160-140.nayatel.pk) |
| 05:44.11 | aptura | I get two seconds of audio out on the DID and then it dies. |
| 05:44.38 | aptura | DId-OUT audio is a issue. DID-in no issue. |
| 05:44.48 | aptura | Possible my wholsaler has a issue. |
| 05:56.34 | *** join/#asterisk salviadud (n=noyb@189.156.171.163) |
| 05:56.42 | salviadud | could someone help me out |
| 05:56.46 | salviadud | what does this mean? |
| 05:56.55 | salviadud | Incoming call: Got SIP response 503 "Server error" back from 189.156.171.163 |
| 05:57.05 | salviadud | that ip is me |
| 05:57.26 | salviadud | i'm trying to register a sip device from somewhere else on the internet |
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| 06:01.21 | salviadud | hey |
| 06:01.26 | salviadud | is anyone awake? |
| 06:01.48 | dahunter3 | Anyone freelance? I'm having a hell of a time getting my asterisk box's T1 to work with the phone company. It just thinks everything is busy or busies it out-- who knows |
| 06:02.13 | dahunter3 | e&m wink |
| 06:04.03 | JT | e&m wink, can you get it changed to pri? it's much nicer |
| 06:06.05 | *** join/#asterisk Fieldy (i=QCGUCORo@gentoo/contributor/Fieldy) |
| 06:06.38 | dahunter3 | JT: Yeah, everyone says that--- but it's $600 versus $200 per month |
| 06:07.08 | JT | hmm |
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| 06:10.46 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 06:22.13 | *** join/#asterisk k-man_ (n=jason@unaffiliated/k-man) |
| 06:22.16 | k-man_ | hello |
| 06:22.34 | ta^3 | Hi! |
| 06:22.35 | ta^3 | I have a funny problem with a dialplan. |
| 06:22.36 | k-man_ | what sort of codec can one use to try and acheive the same voice quality as pstn? |
| 06:22.45 | k-man_ | ta^3, oh, thats funny |
| 06:22.45 | ta^3 | I receive zapata calls to 'inbound' context, from this context I jump to 'dids' context where I define the actions per DID or default ones like XXXX,1,Goto(menu|s|1). no problem for this. the problem begins at 'menu' context, where i have an include to context local and local have various includes. in the s exten on menu context I set timeouts, and playback a welcome greeting also t,1, is defined. at priority 5 on menu context (menu,s,5) I have a WaitExten |
| 06:22.45 | ta^3 | . If any context included on 'local' context have a s,6 priority when menu,s,5 waitexten finish the call jump to othercontext,s,6 rather than menu,t,1. any clues? |
| 06:22.56 | JT | hello k-man_ |
| 06:23.01 | ta^3 | k-man_: ulaw/alaw |
| 06:23.01 | *** join/#asterisk Splat (n=splat@eth112.tas.adsl.internode.on.net) |
| 06:23.06 | k-man_ | hi jt, hows it going? |
| 06:23.28 | JT | not bad |
| 06:23.37 | k-man_ | ta^3, but do you have to up the bitrates to acheive that quality? |
| 06:24.55 | ta^3 | k-man_: not sure. i guess not. |
| 06:24.59 | JT | g.711 u/a uses 64kbit/s for the codec |
| 06:25.37 | k-man_ | jt, and acheives same quality as pstn? |
| 06:25.57 | JT | yes as long as your link has no issues |
| 06:26.04 | JT | like lack of bandwidth |
| 06:26.07 | JT | lag |
| 06:26.09 | JT | jitter |
| 06:26.20 | JT | the pstn uses g.711 |
| 06:26.24 | JT | alaw in australia |
| 06:31.43 | FuriousGeorge | is it fair to say that if SIP is working with NAT on both ends then SIP Video will probably work as well? |
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| 06:36.10 | lokkju_wrk | FuriousGeorge, yes |
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| 06:47.27 | djPepse | Evening, gents. |
| 06:48.15 | djPepse | I'm wondering if my problem is a known issue (if I should bothering rebuilding to upgrade).. I'm running 1.4.0-beta3, connected with an unlocked packet8 ata.. After a while of operation, callerid stops working |
| 06:48.32 | djPepse | Just gives unknown name/number for everything, even though callerid is coming through fine |
| 06:53.35 | FuriousGeorge | lokkju_wrk: belated thanks |
| 07:03.14 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
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| 07:08.12 | gfraysse | <PROTECTED> |
| 07:10.25 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-73-100.dsl.peoril.sbcglobal.net) |
| 07:10.42 | MACscr | why would a trunk show a status of unmonitored |
| 07:10.50 | ta^3 | yug, my problem with dialplan is a bug/feature. Could be a bug, could be a feature. Deserves a note/tip at least at voip-info wiki. |
| 07:10.50 | MACscr | and how can i get it so that it is monitored |
| 07:11.06 | MACscr | basically im trying to check to see if its registered |
| 07:13.26 | MACscr | my bad, figured it out |
| 07:17.50 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
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| 07:26.18 | gfraysse | <PROTECTED> |
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| 07:51.52 | kaldemar | hello. does anyone know an easy and quick to setup solution for postpaid billing? the amount of different billing solutions is huge and they all seem to have and endless amount of functionality that i don't need nor want. |
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| 08:44.49 | Turt|e | i had some trouble using MOH, i run openbsd and did a checkout on the astrisk svn yesterday. I dont run the zaptel timer and i didnt do that eighter on 1.4.1 and i worked at that time. Its mp3 files and i have the mpg123 installed(and it is the mpg123 not mpg321). In the log i see: res_musiconhold.c: Request to schedule in the past?!?!, is this an know issue, or any better is there an know way to solve it? |
| 08:57.44 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 08:58.15 | *** join/#asterisk kremo (n=kremo@213.215.118.28) |
| 08:58.33 | kremo | hi |
| 08:58.51 | kremo | can me anybody advise with problem of sounund quality in Astribank? |
| 09:05.45 | *** join/#asterisk Hmmhesays (n=Neg@24-117-131-41.cpe.cableone.net) |
| 09:06.39 | af_ | how could I check iaxtel is working? |
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| 09:16.50 | *** mode/#asterisk [+o denon] by ChanServ |
| 09:19.25 | kremo | is here any Xorcom user? |
| 09:19.42 | tzafrir | Does a xorcom developer count |
| 09:19.43 | tzafrir | ? |
| 09:20.18 | tzafrir | kremo, what zaptel version do you use? |
| 09:20.44 | zoa | hey i didnt know you worked on xorcom |
| 09:21.32 | zoa | :) |
| 09:23.23 | tzafrir | kremo, here? |
| 09:24.07 | tzafrir | Turt|e, mp3 files for MOH? |
| 09:24.08 | kremo | tzafrir: yes sorry I was out |
| 09:24.13 | kremo | Zaptel 1.2.14 |
| 09:30.27 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
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| 09:41.14 | *** join/#asterisk astersip (i=53f08b07@gateway/web/cgi-irc/ircatwork.com/x-3a77893b1d2df718) |
| 09:41.18 | astersip | can anyone help me on this msg "-- Extension '0' in context 'from-internal' from '232XXXXXX' does not exist. Rejecting call on channel 0/1, span 1" |
| 09:41.49 | *** join/#asterisk nfi|ermes (n=ermes@217.220.121.62) |
| 09:41.50 | Mahmoud | an extention made of one digit is not allowed as i know |
| 09:41.53 | tzafrir | kremo, which distro and which kernel do you use? |
| 09:42.10 | astersip | this is a call from outside |
| 09:42.13 | MACscr | is xorcom just an installer or a gui as well? |
| 09:42.21 | Mahmoud | astersip, check extentions.conf ? |
| 09:42.25 | astersip | i have one PRI E1 and digium TE110P |
| 09:42.52 | Mahmoud | astersip, where are you trying to call from? |
| 09:42.54 | tzafrir | MACscr, FreePBX is included there, generally |
| 09:43.02 | astersip | sorry the message is: "-- Extension '0' in context 'from-zaptel' from '232XXXXXX' does not exist. Rejecting call on channel 0/1, span 1" |
| 09:43.07 | kremo | tzafrir: I am using Xorcom TS1 with astribank 16 |
| 09:43.23 | kremo | tzafrir: I have asterisk 1.0.11.1 |
| 09:43.32 | Mahmoud | astersip, do you have that extention under [from-zaptel] context? |
| 09:43.39 | tzafrir | the version of Zaptel is the one that counts, actually |
| 09:43.39 | astersip | the 0 is the last number of my asterisk box |
| 09:44.13 | MACscr | tzafrir : so not much different than asterisk and freepbx together? reason i ask is that asterisk is giving me fits, i dont like trixbox, and openpbx doesnt have a gui yet |
| 09:44.40 | astersip | Mahmoud: can you explain what you mean ? |
| 09:45.05 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 09:45.06 | Mahmoud | astersip exten => 0,1,whatever_app() |
| 09:45.47 | Mahmoud | astersip, pastebin.ca your extentions.conf file |
| 09:45.55 | tzafrir | MACscr, it's generally better for those who like a more debian-like packages-based stuff without tons of stuff in /usr/local |
| 09:46.41 | MACscr | tzafrir: I havent even messed with debian yet, used RH and Centos up to this point, so i have no clue if i would like it |
| 09:49.02 | astersip | Mahmoud: http://pastebin.ca/404871 |
| 09:51.03 | astersip | Mahmoud: the awnser that i got when i'm calling my asterisk box is that all circuits are busy |
| 09:51.15 | Mahmoud | astersip, yeah.. |
| 09:51.20 | Mahmoud | astersip, use more than one digit |
| 09:51.57 | astersip | Mahmoud: sorry...i'm a newby :P what do you mean ? |
| 09:52.03 | Mahmoud | astersip, currently, you are dialing "0" and expecting that one of your internal phones will ring.. change the "0" to something like "00" |
| 09:52.33 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:53.18 | astersip | Mahmound: i only get one digit from my operator :( .... 0 is the first of my 30 numbers |
| 09:53.18 | astersip | for example if i call to XXXXXXX29 will work ? |
| 09:53.42 | Mahmoud | _X. doesn't match with one digit |
| 09:55.12 | astersip | humm ok i see your point |
| 09:55.22 | Mahmoud | hmmm, i think it's asterisk limitation |
| 09:55.35 | Mahmoud | you can't have one digit even with _X |
| 09:56.31 | *** join/#asterisk aaronr (n=arussell@87.127.234.100) |
| 09:57.21 | Gido-E | one digit is not ok, according to asterisk manuals etc... |
| 09:57.28 | Mahmoud | yeah |
| 09:57.36 | astersip | Mahmound so i got to talk width my operator ? |
| 09:57.40 | *** join/#asterisk mquin (n=mike@pdpc/supporter/active/mquin) |
| 09:57.47 | astersip | Mahmound: so i got to talk width my operator ? |
| 09:57.58 | astersip | .. |
| 09:58.00 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 09:58.07 | Mahmoud | i'm not sure how your network is setup |
| 09:58.31 | Mahmoud | your operator is dialing you by one digit? |
| 09:58.46 | *** join/#asterisk hijacked (i=MGXX@cerebus.clandestineresearch.com) |
| 09:59.31 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:01.26 | Mahmoud | astersip, you can do it |
| 10:01.43 | Mahmoud | astersip, exten => 0,1,foo_app() |
| 10:02.46 | astersip | ... |
| 10:02.47 | astersip | . |
| 10:02.51 | kremo | 2 |
| 10:02.54 | kremo | sorry |
| 10:03.17 | Mahmoud | astersip, currently you are using _X. wild card to match.. if you add one extra exten leterally saying "0" it works |
| 10:04.22 | Mahmoud | exten => 0,1,answer() |
| 10:04.28 | Mahmoud | exten => 0,n,echo() |
| 10:04.34 | Mahmoud | add them to your file |
| 10:05.18 | Mahmoud | then asterisk -rx "reload" |
| 10:05.39 | astersip | but it isn't allways the "0" :( |
| 10:05.53 | astersip | it depends on the number i'm dialing |
| 10:06.08 | astersip | i have 30 numbers on the E1 |
| 10:06.32 | Mahmoud | how can analog phones know if they want to dial "0" or "000" ? |
| 10:06.33 | astersip | if i dial the first i get "0" if i dial the second i get "1" and so on |
| 10:06.54 | Mahmoud | analog telephony sends numbers, and whenever a match is made it thinks that's it |
| 10:07.01 | *** join/#asterisk ThoMe (n=tm@tm.muc.de) |
| 10:07.02 | ThoMe | hihio |
| 10:07.07 | ThoMe | said: what is: asdf |
| 10:07.10 | Mahmoud | try dialing 911 and 911222 it will be the same |
| 10:07.19 | ThoMe | exten => h,n,System(/usr/local/scripts/processfax.sh ${FAXFILE} asdf "${CALLERIDNUM} ${CALLERIDNAME}") << example |
| 10:07.35 | astersip | the number is someting like this "232428700" to "232428730" |
| 10:08.17 | Mahmoud | astersip, so the 1st person is "00" ? |
| 10:10.30 | Gido-E | astersip: ${EXTEN:-1} |
| 10:12.36 | ThoMe | how i can recieve isdn-faxes with asterisk? 64kbit? |
| 10:13.03 | ThoMe | impossible? |
| 10:14.01 | astersip | Mahmoud: i just receive "0" |
| 10:14.37 | astersip | Mahmoud: and the strange thing is that "232428710" i receive "0" again |
| 10:15.15 | Mahmoud | talk to them? |
| 10:16.25 | astersip | Mahmoud: tnkx :) now i have some argoments to talk with them :) (sorry my ingles) |
| 10:16.40 | Mahmoud | np |
| 10:17.30 | astersip | Gido-E:"${EXTEN:-1}" didn't work |
| 10:17.36 | astersip | i got the same thing |
| 10:18.43 | astersip | i put "exten => _X.,1,Set(DID=${EXTEN-1})" |
| 10:18.52 | astersip | and got only "0" |
| 10:18.54 | MrWup | anyone good here with debian? noone in the debian channels is alive |
| 10:19.04 | MrWup | im doing the sarge download and install. ive got to the point of downloading, and lots of sites ive tried it says: gzip: stdin: invalid compressed data format violated failed fetch... packages.gz- sub-process gzip returned error code (1) W: couldn't stat source package |
| 10:19.14 | MrWup | the one i went for is http://debian.virginmedia.com/dists/stable/main/binary-i386/Packages.gz |
| 10:19.20 | MrWup | installing linux26 |
| 10:20.31 | Gido-E | <PROTECTED> |
| 10:20.41 | Gido-E | ${EXTEN-1})" |
| 10:21.35 | *** join/#asterisk key2 (n=key2@81.52.138.22) |
| 10:22.55 | *** join/#asterisk uwe (i=uwe@217.66.226.39) |
| 10:23.42 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 10:24.04 | uwe | hello, i have configured a queue, and when i do show queues, i get most of the extensions with line similar to: Local/1917@from-internal/n (Unknown) has taken no calls yet , and many calls are waiting |
| 10:24.30 | uwe | are those symptoms common for a knowns issue ? |
| 10:24.36 | Mahmoud | <PROTECTED> |
| 10:25.02 | Mahmoud | i mean ${EXTEN:-1} |
| 10:25.46 | *** part/#asterisk inspired (n=mikael@85.221.7.59) |
| 10:27.25 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-73-100.dsl.peoril.sbcglobal.net) |
| 10:27.49 | astersip | the main problem is that i only receive one digit |
| 10:27.54 | MACscr | Timeout, but no rule 't' in context 'numberplan-custom-1' is an error im getting |
| 10:28.00 | astersip | insted of the entire number |
| 10:28.01 | MACscr | what is a rule t? |
| 10:28.19 | Mahmoud | astersip, and you get one number even for 10.. means your operator is actually giving your 10 possible numbers (0 to 9) |
| 10:28.57 | astersip | yes that even more strange :( |
| 10:29.05 | astersip | becouse i have 30 numbers |
| 10:29.29 | astersip | sou how can i distinct from XXXXXXX00 of XXXXXXX10 |
| 10:29.35 | Mahmoud | call him and say "YOU SUCK" and go on hook |
| 10:29.46 | astersip | in both i receiv "0" |
| 10:30.05 | Mahmoud | what about xxxxxx11, do you get 1? |
| 10:30.35 | astersip | yup :P |
| 10:31.08 | astersip | XXXXXXX01 got 1 to |
| 10:31.08 | Mahmoud | what's his number? i'll call him (j/k) |
| 10:31.12 | astersip | lol |
| 10:32.50 | ThoMe | how i can recieve isdn-faxes with asterisk? 64kbit? |
| 10:32.52 | ThoMe | impossible? |
| 10:33.46 | Gido-E | ThoMe it is possible. |
| 10:34.36 | ThoMe | Gido-E: oh. how? |
| 10:37.38 | *** join/#asterisk pnlarsson (n=pnlarsso@c83-248-12-187.bredband.comhem.se) |
| 10:38.20 | *** join/#asterisk Splat (n=splat@eth112.tas.adsl.internode.on.net) |
| 10:39.15 | astersip | anyone have a nuclear bomb to put on my provider ?! |
| 10:39.22 | ThoMe | Gido-E: huhu? |
| 10:39.33 | Gido-E | astersip nope, only cluster bombs. |
| 10:40.11 | ThoMe | Gido-E: how i can find a doku? |
| 10:40.56 | Gido-E | ThoMe you can find documents with www.google.com |
| 10:41.14 | astersip | Gido-E: ok it will work :) |
| 10:41.26 | Gido-E | ok! :-) |
| 10:41.26 | ThoMe | Gido-E: grossschnautze. |
| 10:41.35 | astersip | they saying that is all ok :P |
| 10:44.13 | Gido-E | ThoMe i am not answering al your recursive questions. |
| 10:44.47 | ThoMe | Gido-E: jep. is ok |
| 10:45.31 | ThoMe | Gido-E: is AsterFax posible for resieve faxes with 54 kbit? |
| 10:45.34 | ThoMe | 64 |
| 10:46.15 | *** join/#asterisk giasai68 (n=giasai@ip-240-130.sn2.eutelia.it) |
| 10:47.02 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 10:47.18 | giasai68 | hello i need an information: how can i set atserisk to accept incoming call only from some ip? and not accept from all ip? |
| 10:48.34 | *** join/#asterisk TuxBender (n=Bender@ns2.be-ok.com) |
| 10:48.42 | TuxBender | welcome'! |
| 10:49.03 | Gido-E | Bend over |
| 10:49.04 | TuxBender | does asterisk support early media (rfc 3960)? |
| 10:57.31 | uwe | does anyone know what possibly could be the reason calls are in queue and not all phones are ringing ? |
| 10:57.47 | uwe | i mean some calls stay in the queue for ages |
| 10:58.20 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:01.21 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 11:05.18 | MACscr | im getting this error |
| 11:05.20 | MACscr | WARNING[2952] chan_sip.c: Unknown insecure mode '' on line 59 |
| 11:05.32 | MACscr | but im confused on what file its actually talking about |
| 11:05.43 | MACscr | sip.conf? |
| 11:06.08 | MACscr | what does chan_sip.c mean |
| 11:07.34 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 11:07.57 | uwe | MACscr: i suppose it means that the module chan_sip generated what you got |
| 11:10.22 | MACscr | hmm, im trying to figure out whats going on as im getting a 403 error when i try to call an outside number |
| 11:10.55 | MACscr | the sip trunk is registering fine and i can call asterisk vmail fine |
| 11:11.00 | MACscr | grrr |
| 11:12.28 | MACscr | this is my error log |
| 11:12.30 | MACscr | http://pastebin.ca/404935 |
| 11:13.20 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:15.27 | MACscr | any idea why it would say all my sip peers are unmonitored? |
| 11:16.59 | MACscr | looks like i can call from extension to extension as well |
| 11:17.47 | uwe | seems you have some issue with your dialpaln ...no ? |
| 11:17.54 | uwe | non existing contexts |
| 11:20.01 | MACscr | hmm, the dialplan was setup by asterisknow |
| 11:20.32 | MACscr | i havent tried anything complex, just a simple sip provider and an extension |
| 11:20.33 | MACscr | hmm |
| 11:24.54 | e-ddie | does anyone know how to make the person doing an attended transfer notified (by giving a tone or something) if the transfer falls back to him? |
| 11:26.13 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 11:27.08 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 11:27.27 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 11:27.31 | *** join/#asterisk clintong (n=clinton@89.129.75.128) |
| 11:27.42 | clintong | hi all - i need some advice. |
| 11:27.53 | clintong | what i need to achieve is the following: |
| 11:28.08 | clintong | a customer calls up an automated service |
| 11:28.24 | clintong | the service requests that they type in their identifier |
| 11:28.36 | clintong | then their password |
| 11:28.43 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 11:28.49 | clintong | if accepted, it allows them to record a message |
| 11:29.09 | clintong | that data gets converted to a wav file |
| 11:29.21 | clintong | the service will be interacting with a perl application |
| 11:29.25 | *** join/#asterisk montag___ (n=montag@nat-percro2.sssup.it) |
| 11:29.39 | MACscr | before you go any farther, whats your question |
| 11:29.42 | clintong | so - is asterisk and VoIP the way to go with this? or is there a simpler implementation |
| 11:29.48 | montag___ | my asterisk box permit to user with wrong password to place call, it' s a bug of asterisk 1.2.1 or it's normal ? |
| 11:29.57 | MACscr | there is nothing simple about it |
| 11:29.59 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 11:30.03 | clintong | :) great |
| 11:30.15 | MACscr | asterisk can do it, but your going to have to learn a lot |
| 11:30.22 | clintong | i'm sure |
| 11:30.34 | MACscr | i would charge about 5-10k to do this for a client |
| 11:31.13 | clintong | given that this is a "single function service", would it be reasonable to build a dedicated program (i am an experienced perl developer, but no very little about VoIP) |
| 11:31.14 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 11:31.20 | MACscr | not that i could do it really, just saying the amount of time that willl go into it |
| 11:31.27 | clintong | ok |
| 11:31.38 | clintong | MACscr, would you mind clearing up some concepts for me |
| 11:31.58 | clintong | SIP could be used to initiate the session (clearly ;) ) |
| 11:32.13 | MACscr | clintong : your asking the wrong person |
| 11:32.17 | clintong | but then the sound data itself, how does that fit get transferred |
| 11:32.27 | MACscr | i wouldnt call myself experienced when it comes to asterisk |
| 11:32.31 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 11:32.40 | clintong | ok thanks MACscr |
| 11:32.48 | clintong | anybody else who can give me input on this? |
| 11:32.49 | MACscr | im an IT Consultant with a telco background |
| 11:33.08 | MACscr | i would collect your thoughts and post in the forums |
| 11:33.16 | clintong | ok ta |
| 11:33.50 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 11:33.50 | *** mode/#asterisk [+o russellb] by ChanServ |
| 11:33.51 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 11:35.06 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 11:36.25 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 11:36.34 | *** join/#asterisk luisjose (n=ljd@unaffiliated/luisjose) |
| 11:37.39 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 11:38.58 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 11:40.13 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 11:41.32 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 11:42.20 | MACscr | hello russel |
| 11:42.35 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 11:42.51 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 11:43.32 | MACscr | _clarly is getting quite annoying |
| 11:44.06 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 11:45.25 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 11:46.22 | russellb | hey MACscr |
| 11:46.38 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 11:46.43 | *** mode/#asterisk [+b *!*=kroseneg@*.schmidham.net] by russellb |
| 11:47.13 | russellb | there, he won't be back here :) |
| 11:47.20 | MACscr | thanks =P |
| 11:48.01 | MACscr | russellb: since your an op and all, im guessing your pretty asterisk experienced, do you mind checking out this post for me? |
| 11:48.24 | MACscr | http://forums.digium.com/viewtopic.php?p=47083#47083 |
| 11:48.36 | mvanbaak | gheh, russellb and asterisk experience ? |
| 11:48.49 | MACscr | any insight would be appreciated =) |
| 11:51.34 | russellb | i know nothing |
| 11:52.04 | MACscr | aight, np |
| 11:52.09 | MrWup | guys i need some guidance |
| 11:52.14 | MrWup | the debian people are a bit unhelpful |
| 11:52.19 | MrWup | ive just installed debian and logged in |
| 11:52.25 | MrWup | and i need to get into the desktop |
| 11:52.33 | MrWup | i did tell the installer to download and install desktop stuff |
| 11:52.38 | MrWup | but im not sure how to start it |
| 11:53.57 | russellb | MACscr: most of that pastebin that you link to isn't relevant. it is output from a configuration reload |
| 11:54.15 | russellb | MACscr: however, that second to last line ... |
| 11:54.33 | russellb | MACscr: it says forbidden was received from your soft phone |
| 11:54.49 | montag___ | my asterisk box permit to user with wrong password to place call, it' s a bug of asterisk 1.2.1 or it's normal ? |
| 11:55.09 | MACscr | right, but why the heck is it forbidden? |
| 11:55.34 | russellb | montag___: assuming you're talking about SIP, then you need to set "allowguest=no" in sip.conf |
| 11:55.47 | russellb | MACscr: i'd have to see the "sip debug" output of the call |
| 11:56.11 | montag___ | thanks |
| 11:56.35 | JT | MACscr: do you have a line in sip.conf of insecure= with no parameters? |
| 11:56.41 | russellb | MACscr: but getting that from your softphone when your softphone is making the call doesn't make much sense |
| 11:57.56 | MACscr | http://pastebin.ca/404960 |
| 11:58.15 | MACscr | JT: i looked for that and didnt see insecure= at all |
| 11:58.20 | MACscr | not that wasnt commented out |
| 11:58.21 | BrokenNoze | Hi, I have a load of prompts in g711u format how do i get asterisk to use them rather than the default .gsm files? |
| 11:58.31 | BrokenNoze | or do i have to convert them all? |
| 11:58.38 | JT | BrokenNoze: delete the .gsm ones |
| 11:58.45 | BrokenNoze | that it? |
| 11:59.02 | JT | well it should help |
| 11:59.10 | russellb | yes, that is it |
| 11:59.18 | JT | although if you ever use gsm you might want to keep them |
| 11:59.24 | JT | but transcoding wouldn't be that bad |
| 11:59.28 | JT | and you may not use gsm |
| 11:59.45 | BrokenNoze | Um. ok. I thought I'd tried that, obviously not |
| 12:00.27 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:01.05 | MACscr | russellb : not sure if you noticed that i posted a sip debug ip xxxxx earlier |
| 12:01.15 | slima | dbinit: Unable to open Asterisk database - where is this database? |
| 12:01.44 | puzzled | hi |
| 12:02.01 | russellb | MACscr: yeah, i was just looking at it |
| 12:02.21 | russellb | slima: /var/lib/asterisk/astdb |
| 12:02.39 | russellb | MACscr: i'd like to see the debug of the other end of it, where asterisk calls your trunk |
| 12:02.47 | slima | -rw-r--r-- 1 asterisk asterisk 1024 Mar 21 12:28 /var/lib/asterisk/astdb |
| 12:03.00 | slima | looks good? |
| 12:03.15 | russellb | yeah |
| 12:03.28 | russellb | assuming you are running as "asterisk" |
| 12:03.36 | russellb | and not something else other than root |
| 12:03.40 | MACscr | russellb : what is the command for htat |
| 12:03.44 | MACscr | er, that |
| 12:03.57 | zoa | russel, is mattf in the office ? |
| 12:04.07 | russellb | MACscr: well, if you are not doing any other calls at the moment, just plain "sip set debug" will just show all SIP traffic |
| 12:04.16 | russellb | zoa: i have no idea, i'm out of town |
| 12:04.25 | russellb | zoa: and it's still a bit early for anyone to be in |
| 12:05.30 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:06.28 | slima | russellb: yes, I running asterisk as ''asterisk'' user. |
| 12:06.44 | slima | but i have db.c:66 dbinit: Unable to open Asterisk database flood |
| 12:06.45 | slima | ;) |
| 12:06.57 | MACscr | russellb : this is about all i could see left in the terminal, hope its enough |
| 12:06.59 | MACscr | http://pastebin.ca/404968 |
| 12:07.16 | zoa | oki thanks |
| 12:07.39 | BrokenNoze | Hi, no i've replaced the files but get File vm-login does not exist in any format error |
| 12:08.26 | BrokenNoze | then it says it can't open stream etc and plays nothing to the voicemail caller. just silence. |
| 12:08.40 | BrokenNoze | Have I got to rename the files or something? |
| 12:09.34 | russellb | BrokenNoze: well don't remove the ones you need :) |
| 12:09.43 | MACscr | i noticed this error: SIP/2.0 403 User does not exist |
| 12:10.00 | russellb | only remove the sounds that you are replacing |
| 12:10.00 | MACscr | im not sure why that would exist since its already registered though |
| 12:10.34 | BrokenNoze | russellb : no I have a different library of sounds ( British ) which is a full replacement |
| 12:11.04 | BrokenNoze | for the default US but the extensions are .g711u not .gsm |
| 12:11.53 | BrokenNoze | I don't need to change a config file or something so asterisk recognises the different file extension? |
| 12:12.36 | JT | .ulaw not .g711u afaik |
| 12:12.59 | JT | BrokenNoze: are you in the US? |
| 12:13.15 | BrokenNoze | JT : No, UK |
| 12:13.18 | *** join/#asterisk Zand3r (n=Zand3r@spc2-bolt7-0-0-cust301.bagu.broadband.ntl.com) |
| 12:13.31 | JT | BrokenNoze: they should have been made in alaw format not ulaw |
| 12:13.44 | BrokenNoze | JT : I have both actually |
| 12:13.57 | JT | use alaw |
| 12:14.40 | BrokenNoze | the extensions for those are .g711a and they don't work either. do i need to change the extension to .alaw? |
| 12:15.14 | Zand3r | Hi all... I've been configuring some POlycom phones for use with asterisk. When a call comes in the extension is identified on the phone's screen as the CallerID. When there isno CallerID information however the phone displays the extension as "asterisk". I can't see where this is set. Can the extesion that asterisk identifies itself as be changed from the word "asterisk" to something else? |
| 12:15.29 | *** join/#asterisk martineyles_ (n=martiney@adsl-w-234.as15758.net) |
| 12:15.33 | martineyles_ | Hi |
| 12:15.43 | martineyles_ | Quick query about agents |
| 12:15.53 | JT | BrokenNoze: yes |
| 12:16.22 | martineyles_ | Is it possible to log in agents automatically at 8:30 every weekday, and out at 5pm |
| 12:16.43 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
| 12:17.28 | russellb | MACscr: i'm sorry, but i am going back to bed now. i just woke up and couldn't sleep so stopped in for a few minutes. good luck ... |
| 12:17.50 | martineyles_ | and then can the queue have voicemail if no agents are logged in |
| 12:19.20 | MACscr | JT: i know i was a jerk to you yesterday, but do you mind looking at my debug data? |
| 12:22.36 | BrokenNoze | JT : Doh, thanks. know this is off topic but is there a quick way to change all the file extensions from .g711a to .alaw |
| 12:23.51 | Strom_M | good morning |
| 12:24.10 | Strom_M | BrokenNoze: something along the lines of: |
| 12:24.51 | Strom_M | for i in *; do mv $i ${i%%g711a}.alaw; done |
| 12:24.58 | JT | exactly ;) |
| 12:25.01 | Strom_M | i dont remember my bash substring syntax though |
| 12:25.10 | Strom_M | so you may want to verify that before running it |
| 12:25.27 | BrokenNoze | Strom_M : Thanks alot |
| 12:26.02 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:27.49 | *** join/#asterisk punkgode (n=Punkgode@rev-200-40-119-222.netgate.com.uy) |
| 12:28.26 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:28.43 | Strom_M | oh hey, that is the right syntax |
| 12:29.57 | Strom_M | woot...not bad for (a) not having used it for two years, and (b) having just woken up |
| 12:30.33 | creativx | how about... (C) |
| 12:31.06 | Strom_M | (c) not really being a bash scripter anyway? :) |
| 12:31.24 | blitzrage | (d) not getting yourself into situations where you need to script |
| 12:31.41 | blitzrage | s/script/strip/g |
| 12:31.50 | blitzrage | hehe |
| 12:32.50 | *** part/#asterisk martineyles_ (n=martiney@adsl-w-234.as15758.net) |
| 12:33.28 | zoa | leif |
| 12:33.30 | zoa | darling |
| 12:33.32 | zoa | did you try it ? |
| 12:33.44 | zoa | btw, i saw you today in a neighbouring office |
| 12:33.46 | zoa | very strange |
| 12:33.50 | zoa | i was certain it was you |
| 12:33.51 | blitzrage | zoa: not yet... but I just woke up, so maybe I'll play with it now! |
| 12:33.54 | zoa | even after a second look |
| 12:34.00 | blitzrage | zoa: really.... that's neat :) |
| 12:34.13 | zoa | if i see the guy again, and i still think he looks like you i will take a picture |
| 12:34.37 | blitzrage | great! |
| 12:34.45 | blitzrage | that'd be neat |
| 12:34.54 | Corydon76-home | Did he have pretty pretty blue eyes? |
| 12:35.11 | zoa | nah, he looked bad, like the original :p |
| 12:35.12 | coppice | blue eyes are a PITA |
| 12:35.31 | Strom_M | are they? |
| 12:35.56 | Corydon76-home | I have blue eyes, too, but they aren't the pretty baby-blue color that Leif has |
| 12:36.02 | coppice | yep. the sun pours through that blue bit, so when the iris stops down the world goes hazy |
| 12:36.04 | zoa | coppice probably was married to a blond blue eye girl before :) |
| 12:36.14 | *** join/#asterisk marexz (n=marexz@marexz.mil.lv) |
| 12:36.30 | zoa | ah no |
| 12:36.35 | zoa | he has some scientific explanation |
| 12:36.38 | blitzrage | coppice: hrmmm... I've never really noticed that... but I'll look for it now |
| 12:36.39 | zoa | who would have thought :) |
| 12:36.41 | *** join/#asterisk step_quasar (n=step_qua@250-171-114-200.fibertel.com.ar) |
| 12:36.41 | ThoMe | guckguck. |
| 12:36.53 | ThoMe | hat hier schon mal versucht asterisk mit hylafax zu verknuepfen? |
| 12:36.54 | zoa | look into the sun with a telescope |
| 12:36.59 | zoa | just to be sure |
| 12:37.00 | coppice | zoa: my wife is married to a blonde with blue eyes |
| 12:37.03 | blitzrage | zoa: lol |
| 12:37.18 | blitzrage | coppice: you must be norwiegen? :) |
| 12:37.28 | zoa | omg, same sex marriage is allowed there? :) |
| 12:37.45 | coppice | why norwegian? |
| 12:37.52 | *** join/#asterisk friedrich| (n=friedric@e177243233.adsl.alicedsl.de) |
| 12:37.57 | zoa | steve doesnt sound like norwegian |
| 12:38.01 | blitzrage | not sure... just thinking of the Swedish bikini team that is blonde with blue eyes |
| 12:38.12 | zoa | coppice, he just knows only 3 countries in europe |
| 12:38.13 | blitzrage | I've never heard Steve talk :) |
| 12:38.22 | blitzrage | zoa: yah -- the good ones :) |
| 12:38.22 | zoa | and apparently he thinks sweden and norway is the same |
| 12:38.24 | zoa | :) |
| 12:38.28 | Strom_M | you can get that bikini team for $23 at IKEA |
| 12:38.29 | Corydon76-home | blitzrage: there are men on the Swedish bikini team? |
| 12:38.40 | blitzrage | Corydon76-home: not to my knowledge |
| 12:38.53 | friedrich| | what are 3 european countries you could know about? |
| 12:39.03 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 12:39.13 | coppice | england, scotland and wales |
| 12:39.19 | zoa | :) |
| 12:40.23 | Corydon76-home | Great, three countries that are politically Europe, but not geographically Europe. |
| 12:40.42 | blitzrage | hrmmm... this RTPAUDIOQOS variable not always showing up is .... interesting |
| 12:40.58 | coppice | don't islands count? |
| 12:41.08 | Strom_M | I was going to say Sweden, Latvia, and Luxembourg, but what do I know |
| 12:41.13 | Corydon76-home | blitzrage: it depends on who sends the BYE on whether or not it has a chance to show up |
| 12:41.32 | blitzrage | Corydon76-home: yah, that's what I've been noticing.... sometimes NULL, sometimes populated |
| 12:41.41 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 12:41.56 | blitzrage | and I agree that it seems to matter who hangs up |
| 12:42.09 | Corydon76-home | blitzrage: a host that sends the BYE has no chance whatsoever of getting that variable, since it institutes the variable just before channel destruction |
| 12:42.25 | blitzrage | seems like the code for that variable is in a bad spot.... |
| 12:42.35 | Corydon76-home | Precisely |
| 12:42.54 | blitzrage | and I'm guessing there is no other obvious place to put it :) |
| 12:43.17 | Corydon76-home | Well, the problem is that it refers to accumulated statistics |
| 12:43.37 | Corydon76-home | So it may be invalid immediately after it was retrieved |
| 12:43.51 | Corydon76-home | so the idea was to place it after the RTP stream had ended |
| 12:44.01 | Corydon76-home | which also makes it irretrievable |
| 12:44.25 | Corydon76-home | It should probably be reimplemented as a dialplan function |
| 12:44.32 | blitzrage | I'd agree with that |
| 12:44.43 | Corydon76-home | so it becomes a snapshot, without unduly burdening the process |
| 12:44.45 | blitzrage | plus then you can get the field you want, and not all of them |
| 12:44.51 | Corydon76-home | Right |
| 12:46.07 | Corydon76-home | The only question then is, do we patch the release branch or only trunk? |
| 12:46.26 | blitzrage | I'd say the release... |
| 12:46.29 | blitzrage | that's pretty much a bug |
| 12:46.53 | Corydon76-home | Okay, get russell to sign off on the concept for release branch, and I'll write it |
| 12:46.59 | blitzrage | but I guess it depends how "independent" the DP function is (i.e. what it has to touch -- what errors it could introduce, etc... |
| 12:47.23 | blitzrage | ok, I'll msg him when I see him come online in a few hours |
| 12:47.35 | blitzrage | he's on the west coast, so I don't expect to see him until 11-12am EST |
| 12:47.48 | Corydon76-home | Well, the problem is that the dialplan function needs to reside in chan_sip.c because that's where the SIP channel is... but the statistics all exist in rtp.c, in private structures |
| 12:47.56 | Corydon76-home | So it's a fairly involved change |
| 12:48.01 | blitzrage | ahhhh I see |
| 12:48.22 | blitzrage | well... either way it needs to change.... so definitely needs to go into trunk, but I guess you're looking at building it once in 1.4 then merge it forward? |
| 12:48.39 | Corydon76-home | I'll probably need to define a new public struct and a public API to fill or retrieve the struct |
| 12:49.13 | Corydon76-home | blitzrage: right |
| 12:49.45 | blitzrage | zoa: ping |
| 12:50.01 | zoa | pang |
| 12:50.29 | blitzrage | libssl.so.0.9.7: cannot open shared object file <-- FC6, only have 0.9.8b it looks like |
| 12:50.31 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 12:51.18 | blitzrage | Corydon76-home: I guess the DP function would have the stats for both channels then right? tx would be originator, and rx would be terminator? |
| 12:51.33 | blitzrage | i.e. 100 (tx) calls 102 (rx) |
| 12:51.49 | Corydon76-home | blitzrage: I'll check later |
| 12:51.56 | Corydon76-home | Shower time |
| 12:51.58 | blitzrage | Corydon76-home: coolio foolio |
| 12:52.00 | blitzrage | have fun |
| 12:52.25 | MACscr | do i have to restart asterisk or anything like that after making a change to a conf? |
| 12:52.47 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:52.48 | blitzrage | MACscr: depends on the .conf file... more than likely you have to reload the module |
| 12:52.54 | MACscr | users? |
| 12:53.00 | MACscr | whats the process for that |
| 12:53.12 | blitzrage | i.e. sip.conf requires 'module reload chan_sip.so' |
| 12:53.24 | blitzrage | or you can just reload everythign with 'reload' |
| 12:55.02 | *** join/#asterisk Fieldy (i=ZjL9WGgl@gentoo/contributor/Fieldy) |
| 12:55.49 | blitzrage | crappy.... 416-420 is a Bell Canada exchange and not Rogers AT&T |
| 12:55.53 | blitzrage | that'd have been sweet |
| 12:57.38 | blitzrage | hrmmm... 416-616 is kinda neat |
| 12:58.08 | blitzrage | oh they have 416-666! nice |
| 12:58.12 | ThoMe | what is ttyIAX in conjunction with hylafax and asterisk? |
| 12:58.33 | ThoMe | i have misdn. can i altrough use iax? |
| 13:00.05 | MACscr | sweet, finally got asterisknow to make an outgoing call |
| 13:01.34 | MACscr | now i can finally go to bed =P |
| 13:01.42 | blitzrage | I know how that is :) |
| 13:02.01 | *** join/#asterisk deeperror (n=deeperro@mail.banctel.com) |
| 13:02.23 | blitzrage | bet you said, "just one more thing", 3 hours ago |
| 13:05.00 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 13:05.18 | blitzrage | anyone here have contacts at Mitel that can flash 5220 phones? |
| 13:05.44 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 13:07.30 | *** join/#asterisk surfless (n=craig@dsl-37-83.nsw1.net.au) |
| 13:10.26 | *** join/#asterisk MarkWD (n=MarkWD@rrcs-67-78-88-186.sw.biz.rr.com) |
| 13:12.30 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:14.22 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:21.47 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 13:21.47 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:22.54 | *** join/#asterisk ppyy (i=ppyy@218.93.153.36) |
| 13:24.49 | *** join/#asterisk Corydon76-home (i=pink@pdpc/supporter/sustaining/Corydon76-home) |
| 13:24.49 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 13:30.36 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 13:32.35 | *** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it) |
| 13:37.56 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-80b0fc79aef14f4a) |
| 13:38.51 | *** join/#asterisk earthsound (n=one_down@138.26.117.12) |
| 13:39.18 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-8227ae7cc4b3e42f) |
| 13:41.39 | blitzrage | zoa: p i n g :) |
| 13:42.41 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 13:44.51 | zoa | pong |
| 13:45.07 | blitzrage | did you see my libssl error before? |
| 13:45.22 | zoa | who what where ? |
| 13:45.34 | zoa | aha |
| 13:45.35 | zoa | see it now |
| 13:45.57 | blitzrage | nice |
| 13:46.06 | zoa | will check |
| 13:47.21 | *** join/#asterisk lbow (n=lbow@dsl-146-6-62.telkomadsl.co.za) |
| 13:47.30 | blitzrage | merci |
| 13:48.09 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 13:48.34 | lbow | did anybody ever see this: ztdummy doesn't give timing for first 5 minutes after startup and then starts working like nothing was ever wrong |
| 13:49.12 | lbow | intel 945 chipset, Gigabyte 945GZM-S2 mobo, 2.6.17 kernel |
| 13:50.19 | ThoMe | zoa: hiho |
| 13:50.29 | RoyK | spam, spam, spam, egg, sausage and spam? http://video.google.com/videoplay?docid=5627694446211716271&q=monty+python+spam |
| 13:50.39 | ThoMe | zoa: how i can use hylafax, iaxmodem and asterisk with ISDN-speed 64kbit ? |
| 13:51.49 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
| 13:51.57 | zeeesh | hi |
| 13:52.21 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 13:52.31 | JoNate | hey guys...how can I check which codecs my asterisk is able to use? |
| 13:53.18 | deeperror | read the manual? |
| 13:53.32 | angryuser | can somebody tell me what is happening on my misdn ports ? (asterisk 1.4 debian) using B410P |
| 13:53.34 | angryuser | http://www.pastebin.ca/405038 |
| 13:55.05 | lbow | angryuser: what are you connecting to? why are you in NT mode? |
| 13:56.27 | angryuser | lbow: i am connecting it directly from isdn provider |
| 13:56.35 | uwe | hello ... im using asterisk 1.2.16, calls are staying in queue, although not all agents are busy, strategy is set to ringall, what could the issue be? |
| 13:58.05 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 13:58.05 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.2 (Mar. 19, 2007), Asterisk 1.2.17 (Mar. 19, 2007), Zaptel 1.2.16 (Mar. 19, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. |
| 13:58.38 | coppice | I've never heard of a real life fax machine with an ISDN plug on it |
| 13:58.50 | lbow | nt = network termination. te = terminal equipment. |
| 13:59.19 | csplinter | Is asterisk compatible with norstar's digital phones |
| 13:59.26 | angryuser | lbow: ok thx, i received a wrong answer yesterday then |
| 13:59.33 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 13:59.53 | *** join/#asterisk bkw_ (i=brian@70.143.50.36) |
| 14:00.15 | *** join/#asterisk AlienPenguin (n=Miranda@ip-145-151.sn2.eutelia.it) |
| 14:00.16 | coppice | and never having heard of a real G4 machine, I've never done anything to support one :-) |
| 14:00.23 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:00.33 | csplinter | sorry i see this is the wrong channel |
| 14:01.21 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 14:02.00 | *** join/#asterisk Tili (n=tili@cm109.gamma248.maxonline.com.sg) |
| 14:03.42 | AlienPenguin | Hi, in * 1.2.x i had a context in my dialplan that i included in every other context and that would simply Goto(s,1) for extensions 'i' and 't'. In * 1.4 it does not work and it complains about UNKNOWN STATE. Any suggestions on how to accomplish this in 1.4? |
| 14:06.31 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:08.42 | [TK]D-Fender | AlienPenguin: Pastebin your dialplan and the CLI output of a failed call. |
| 14:08.44 | [TK]D-Fender | ~pb |
| 14:08.45 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 14:11.20 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:11.20 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:11.31 | AlienPenguin | http://pastebin.ca/405062 |
| 14:11.57 | *** part/#asterisk earthsound (n=one_down@138.26.117.12) |
| 14:14.00 | giasai68 | hello i need an information: how can i set atserisk to accept incoming call only from some ip? and not accept from all ip? |
| 14:15.12 | hijacked | firewall rule? |
| 14:17.03 | *** join/#asterisk SplasPood (i=jwb@jwb.sh) |
| 14:17.41 | [TK]D-Fender | AlienPenguin: == Auto fallthrough, channel 'SIP/102-081cf730' status is 'UNKNOWN' |
| 14:18.04 | [TK]D-Fender | AlienPenguin: I'm betting your did not set "autofallthrough=no under [globals] |
| 14:18.40 | AlienPenguin | err... no, but i am not sure i did either in 1.2. was the default setting different? |
| 14:18.44 | [TK]D-Fender | errrrr [general] |
| 14:19.04 | [TK]D-Fender | AlienPenguin: Itshanging up immediately isn't it? |
| 14:19.11 | AlienPenguin | yes it is |
| 14:19.22 | [TK]D-Fender | AlienPenguin: Then add the line I mentioned to [general] |
| 14:19.56 | AlienPenguin | testing it right now... |
| 14:20.00 | [TK]D-Fender | AlienPenguin: 1.2's default behaviour is to just fall-through. They extect us to use the "WaitExten" app usually now. |
| 14:20.56 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 14:22.26 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 14:24.01 | AlienPenguin | [TK]D-Fender: thanks :) it works now :) |
| 14:24.28 | luisjose | whats the way to recognize the busy tone on asterisk to make a redial macro? |
| 14:27.53 | blitzrage | luisjose: Goto(s-${DIALSTATUS},1) |
| 14:28.04 | blitzrage | s-BUSY,1,Macro(redial) |
| 14:28.35 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 14:29.15 | [TK]D-Fender | blitzrage: recursioncrash=true ;) |
| 14:30.47 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 14:31.19 | blitzrage | [TK]D-Fender: ! ! ! |
| 14:31.45 | [TK]D-Fender | blitzrage: I don't want to be at work.... |
| 14:31.57 | blitzrage | I just want... |
| 14:32.40 | [TK]D-Fender | blitzrage: ! ! ! |
| 14:32.47 | blitzrage | w00t! |
| 14:34.52 | *** join/#asterisk ars247 (n=no@64-142-43-180.dsl.static.sonic.net) |
| 14:34.53 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 14:35.21 | luisjose | blitzrage, let me check, ty. |
| 14:35.27 | anonymouz666 | Strom_M |
| 14:39.31 | [TK]D-Fender | Guess I should get ready to adapt to 1.4 series soon.... |
| 14:39.47 | creativx | brattttttislava |
| 14:41.08 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 14:42.38 | *** join/#asterisk Ac1dcrawl (n=cow@64.31.169.118) |
| 14:42.51 | Ac1dcrawl | Is there anyone familiar with ss7 on? |
| 14:44.50 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 14:44.51 | *** join/#asterisk Mercestes (n=Merceste@cpe-24-175-82-3.houston.res.rr.com) |
| 14:46.20 | blitzrage | [TK]D-Fender: 1.4 is niiiiice |
| 14:47.19 | [TK]D-Fender | blitzrage: Have SIP issues been worked out? Any other know reason to avoid? |
| 14:47.39 | [TK]D-Fender | blitzrage: Specific points, and pertinent of course. |
| 14:47.48 | blitzrage | [TK]D-Fender: I've been using it in my development environment for over a month now with no issues |
| 14:48.05 | [TK]D-Fender | blitzrage: With the the SLA out I'm tempted... |
| 14:48.19 | blitzrage | I also use SER as the registration point, so my setup is non-standard, but it works great for me |
| 14:48.40 | blitzrage | I found a bunch of bugs and segfaults over the last 2-3 months, but they've all been fixed now |
| 14:48.59 | blitzrage | pretty rock solid now using func_odbc, odbc_cdr, SER, etc... |
| 14:49.34 | deeperror | anyone know of a way to separate channel banks from a server? |
| 14:49.59 | blitzrage | deeperror: huh? |
| 14:50.09 | deeperror | so the banks are not directly into the server |
| 14:50.23 | deeperror | if we run a load balanced asterisk setup or distributed setup |
| 14:50.27 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 14:50.42 | deeperror | the banks can't plug directly into the server |
| 14:50.43 | blitzrage | I think they have to be connected to *something*... or else where do they send calls / get logic from? |
| 14:50.49 | blitzrage | what would they connect to? |
| 14:50.53 | deeperror | not sure |
| 14:50.55 | deeperror | thats the question |
| 14:50.59 | blitzrage | I don't think it makes any sense |
| 14:51.02 | deeperror | if they go into the server how do you load balance? |
| 14:51.09 | deeperror | if the server goes down |
| 14:51.20 | deeperror | i lose all 4 banks plugged into that box |
| 14:51.26 | blitzrage | yep... it's hardware |
| 14:51.50 | blitzrage | unless there is some sort of weird load balancer box you can plug it into... but I'm not aware of such a device |
| 14:52.02 | deeperror | no one seems to have heard of anything |
| 14:52.11 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:52.13 | blitzrage | yah -- probably doesn't exist |
| 14:52.15 | deeperror | were moving to setup 200 agents |
| 14:52.21 | deeperror | analog phone4s |
| 14:52.32 | blitzrage | how would you do it on a traditional PBX? |
| 14:53.11 | blitzrage | (I think the answer is -- you don't) |
| 14:53.11 | blitzrage | advantage of SIP phones -- they have a backup proxy field |
| 14:53.11 | deeperror | so we just run 2 boxes each with 4 banks |
| 14:53.11 | blitzrage | pretty much yah |
| 14:53.22 | tzafrir | deeperror, what type of channel banks? e.g: if they are SIP: they connect via IP |
| 14:53.34 | deeperror | no rhino 24 fxs |
| 14:53.39 | deeperror | we have all PSTN now |
| 14:53.42 | blitzrage | someone might come along and correct me.... but I don't see how you can not plug it into something |
| 14:53.44 | deeperror | 200 analog phones and station |
| 14:54.02 | deeperror | i know it would plug into something haha |
| 14:54.31 | deeperror | but it seems like having them directly into the asterisk system isn't a very good solution for distributed setup or larger setups |
| 14:54.48 | *** join/#asterisk codestr0m (n=asura@ns1.netsyncro.com) |
| 14:55.16 | deeperror | would each * just manage its own group of banks and send calls for internal xfers to 100@pbx0 |
| 14:55.21 | deeperror | 200@pbx1 |
| 14:55.25 | deeperror | something like that? |
| 14:58.28 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 14:59.30 | giasai68 | hello i need an information: how can i set atserisk to accept incoming call only from some ip? and not accept from all ip? |
| 15:00.14 | gambolputty | examine the SIP From header |
| 15:01.16 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:01.28 | mquin | giasai68: sledgehammer option - set your firewall to only accept inbound connections from the address in question |
| 15:01.30 | [TK]D-Fender | deeperror: In channel-bank scenarios, yes if the connected server goes down you lose those extensions. If you want something more survivable, use SIP gateways like MediaTrix or AudioCodes which can have a secondary proxy, and run SER. |
| 15:01.43 | mquin | there's probably a smarter way to do it within *, though |
| 15:02.47 | deeperror | fender: just the nature of the analog best eh? |
| 15:03.06 | deeperror | best = beast |
| 15:03.51 | *** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn) |
| 15:04.44 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 15:04.54 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 15:05.29 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 15:05.29 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:05.45 | Katty | Rumors of my death have been greatly exaggerated |
| 15:05.53 | zoa | hey katty! |
| 15:05.57 | zoa | high five! |
| 15:06.02 | zoa | its alive |
| 15:07.52 | file | Kattttty |
| 15:08.54 | blitzrage | Katty: you're alive! |
| 15:10.20 | Katty | i am!! |
| 15:10.36 | blitzrage | w00t |
| 15:10.42 | Katty | i've also managed to get a headache while trying to figure stuff out this morning |
| 15:10.46 | Katty | i feel dumb. |
| 15:10.46 | Katty | i need help. |
| 15:10.50 | Katty | from smart peoples. *nodnod* |
| 15:11.03 | Katty | careful. i think it's gonna splode. |
| 15:11.16 | blitzrage | don't worry... I always wear this bomb suit |
| 15:11.32 | Katty | my boss gave me a nearly impossible task. |
| 15:11.37 | blitzrage | sounds like my job |
| 15:11.44 | blitzrage | you make me next Vonage! |
| 15:11.47 | Katty | but! right now i'm only doing 'phase one' he calls it. |
| 15:11.55 | Katty | it's...mostly accomplished. |
| 15:12.01 | Katty | flash operator panel is working (woo!) |
| 15:12.02 | blitzrage | wuz da issue |
| 15:12.02 | MrWup | how do u recongifure the mirror that apt-get is using? |
| 15:12.16 | Katty | welllll.......i think the issue is me ;) |
| 15:12.17 | blitzrage | MrWup: sounds like a #debian issue... ? |
| 15:12.23 | Katty | i've been looking over http://www.voip-info.org/wiki/index.php?page=Asterisk+call+notification |
| 15:12.25 | *** join/#asterisk shinux__ (n=shinux@62.128.161.160) |
| 15:12.30 | Katty | and i've made a lil popup window thingy. |
| 15:12.37 | Katty | but i need another variable. |
| 15:12.47 | max_______ | MrWup: /etc/apt/sources.list |
| 15:12.59 | Katty | like a "line 1" |
| 15:13.14 | Katty | something like ${linenumber} |
| 15:13.28 | tclark | hehe that looks interesting http://sandman.com/echo.html LINE IMPEDANCE MATCHER |
| 15:13.42 | Katty | incoming call from $callerid on $linenumber @ $datetime |
| 15:13.48 | Mercestes | Katty!!! Zomg! *tacklehugs* I missed you |
| 15:13.54 | Katty | Mercestes: !! |
| 15:14.02 | codestr0m | what's a common program to use to play .gsm files on linux... mplayer isn't playing well |
| 15:14.03 | Mercestes | :D |
| 15:14.23 | macTijn | codestr0m: convert them with sox, play with mplayer |
| 15:14.31 | anonymouz666 | I am reading two data with read and storing in arg1 and arg2... I did a noop and the value is the same... but this does not work: exten => s,n,GotoIf($["${ARG1}" = "${ARG2}"]?s,allow) |
| 15:14.36 | Mercestes | Katty: Is it zap that you rmonitoring in the popup? |
| 15:14.38 | macTijn | sox <blah.gms> <blah.wav> |
| 15:14.42 | Katty | Mercestes: yesyes. |
| 15:14.53 | anonymouz666 | it does not jump to label allow |
| 15:14.57 | codestr0m | macTijn: that's one way, but not exactly what I had in mind.. |
| 15:15.16 | macTijn | codestr0m: or play them with asterisk, and call to them ;) |
| 15:15.16 | anonymouz666 | any idea? |
| 15:16.03 | codestr0m | macTijn: who said I would actually install asterisk anywhere ;) |
| 15:16.21 | blitzrage | anonymouz666: does the label actually exist? I've typo'd creating labels a few times |
| 15:16.28 | Mercestes | Katty: Can't you parse out the "channel" variable to get the Group-Channel number? |
| 15:16.42 | macTijn | codestr0m: why are you playing with .gsm files then ? :) |
| 15:16.57 | anonymouz666 | exten => s,n(allow),Set(${ARG4}=.... |
| 15:16.58 | codestr0m | macTijn: It's a recent hobby :P |
| 15:17.04 | Katty | hmm. channel variable. |
| 15:17.05 | anonymouz666 | its there |
| 15:17.12 | Mercestes | Katty: 1-1, 1-2 1-23 etc. |
| 15:17.12 | Katty | is channel variable something like Zap/3 |
| 15:17.30 | Mercestes | Katty, Zap/3 is Zap protocol, group 3. But 3 has 1-23 behind it if you watch yoru CLI |
| 15:17.36 | Katty | hrmm. |
| 15:17.44 | Mercestes | Katty: So you will have call going out on 3-14 or something. |
| 15:17.53 | Katty | oh! |
| 15:17.54 | Mercestes | Katty: I *think* that' show it works. |
| 15:18.05 | anonymouz666 | blitzrage: the args should be between " " ? |
| 15:18.16 | *** join/#asterisk astersip (i=53f08b07@gateway/web/cgi-irc/ircatwork.com/x-08e7e702df78168e) |
| 15:18.36 | *** part/#asterisk codestr0m (n=asura@ns1.netsyncro.com) |
| 15:18.40 | blitzrage | anonymouz666: doesn't really matter |
| 15:18.51 | astersip | Mahmoud: my problem is solved |
| 15:19.00 | blitzrage | do a Verbose(1|ARG1: ${ARG1} and ARG2: ${ARG2}) before the GotoIf() |
| 15:19.20 | Strom_M | file: I'm eating a muffin |
| 15:19.23 | Mahmoud | astersip, dumb operator ? |
| 15:19.30 | file | Strom_M: what kind? |
| 15:19.52 | Strom_M | it tastes like banana bread or pumpkin or something |
| 15:19.54 | anonymouz666 | -- Executing NoOp("SIP/6000-08863598", "**** 123 123") in new stack |
| 15:19.54 | anonymouz666 | Mar 21 12:11:40 DEBUG[14696]: pbx.c:1609 pbx_substitute_variables_helper_full: Expression result is '0' |
| 15:19.54 | anonymouz666 | <PROTECTED> |
| 15:20.06 | Strom_M | whichever Atlanta Bread makes |
| 15:20.31 | anonymouz666 | Mar 21 12:11:40 DEBUG[14696]: pbx.c:6204 pbx_builtin_gotoif: Not taking any branch |
| 15:20.44 | Katty | right. |
| 15:20.49 | Katty | so...this channel variable |
| 15:21.10 | anonymouz666 | as you can see 123 = 123 :D |
| 15:21.14 | Katty | i've been staring at extensions.conf for the last 20 minutes...and didn't see it. is it ${Channel}? |
| 15:21.26 | blitzrage | anonymouz666: what is the dialplan line for the NoOp() |
| 15:21.42 | blitzrage | you must have a typo or something somewhere |
| 15:21.46 | Mahmoud | astersip, how did you solve it? |
| 15:21.49 | anonymouz666 | exten => s,n,NoOp(**** ${ServiceNumber} ${ServiceNumber_}) |
| 15:21.51 | blitzrage | infact... just pastebin the relevant section |
| 15:22.02 | blitzrage | anonymouz666: but you're doing ${ARG1} and ${ARG2} ? |
| 15:22.15 | blitzrage | it'll be easier to debug with a pastebin |
| 15:22.42 | anonymouz666 | yes, but arg1 is actually servicenumber and arg2 servicenumber_ ... but I think my logic is wrong |
| 15:22.52 | uwe | um, when i start asterisk with the -g option, where does it write the dump ? |
| 15:23.39 | blitzrage | uwe: whatever directory you started asterisk in |
| 15:23.44 | blitzrage | core.XXXX |
| 15:23.54 | astersip | Mahmoud: the problem was from they side |
| 15:23.56 | JunK-Y | uwe: u better use safe_asterisk, which will drop ur core in /tmp |
| 15:24.01 | Katty | JunK-Y: !! |
| 15:24.02 | Mahmoud | astersip, i see |
| 15:24.15 | astersip | Mahmoud: but they insist that was my configuration |
| 15:24.15 | Katty | how do i make asterisk give me a list of available variables...like callerid and datetime |
| 15:24.17 | JunK-Y | whats up honey? |
| 15:24.23 | Katty | my hair! :P |
| 15:24.26 | tzafrir | uwe, in a file called core, unless you set the kernel to dump core in a different place (linux) |
| 15:24.28 | astersip | Mahmoud: and even send a tecnichan here |
| 15:24.31 | JunK-Y | katty: use DumpChan() |
| 15:24.32 | Katty | JunK-Y: nothing really (= |
| 15:24.35 | Katty | JunK-Y: thanks. |
| 15:25.08 | Mercestes | Katty: Let me google it. Only seen it in CDRs thus far. |
| 15:25.19 | tzafrir | uwe, you can generally ask the kernel to dump a core file with a format of your choosing. This is documented in the kernel docs in the sysctl dir, IIRC |
| 15:25.20 | JunK-Y | katty: also, ya can give a try to doc/channelvariables.txt |
| 15:25.44 | *** join/#asterisk ToyMan (n=Stuart@74-32-55-210.dsl1.mdl.ny.frontiernet.net) |
| 15:25.45 | Mercestes | Yea, it should jsut be ${channel} Katty |
| 15:25.53 | astersip | Mahmoud: he talk width 3 guys over the phone.... just the last one admited that was a problem from their side |
| 15:26.13 | tzafrir | here it is: look for core_pattern in http://lxr.linux.no/source/Documentation/sysctl/kernel.txt |
| 15:26.34 | astersip | MAKING AN ANOUNCEMENT: Portugal Telecom (a portuguese operator) suck's hehehehehe |
| 15:26.53 | Mahmoud | astersip, they wasted time :/ |
| 15:27.05 | astersip | i spend 2 days on this :( |
| 15:27.17 | Katty | Mercestes: awesome. |
| 15:27.20 | Katty | here goes nothing!! |
| 15:27.22 | Turt|e | Hi, if i have 3 mp3player extension in my dialplan there is an delay between each file i played, can this be adjusted anywhere ? |
| 15:27.36 | JunK-Y | its ${CHANNEL} |
| 15:27.48 | astersip | i'm a newbie on this .... so i was thinking that the problem was from my side |
| 15:28.10 | astersip | now even if i see my network cable disconnect i call them here |
| 15:28.13 | astersip | ;) |
| 15:28.31 | uwe | thank you , i found it |
| 15:28.38 | astersip | Mahmoud tnkx again :) |
| 15:28.45 | Mahmoud | astersip, what did i do? :P |
| 15:28.59 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 15:29.00 | Mahmoud | just told you to call them? |
| 15:29.37 | uwe | well, i have no idea what to do with the dump, to whom shall i send it? |
| 15:29.56 | uwe | its relatively small |
| 15:29.58 | uwe | 28 MB |
| 15:31.07 | Juggie | ummm... theres a doc on that |
| 15:31.09 | Juggie | ~core |
| 15:31.11 | jbot | core is, like, most importantly the memory used in many classic machines. It operates under the concept of changing polarities on an array of magnetic toroids. |
| 15:31.11 | JunK-Y | read the file backtrace.txt |
| 15:31.27 | JunK-Y | ~core |
| 15:31.29 | jbot | i heard core is most importantly the memory used in many classic machines. It operates under the concept of changing polarities on an array of magnetic toroids. |
| 15:31.29 | Juggie | there you go, do what junky says :) |
| 15:32.14 | Juggie | that should be updated to something more useful. |
| 15:32.34 | Strom_M | jbot: core is also apparently a type of hat. |
| 15:32.36 | jbot | Strom_M: okay |
| 15:32.36 | uwe | hmmm |
| 15:33.15 | uwe | i dont have any backtrace.txt |
| 15:33.49 | Strom_M | no one got my stupid joke |
| 15:36.33 | *** join/#asterisk gammah (n=gammah@70-253-197-131.ded.swbell.net) |
| 15:37.40 | *** join/#asterisk [shodan] (n=shodan@ip149.99-113-216.pppoe4.joliette.intermonde.net) |
| 15:38.26 | *** join/#asterisk codestr0m (n=asura@ns1.netsyncro.com) |
| 15:38.38 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 15:39.44 | Strom_M | lol in a box |
| 15:40.44 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 15:41.59 | Katty | smbclient -M requires the messenger service to be on in windows, right? |
| 15:42.01 | Katty | anything else? |
| 15:42.10 | Katty | i'm getting connection failed issues |
| 15:45.24 | *** join/#asterisk Exhar (n=Roy@84-105-192-215.cable.quicknet.nl) |
| 15:46.40 | astersip | Mahmoud: you help me getting some bases to call them ;) |
| 15:46.47 | key2 | kaldemar ? |
| 15:46.50 | key2 | i eman |
| 15:46.51 | key2 | mean |
| 15:46.55 | Mahmoud | np |
| 15:47.04 | key2 | Katty ? |
| 15:47.13 | key2 | Katty: wasup |
| 15:47.48 | astersip | Mahmoud: and you tryed to help this poor man ;) that was around this msg widthout any anwser in the last 2 days |
| 15:48.06 | astersip | Mahmoud: i was here without beer in the last 2 days :P |
| 15:48.10 | codestr0m | I've loaded app_playback.so and codec_adpcm.so and followed the guide for wav formats here (http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk) , but I get this error ( file.c:512 ast_openstream_full: File netsyncro_com_welcomes_you_asterisk_version does not exist in any format) with * 1.2.14 .. which modules do I need |
| 15:48.17 | Mahmoud | astersip, man, stop thanking me.. next time i won't help any one.. it's getting annoying |
| 15:48.23 | *** join/#asterisk Assid (n=assid@203.212.204.107) |
| 15:49.55 | blitzrage | codestr0m: that msg typically means that file does not exist in /var/lib/asterisk/sounds/ |
| 15:52.13 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 15:52.32 | Katty | i'm dumb. i forgot that net send used the /username/ not the /computer/ name |
| 15:52.36 | *** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it) |
| 15:52.41 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:53.33 | Mercestes | I thought it used the compuer name. lol |
| 15:53.50 | Katty | so did i |
| 15:53.55 | Katty | till i started reading the syntax |
| 15:54.05 | Katty | but i still can't make a connection |
| 15:54.12 | Katty | connection to $me failed |
| 15:54.14 | Katty | *sob* |
| 15:54.20 | *** join/#asterisk shinux__ (n=shinux@62.128.161.160) |
| 15:55.20 | deeperror | http://lists.grok.org.uk/pipermail/full-disclosure/2007-March/053052.html |
| 15:55.45 | Katty | is smbclient -M the username, ip address, netbios name... |
| 15:55.52 | Katty | should be machine, if i recall correctly. |
| 15:56.36 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:56.51 | Katty | yeah, manpages say netbios name |
| 15:57.45 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 15:58.57 | *** join/#asterisk stefmtl (n=stef@stef.istop.com) |
| 15:59.02 | `Sauron | KAtTY! |
| 15:59.10 | Katty | i can't even get a list of resources :< |
| 15:59.13 | JunK-Y | katty: plan to go at cluecon this year? |
| 15:59.26 | Katty | JunK-Y: if they'll let me (= |
| 15:59.41 | Katty | JunK-Y: and a sales rep doesn't schedule me for a network setup *growl* |
| 15:59.45 | JunK-Y | they wont, they'rent crazy! :P |
| 16:00.00 | deeperror | what is that all about? |
| 16:00.26 | JunK-Y | www.cluecon.com |
| 16:01.05 | deeperror | yea i'm there ha |
| 16:01.48 | stefmtl | I have a problem with IAX2 on one of my server : calls can't go out : http://paste.uni.cc/13825 I have version 1.2.17 . 9 servers on 10 are OK with IAX2 outband calls with this version, I don't understand why... |
| 16:02.49 | Strom_M | auto-congesting call due to slow response |
| 16:02.56 | Strom_M | there be latency in that there network |
| 16:03.20 | stefmtl | Strom_M : The ping is exactly the same with one of my working server |
| 16:03.33 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 16:03.35 | Strom_M | are the host entries the same? |
| 16:03.46 | Strom_M | i notice you're trying to set a call up to 10.0.0.1 |
| 16:03.48 | *** join/#asterisk IgorG (n=FeedomPa@host-195-162-53-193.pppoe.omsknet.ru) |
| 16:03.50 | stefmtl | exactly the same |
| 16:03.56 | stefmtl | exactly |
| 16:04.25 | Strom_M | pastebin the relevant sections of iax.conf |
| 16:04.28 | [TK]D-Fender | Katty: Mew. |
| 16:04.36 | Katty | stupid bloody windows firewall |
| 16:04.37 | Katty | it works |
| 16:04.41 | Katty | windows is just screwing with me. |
| 16:04.44 | Katty | hey fender (= |
| 16:04.48 | JunK-Y | iax2 debug from 10.0.0.1 ? |
| 16:04.48 | Qwell[] | Katty: what's new? |
| 16:05.00 | Qwell[] | the windows screwing people thing, that is |
| 16:05.09 | zoa | qwell!!1 |
| 16:05.11 | zoa | Whiii |
| 16:05.15 | [TK]D-Fender | <Katty>windows is just screwing with me. <--- But did it call the next day or send flowers? |
| 16:05.16 | Qwell[] | zoa: see above ;) |
| 16:05.24 | Katty | [TK]D-Fender: no! no it didn't! |
| 16:05.25 | Strom_M | [TK]D-Fender: haha |
| 16:05.30 | Katty | not even a thank you |
| 16:05.40 | Qwell[] | Katty: Did it at least buy you dinner first? |
| 16:05.46 | Qwell[] | ...or breakfast, as it were |
| 16:05.49 | zoa | Qwell: i dont see anything |
| 16:05.57 | Qwell[] | zoa: re; windows screwing people :p |
| 16:05.58 | [TK]D-Fender | Qwell[]: Windows is often very hard to swallow.... |
| 16:06.00 | zoa | ah yes |
| 16:06.26 | Qwell[] | zoa: other than that, it looks good |
| 16:06.49 | Qwell[] | small unhandled exception (which was handled gracefully) when I changed the STUN setting, but yeah |
| 16:06.53 | Katty | vista doesn't have a messenger service :< |
| 16:07.06 | Qwell[] | Katty: windows live messenger :p |
| 16:07.24 | zoa | wow, we should not have an unhandled exception |
| 16:07.25 | Katty | hold that thought |
| 16:07.28 | zoa | can you tell me how you did that ? |
| 16:07.29 | Katty | i'm going to lunch (= |
| 16:07.38 | Qwell[] | zoa: changed the stun setting to don't use stun, and hit apply |
| 16:07.46 | Qwell[] | it continued on afterwards just fine |
| 16:08.27 | deeperror | so how much does cluecon run for the 3 days? |
| 16:08.36 | uwe | um, does anybody know if file formats played as MoH or announcements could possibly crash asterisk??? |
| 16:08.44 | zoa | but you also had static you said ? |
| 16:08.53 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 16:08.55 | Qwell[] | uwe: anythings possible.. it's highly unlikely though |
| 16:09.00 | Qwell[] | zoa: yeah, until I muted the mic |
| 16:09.09 | Qwell[] | I'm gonna blame the driver until I can prove otherwise |
| 16:09.19 | stefmtl | Junky : I don't see anything on 10.0.0.1 |
| 16:09.25 | zoa | well we should handle the driver anyways |
| 16:09.30 | zoa | did you have that on an old version too ? |
| 16:09.34 | zoa | like on idefisk 1.x ? |
| 16:09.37 | Qwell[] | didn't try |
| 16:09.41 | Qwell[] | I will tonight |
| 16:09.49 | zoa | ah i thought you used idefisk 1.x before :) |
| 16:09.54 | uwe | argh! ive been trying to get this working for a couple of months now ! i really hate when things dont go right ! damn it!! |
| 16:09.56 | JunK-Y | stefmtl: so try to find, why theres no packet rx in that server. |
| 16:10.07 | aptura | What would cause the audio on a outgoing DID to die after two seconds? |
| 16:10.13 | Qwell[] | zoa: yeah, I have, but not on that system |
| 16:10.14 | JunK-Y | are they really sent via ehereal? |
| 16:10.19 | zoa | Qwell[]: how do you like the new look / config menu ? |
| 16:10.26 | Qwell[] | looks better |
| 16:10.34 | Qwell[] | and it's nowhere near as ugly as xlite ;) |
| 16:10.45 | zoa | hehe |
| 16:10.50 | Qwell[] | zoa: You need to make it black and green, and look like a 1980s cellphone, just like xlite! |
| 16:10.56 | aptura | xlite is okay |
| 16:11.14 | zoa | Qwell[]: we are working on a skin like that too |
| 16:11.15 | Qwell[] | zoa: idefisk --ugly |
| 16:11.28 | zoa | for the people that want to have a phone |
| 16:11.29 | Qwell[] | You should make that the command line option |
| 16:11.32 | zoa | something looking like a phone |
| 16:11.50 | Qwell[] | if you do that, PLEASE make it look like a phone that isn't ugly as hell :p |
| 16:11.54 | zoa | :) |
| 16:12.31 | Qwell[] | You should also have an option to randomly drop calls.. that would be sweet |
| 16:14.15 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 16:14.27 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:14.29 | *** join/#asterisk d3bian83 (n=d3bian@217.201.170.247) |
| 16:14.32 | Strom_M | make it look like a western electric 2500 set |
| 16:14.34 | Strom_M | that would be so retro |
| 16:15.00 | d3bian83 | hi |
| 16:15.06 | d3bian83 | my name is davide |
| 16:15.11 | Strom_M | i'm sorry |
| 16:15.16 | Strom_M | but we can't help you with that problem |
| 16:15.38 | CunningPike | Strom_M: Behave! ;) |
| 16:15.42 | Strom_M | ok ok :) |
| 16:15.52 | CunningPike | Heh heh |
| 16:15.55 | Qwell[] | Strom_M: see msg :D |
| 16:15.56 | CunningPike | How's it going? |
| 16:16.00 | d3bian83 | i'm new with asterisk, i'm in trouble trying to connect two asterisk pbx between them |
| 16:16.09 | Qwell[] | in trouble with who? |
| 16:16.23 | *** join/#asterisk dasenjo (n=dasenjo@190.24.176.87) |
| 16:16.25 | d3bian83 | sorry, my english isn't well |
| 16:16.28 | d3bian83 | i'm italian |
| 16:16.32 | d3bian83 | :-) |
| 16:16.40 | CunningPike | d3bian83: Look for 'IAX trunking' on the wiki - there's a good article there |
| 16:17.12 | d3bian83 | yes, i've already read the article |
| 16:17.55 | d3bian83 | but i can't understand the switch statement |
| 16:17.59 | d3bian83 | how does it work? |
| 16:18.05 | Strom_M | you don't need it |
| 16:19.37 | zoa | Qwell: can you reproduce that acces violation ? we cant |
| 16:19.45 | Qwell[] | nope, only happened the one time |
| 16:20.31 | Qwell[] | zoa: idefisk has an echo can? hmm, I didn't see that |
| 16:22.26 | giasai68 | hello, can you help me to find the error. if i meke the same query in mysql the query work, but in asterisk this query don't work |
| 16:22.28 | giasai68 | exten => _39X.,2,MYSQL(Query resultid ${coccobill} SELECT\ number\ from\ portability\ where\ match(number) against('${EXTEN}') |
| 16:23.37 | giasai68 | thi is query that work in mysql :select * from gnugk.portability where match(number) against('393392236199') |
| 16:24.48 | zoa | options -> audio settings i think |
| 16:24.59 | Qwell[] | oh |
| 16:28.57 | d3bian83 | could someone help me with iax trunk between 2 asterisk? |
| 16:29.19 | Mercestes | d3bian83, We have lots of consultants here for very reasonable hourly rates |
| 16:30.10 | *** join/#asterisk Exhar (n=Roy@84-105-192-215.cable.quicknet.nl) |
| 16:30.14 | d3bian83 | really? |
| 16:30.21 | d3bian83 | how much? |
| 16:30.53 | aptura | clients always ask that |
| 16:31.18 | deeperror | just send the bill |
| 16:31.43 | *** join/#asterisk kRutOn (i=locutus@of.the-b.org) |
| 16:31.46 | kRutOn | Hello. |
| 16:32.20 | Assid | eh.. whats this? 1231190014|net01|4|01|NWIF: nw_task() - Can't find associated CCB! |
| 16:33.15 | Assid | [TK]D-Fender F1 F1 !! |
| 16:33.55 | Mercestes | d3bian83, I'm $50 an hour but I'm sure others are cheaper. You also have the wiki link for IAX2 trunking. |
| 16:34.00 | boch | giasai68, paste the cli error |
| 16:34.21 | Mercestes | *or* you could try posting your configs and your error on pastebin.ca and asking for a specific solution to the exact problem you are having. |
| 16:34.23 | Mercestes | ~pastebin |
| 16:34.24 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 16:34.45 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 16:36.14 | uwe | btw, what does it mean when i get in the logs cdr.c CDR on channel XX/XX@XX not posted or lacks end ?, i couldnt find anything relevant except chunks of code and patches |
| 16:38.48 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 16:40.53 | kRutOn | I have several SIP phones that are not always connected. When I do Dial(SIP/1&SIP/2&SIP/3) if one of them isn't connected, it doesn't ring to the calling party |
| 16:40.58 | kRutOn | Is there any way to fix that? |
| 16:44.18 | *** join/#asterisk qdk (n=qdk@80.243.125.204) |
| 16:44.33 | *** join/#asterisk ping2921 (n=marc3234@206-248-157-98.dsl.teksavvy.com) |
| 16:46.37 | Mercestes | kRutOn, That's not expected behavior I do not believe. |
| 16:49.01 | *** join/#asterisk pillowhead (n=chatzill@pool-141-157-198-141.ny325.east.verizon.net) |
| 16:49.20 | pillowhead | anyone around? |
| 16:49.29 | Mercestes | no. |
| 16:50.26 | pillowhead | aww :( |
| 16:50.36 | Mercestes | Sorry |
| 16:50.56 | pillowhead | you ever install Asterisk on OS X? |
| 16:51.05 | Mercestes | not recently. |
| 16:51.07 | pillowhead | im thinking about getting a Mac Mini for a server |
| 16:51.19 | Mercestes | Is it listed as a supported operating system? |
| 16:51.28 | pillowhead | but I'm wondering if I should try and install Linux on it |
| 16:51.35 | pillowhead | or install Asterisk under OS X |
| 16:51.39 | mquin | they're nice little boxes, not tried * on mine yet |
| 16:51.41 | pillowhead | it is listed as supported |
| 16:52.30 | Mercestes | did you read any instructions? |
| 16:52.57 | pillowhead | i've read some |
| 16:53.13 | pillowhead | i've used Asterisk before, but never installed |
| 16:53.31 | pillowhead | my main concern is that I will be running a Java app and some other stuff on it |
| 16:54.03 | pillowhead | and I've heard from some people that the X version is not great |
| 16:54.05 | Mercestes | ... |
| 16:54.22 | Mercestes | I've used a car before, but I've never driven one. Can I use a Dune Buggy on the Indy circuit? |
| 16:54.34 | pillowhead | just trying to get as much info as possible before buying hardware is all |
| 16:54.48 | pillowhead | and i don't have a ton of time to debug |
| 16:54.53 | Mercestes | There are lots of instructions on the wiki for Mac OSx installs, jus tgoogling it on my own. |
| 16:55.03 | pillowhead | k |
| 16:55.07 | Mercestes | whether it works or not?? that would be the responsibility of yoru MAC community |
| 16:55.18 | Mercestes | The primary support channel is on Linux... |
| 16:55.34 | *** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it) |
| 16:55.36 | Mercestes | if yo uwant it to ***work*** then I suggest linux. If you want it to work on Mac OSX then I suggest the wiki. |
| 16:55.50 | ping2921 | I would like to store the incoming context into a cdr field. Is there a way to do this? |
| 16:56.33 | kRutOn | Mercestes: You mean that it should ring no matter what or that I should always have the SIP extensions connected? |
| 16:56.39 | Mercestes | But asterisk on linux v/s asterisk on Mac OSX would be like the difference between a Honda and a Dune Buggy. One is a means to an end, th eother isa hobby requiring lots of personal effort and customization. Only one is a reasonable means to an end. |
| 16:57.21 | Mercestes | kRutOn, Correct. * should attempt to connect to all Peers and each peer should respond with "Ringing" individually. AFAIK asterisk should only return a dialstatus if all the peers error for some reason or another. |
| 16:57.33 | boch | do you know how can i play a 16khz tone when the channel is answered ? |
| 16:57.35 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 16:57.46 | Mercestes | boch: You mean like DISA? |
| 16:58.18 | *** join/#asterisk mooey (n=cdr@service1.off-network.vault5.net) |
| 16:59.44 | boch | my billing hw needs a polarity revert or a 16khz tone to know the start and the end of the call |
| 17:00.55 | Mercestes | boch: I see. That sounds retarded to me. Can you maybe make a recording of a 16khz tone and Playback() it? |
| 17:01.30 | Mercestes | boch: Personally.....I suggest a programmer to write a new billing software and scrap the sillly billing HW. |
| 17:01.50 | *** join/#asterisk FlatFoot (n=simon@80.88.192.83) |
| 17:01.55 | FlatFoot | afternoon all |
| 17:02.07 | `p4r14h`work | anyone know of a gxp-2000 update for daylight savings time switch? |
| 17:02.24 | FlatFoot | anyone used the Innocom GSMline 900/1800 ? |
| 17:02.51 | boch | Mercestes, thats what i thought, but Dial() wont return until the end of the call, so i cant know when the call was answered |
| 17:03.28 | boch | Mercestes, i have many clients with this issue, cant tell them to buy a proper billing system.. |
| 17:03.33 | aptura | My two way audio does not drop off anymore on my DID with no changes to my asterisk. |
| 17:04.06 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-73-100.dsl.peoril.sbcglobal.net) |
| 17:04.19 | MACscr | does asterisk support G.723? |
| 17:04.22 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 17:04.37 | zoa | yes |
| 17:04.42 | zoa | but you need an add-on card |
| 17:04.47 | zoa | if you want to do more than just pass through |
| 17:04.54 | Mercestes | boch: ... I didn't know retarded billing hw became a standard. you could recode dial. |
| 17:05.09 | Mercestes | boch: or you could AGI it to playbakc a tone just before dial. |
| 17:05.19 | zoa | http://www.attractel.com/rateengine.html -> shameless plug |
| 17:05.20 | wunderkin | boch, you can, unless you are using analog... also there is the cdr |
| 17:05.37 | Mercestes | boch: Or you could run * through this billing hw you have and let it make thje calls, or have it call through asterisk or something. |
| 17:05.37 | MACscr | all im looking for is a codec to reduce bandwidth between my provider and I |
| 17:06.08 | Qwell[] | MACscr: g729, or buy hardware that can do g723 |
| 17:06.15 | Mercestes | boch: I have to wonder tho, if you are using old billing hw on new PBX software if your not charing them old rates for cheaper service. ;) |
| 17:06.41 | Sweeper | Mercestes: Progress! |
| 17:06.54 | Mercestes | Sweeper, Yay! Congratz |
| 17:06.54 | MACscr | Qwell[]: either way i would have to buy something :P |
| 17:07.16 | Sweeper | Mercestes: I was referring to the cheap service/old rates |
| 17:07.19 | Qwell[] | MACscr: welcome to the world of patents |
| 17:07.22 | aptura | qwell ever seen a case of bridging failing after 2 seconds? Just a second ago was able to test my DID and audio was passed both ways for 10 seconds. now its failing after 2 seconds. |
| 17:07.23 | Mercestes | boch: ${DIALSTATUS} should give yo usome dial feedback. |
| 17:07.29 | Sweeper | but there is also job progress, I've got two interviews today :3 |
| 17:07.32 | Mercestes | Sweeper: oh, lol. Yea. |
| 17:07.41 | Mercestes | Sweeper, good luck |
| 17:07.47 | boch | Mercestes, its the only way to bill here in argentina, how do you call the "phones shops" ? |
| 17:08.18 | Sweeper | boch: di lo en espan~ol |
| 17:08.18 | Mercestes | boch: I wasn't aware argentina had an exclusive phone switch market. |
| 17:08.40 | boch | i mean locutorios, where you pay for the calls you make |
| 17:09.06 | Mercestes | boch: I think your billing HW should be running through asterisk, or asterisk should be running through yoru billing HW, in some way that gives yoru billing HW answer supervision. |
| 17:10.45 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 17:10.48 | diclophis-work | hello all |
| 17:11.25 | Mercestes | 'ello |
| 17:11.26 | diclophis-work | so i have an interesting problem |
| 17:11.48 | Mercestes | hit us with it. |
| 17:12.07 | diclophis-work | when I call number A from asterisk box 1->asterisk box 2->zap channel to a t1 pstn connection, i get a "circuit-busy" |
| 17:12.28 | diclophis-work | but when i call number A from sjphone->asterisk box2->zap channel it goes through fine |
| 17:12.31 | Mercestes | boch: I collect the CDRs that asterisk provides and write my own billing software instead of trying to interface with som eretarded piece of HW that wants me to flip around my polarities and play audible tones to facilitate answer supervision. |
| 17:12.44 | diclophis-work | SIP being the connection between my softphone and asterisk |
| 17:12.48 | diclophis-work | and between the two asterisk boxes |
| 17:13.06 | Mercestes | diclophis-work, What does the CLIs say at each hop? |
| 17:13.30 | diclophis-work | damn i wasnt watching the CLI on the first asterisk box |
| 17:13.41 | boch | Mercestes, i would love to do it on that way, but i cant :( my work is to get this shit working |
| 17:13.49 | diclophis-work | the second one (my outbound service box) says Everyone is busy/congested at this time (1:0/1/0) |
| 17:13.56 | Mercestes | boch: Shit is a good word for it. |
| 17:13.56 | diclophis-work | i have some pri debug output too |
| 17:14.24 | diclophis-work | note that dialing other numbers through the dual asterisk setup works fine |
| 17:14.25 | Mercestes | diclophis-work, sj phone on both asterisk A and asterisk B? |
| 17:14.31 | diclophis-work | no |
| 17:14.37 | Mercestes | diclophis-work, So it's just one number that doesn't work? |
| 17:14.47 | diclophis-work | the first case is a call originating from an asterisk box to asterisk b |
| 17:14.55 | diclophis-work | the second case (the working case) is sjphone to asterisk b |
| 17:15.09 | diclophis-work | but it in the first case, its only that one number thats broken.. |
| 17:15.10 | *** join/#asterisk SoftIce (n=phil@vc-196-207-45-253.3g.vodacom.co.za) |
| 17:15.12 | Mercestes | diclophis-work, .call file? voIP phone? Soft phone? Telepathy? |
| 17:15.14 | [TK]D-Fender | diclophis-work: please pastebin CLI output from both boxes for this call..... |
| 17:15.16 | diclophis-work | which doesnt make sense |
| 17:15.31 | SoftIce | goodday, please can somebody tell me with rtp, if I have a range of say 50 ports does my carriers have to use the same port range? |
| 17:15.39 | diclophis-work | it is a call originated with the manager API |
| 17:15.46 | diclophis-work | but half of the call is in a Local channel |
| 17:15.46 | Mercestes | diclophis-work, yea, I agree with Fender. We need CLI errors on both boxes. |
| 17:15.48 | SoftIce | or does both sides port range have to be the same? |
| 17:15.55 | Mercestes | grabbign a sammich. BRB |
| 17:15.58 | diclophis-work | and the other half of the call is in an agi script doing the dialing |
| 17:18.24 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 17:18.36 | __freedom__lover | hi all |
| 17:19.43 | diclophis-work | http://pastie.caboo.se/48531 |
| 17:19.50 | diclophis-work | i have changed the last 4 digits of the number to XXX |
| 17:19.52 | diclophis-work | er XXXX |
| 17:20.47 | boch | Mercestes, M() option of Dial() is what i was looking for :D hope it works |
| 17:21.02 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 17:21.49 | ping2921 | i use cdr mysql; how do I set the userfield? I have already tried set(CDR(userfield)=xxx |
| 17:21.53 | diclophis-work | what would cause 2 numbers to behave differently though |
| 17:22.12 | Corydon-w | ping2921: in the config file |
| 17:22.20 | diclophis-work | ping2921: have you tried SetUserField ? |
| 17:23.10 | JunK-Y | ping2921: which * version? |
| 17:23.20 | Corydon-w | ping2921: [global] userfield=1 in cdr_mysql.conf |
| 17:23.55 | SoftIce | how can I have set my cdr logs to rotate every 100k ? |
| 17:24.04 | Corydon-w | SoftIce: you can't |
| 17:24.12 | SoftIce | Corydon-w: you can't? |
| 17:24.19 | Corydon-w | you can't |
| 17:24.28 | SoftIce | so what is the solution |
| 17:24.29 | Corydon-w | Not within Asterisk, anyway |
| 17:24.32 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool2-40.mtco.com) |
| 17:24.57 | SoftIce | isn't logs generated with syslog anyway |
| 17:25.21 | SoftIce | Corydon-w: please answer this |
| 17:25.26 | Corydon-w | The solution is use an external process to monitor logfile size, then send an "asterisk -rx 'logger reload'" after you swap the file |
| 17:25.27 | SoftIce | <PROTECTED> |
| 17:25.35 | Corydon-w | SoftIce: not usually, no |
| 17:25.44 | ping2921 | it works now, I had to enable userfield in cdr_mysql.conf as suggested by corydon. |
| 17:25.50 | SoftIce | so it doesn't matter what port it goes out on |
| 17:25.52 | Corydon-w | SoftIce: you can use syslog, but that is not the default configuration |
| 17:26.03 | diclophis-work | wtf is the ISDN signal "ALERTING" ? |
| 17:26.42 | Qwell[] | my guess is that it's an alert of some type |
| 17:26.57 | Qwell[] | Corydon-w: I had a brilliant idea yesterday |
| 17:26.59 | Corydon-w | ALERTING iirc is aka RINGING |
| 17:27.08 | Qwell[] | Duff's Device... |
| 17:27.11 | Qwell[] | ...in dialplan logic |
| 17:27.19 | Mercestes | boch: Let me know incase I run into that question again. :) |
| 17:27.39 | diclophis-work | Mercestes: did you see the pastie i sent? |
| 17:27.54 | *** join/#asterisk Flosoft (n=admin@d51A47591.access.telenet.be) |
| 17:28.04 | Flosoft | hey |
| 17:28.08 | Corydon-w | Qwell[]: you're a sick, sick boy |
| 17:28.11 | Flosoft | I have got a question |
| 17:28.26 | Flosoft | does anyone know Voiceone? |
| 17:28.33 | diclophis-work | i dont |
| 17:28.34 | Flosoft | the Webinterface for asterisk? |
| 17:28.35 | Mercestes | diclophis-work, Looking at it now. This doesn't look much like CLI output tho. :( |
| 17:28.53 | diclophis-work | Mercestes: its intense pri debug + CLI output... |
| 17:29.41 | __freedom__lover | hey brothers, i have a question about moh. can anyone help me? |
| 17:29.46 | mquin | diclophis-work: http://www.ciscopress.com/articles/article.asp?p=29737&seqNum=3&rl=1 <=- take a look at the section titled "ISDN call flows" |
| 17:30.08 | diclophis-work | mquin: i have been looking for something like that forever |
| 17:30.10 | diclophis-work | thanks |
| 17:30.29 | Mercestes | brb. They screwed up my sammich |
| 17:31.04 | diclophis-work | damnit man |
| 17:31.11 | diclophis-work | a sammich is a sammich |
| 17:31.18 | cpm | indeed |
| 17:31.19 | Qwell[] | except when it's bologna |
| 17:31.20 | cpm | eat it! |
| 17:31.35 | diclophis-work | yea, if its bologna take it back |
| 17:31.40 | cpm | this is serious! We are professionals here! |
| 17:31.53 | Qwell[] | sammiches are serious business though |
| 17:32.17 | Qwell[] | ...now I want one |
| 17:32.27 | zoa | me too |
| 17:32.51 | cpm | buck up, get yer own bologna |
| 17:33.18 | diclophis-work | mquin: do you recomend this book? |
| 17:35.03 | aptura | There is a sammich bc. The city just lost the bid for the 2010 olympic bid part of the olympics. |
| 17:35.47 | diclophis-work | so your saying that sammich bc can't cut the mustard? |
| 17:36.03 | SoftIce | hmm, setting assured forwarding and expedited forwarding with iptables does that actually prioritize the traffic, or what would need to be used in conjunction |
| 17:36.04 | cpm | wow! that's a reach! |
| 17:37.03 | SoftIce | with something else |
| 17:37.08 | SoftIce | sorry forgot the rest of my setence :) |
| 17:37.56 | Corydon-w | Don't you hate that? |
| 17:39.08 | aptura | diclophis-work I did not read the article on the front page of the vancouver sun but it said the city is just realing from the loss of the olympic bid. |
| 17:39.35 | aptura | I think some of the games would have been held in the city. |
| 17:39.37 | cpm | where did they lose it? did they ask their wife if it's on the back of the toilet? |
| 17:40.41 | aptura | Dont know. right now there is at least 1 billion of construction projects being build because of the pending olympics |
| 17:41.15 | *** join/#asterisk marc\cba (n=marc@cpc1-whit2-0-0-cust972.cdif.cable.ntl.com) |
| 17:42.06 | SoftIce | I have setup tc, to asure 80% of my tested b/w to rtp port range. would that and expedited forwarding for rtp/iax/sip and then setup assured forwarding for udp |
| 17:42.18 | Mercestes | back. |
| 17:42.26 | SoftIce | would that soltion best fit low b/w networks? |
| 17:43.05 | marc\cba | folks |
| 17:43.14 | Mercestes | diclophis-work, The intense pri debug out put is more distracting than helpful at thsi point but let me pic through it. and they pu tno veggies on my sammich. That's nasty |
| 17:43.22 | marc\cba | is there someone who can explain the G729 licencing to me? |
| 17:43.37 | marc\cba | i get the impression that i only need a licence for my trunk communications |
| 17:43.51 | marc\cba | what if i want to connect my extensions to the * server using G729? |
| 17:43.55 | marc\cba | do i require a licence then? |
| 17:44.16 | *** join/#asterisk Fieldy (i=yvUo9jWf@gentoo/contributor/Fieldy) |
| 17:44.20 | Corydon-w | marc\cba: generally, yes |
| 17:44.29 | *** join/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net) |
| 17:44.48 | Corydon-w | marc\cba: there are a few occasions where a license won't be used, but for general usage, a license is suggested |
| 17:45.44 | anonymouz666 | exten => s,n(allow),Set(${ARG4}=${CURL(http://www.xxx.com/secure/test.cfm?${ARG5}=${ARG1})} - Asterisk says it is missing a } |
| 17:45.47 | anonymouz666 | where ? |
| 17:46.00 | Mercestes | diclophis-work, This is PBX #2 isn't it? |
| 17:46.00 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 17:46.08 | marc\cba | Corydon-w |
| 17:46.11 | Corydon-w | anonymouz666: at the end |
| 17:46.22 | Corydon-w | anonymouz666: andit's missing a ), not a } |
| 17:46.22 | Dovid | anyone here in Israel that can help me with a traceroute ? |
| 17:46.45 | Mercestes | diclophis-work, I give up. Put this in pastebin.ca pastie is cutting off important lines. |
| 17:46.49 | Corydon-w | anonymouz666: specifically, it's missing the closing ) for the Set( |
| 17:47.20 | anonymouz666 | oh that's right |
| 17:47.23 | anonymouz666 | i am blind |
| 17:47.42 | anonymouz666 | thanks again Corydon-w |
| 17:48.01 | Corydon-w | anonymouz666: if you use vim, you can type % on top of any delimiter and it will find the matching delimiter for you |
| 17:48.02 | diclophis-work | Mercestes: what does this mean: Got SIP response 503 "Service Unavailable" back from 192.168.55.175 |
| 17:48.26 | diclophis-work | thats the message i get from case 1 on the first asterisk box (the box the call is originating from) |
| 17:48.48 | Mercestes | diclophis-work, it's bad and i needed that line |
| 17:49.07 | diclophis-work | ... is it something on my side or my telcos side? |
| 17:49.14 | anonymouz666 | Corydon-w: thanks i didn't know that and yes I use sim |
| 17:49.17 | Mercestes | turn pri debug off give me sip stuff first |
| 17:49.18 | anonymouz666 | vim |
| 17:50.41 | Mercestes | I'm betting it's a funny dialplan thin in extyensions.conf |
| 17:50.52 | SoftIce | hmm, so if I see no 'unsupported' this that or the next thing in the CLI then my strings should be correct? and upgrading to 1.4 wouldn't be to much of a problem? |
| 17:53.13 | *** join/#asterisk [jwb] (i=jwb@jwb.sh) |
| 17:54.02 | *** join/#asterisk barrys (n=barrys@128.227.123.61) |
| 17:54.02 | JerJer | Is there not like a Solaris version of G.729 ? |
| 17:56.51 | Qwell[] | JerJer: there is a version up, but it's old |
| 17:56.57 | JerJer | hmm ok |
| 17:57.16 | JerJer | i just had a customer whine about not having G.729 on Solaris |
| 17:57.41 | *** join/#asterisk goldenear (n=goldenea@2001:6f8:392:1:213:2ff:fe4a:53a7) |
| 17:58.05 | SoftIce | what is he using a fbsd version :) |
| 17:59.09 | *** join/#asterisk Exhar (n=Roy@84-105-192-215.cable.quicknet.nl) |
| 17:59.12 | *** join/#asterisk elaps (n=brokersb@ool-18bb695a.dyn.optonline.net) |
| 17:59.36 | anonymouz666 | how can I concat a string with an ${ARG1} ? |
| 17:59.47 | elaps | how do you set up a dialplan that accepts all calls to that one extension |
| 18:01.40 | *** join/#asterisk mquin_ (n=mike@pdpc/supporter/active/mquin) |
| 18:03.58 | generalhan | does anyone have a good reference for setting up remote SIP phones behind NAT ? i have been trying this for 2 weeks now and i still cant get any of my remote phones to register |
| 18:04.01 | *** part/#asterisk mooey (n=cdr@service1.off-network.vault5.net) |
| 18:04.47 | generalhan | im using a Cisco 7960, an Aastra 9112i, and X-Lite 3.0 & 2.0 and i cant get any of them to register |
| 18:05.01 | *** join/#asterisk crashev (i=crashev@bioinfo.pl) |
| 18:06.04 | elaps | xlite should not have any problem registering |
| 18:06.12 | codestr0m | generalhan: have you tried plugging the other computer in? |
| 18:06.27 | codestr0m | is our internet connected. ;) |
| 18:06.30 | codestr0m | your* |
| 18:06.36 | diclophis-work | Mercestes: why would it only effect that one number dialed thoughj? |
| 18:07.13 | elaps | generalhan: make sure you point x-lite to your * server |
| 18:07.23 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 18:07.55 | Mercestes | diclophis-work, If it was a dialplan thing, yea... and the service unavailable would affect any number that got that message. |
| 18:07.57 | diclophis-work | http://pastie.caboo.se/48542 is my extensions.conf on my outbound machine |
| 18:08.45 | diclophis-work | http://pastie.caboo.se/48543 is my extensions.conf on my origination machine |
| 18:08.56 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:09.02 | diclophis-work | its for a click-to-talk system |
| 18:09.17 | diclophis-work | the calls are originated using the manager api |
| 18:09.36 | diclophis-work | then sent to an local channel, and an agi script |
| 18:09.44 | diclophis-work | where they then dial through my outbound machine |
| 18:11.20 | Mercestes | .... pastie.caboo.se cuts off lines. |
| 18:11.39 | Mercestes | can I have a cli of just verbose 99 first not on pastie sinc eit cuts off long lines? |
| 18:11.49 | Katty | so is there a way to format the text coming out of smbclient -m into a net send command spammy message? |
| 18:11.53 | Katty | short of \\r for a line break |
| 18:12.23 | diclophis-work | the problem is i cant get the debug again because i cant bother the customer |
| 18:12.57 | SoftIce | hmm, anyone know what port offhand h323 uses |
| 18:12.57 | Mercestes | can you ssh in to these boxes?? |
| 18:13.20 | Mercestes | SoftIce, I bet google h323 port would answer your question on th efirst page. |
| 18:13.40 | elaps | how do you set up a dialplan that accepts all calls to that one extension |
| 18:14.20 | diclophis-work | if you have logrotated you can setup a rule to rotate the logs, then reload/restart asterisk |
| 18:14.35 | diclophis-work | Mercestes: yes |
| 18:14.45 | Mercestes | diclophis-work, Does this number work???? |
| 18:15.02 | diclophis-work | yea i was able to call the number using my cell phone |
| 18:15.18 | diclophis-work | and call the number in case 2, where i connect to my outbound machine and originate the call using sjphone |
| 18:15.20 | Mercestes | diclophis-work, Does it work in the setup in question taht I am troubleshooting? |
| 18:16.15 | generalhan | elaps: the x-lite is pointed to the WAN address of the router that the * server is attached to. the ports are forwarded from that interface to my |
| 18:16.19 | generalhan | * box |
| 18:16.47 | diclophis-work | i know this has to be difficult to help with |
| 18:16.55 | diclophis-work | the system is convoluted |
| 18:17.03 | diclophis-work | the number is a 3rd party customer |
| 18:17.03 | [TK]D-Fender | generalhan: You do NOT have to forward ports for a single remote phone behind NAT |
| 18:17.10 | diclophis-work | i have no idea what the hell i am doing |
| 18:17.26 | diclophis-work | and I can't replicate the scenario in less that 5000 wrds |
| 18:17.33 | generalhan | [TK]D-Fender: on the remote side or the local side ? |
| 18:17.55 | generalhan | or both ? |
| 18:18.01 | Mercestes | diclophis-work, My point is.....if you can ssh in, and the # doesn't work in the scenario we are testing...then I fail to see at wha tponit we are disturbing the customer. |
| 18:18.02 | [TK]D-Fender | generalhan: the NAT the phone is beinhind should not need to do anything. if your SERVER is behind NAT then you have your OWN settings to do. |
| 18:18.08 | anonymouz666 | for example, if I have in a var this values: var=512,520,530,540 - which is the best way to parse it to say: for 512 press 1 and so on ? |
| 18:18.28 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 18:19.06 | generalhan | [TK]D-Fender: thats what im talking about ,... the * server is behind an IPCop box, so i forwarded my RTP range, and port 5060 from the IPCop box to the * machine |
| 18:19.47 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 18:20.08 | codestr0m | generalhan: have you tried using tcpdump to see if the packets are actually getting to the * box 2) sip debug will be your friend.. two weeks on this? |
| 18:20.10 | diclophis-work | Mercestes: to recreate the output |
| 18:20.12 | generalhan | [TK]D-Fender: but i did also forward the ports on the remote router, that the phone is behind, will that mess everything up? or is it just an uneccesary step |
| 18:20.15 | diclophis-work | i am digging through the log files now |
| 18:20.34 | [TK]D-Fender | generalhan: Definately unneccesary and CAN possible screw things up. |
| 18:20.46 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 18:21.08 | generalhan | codestr0m: sip debug shows me nothing, because i dont think im even getting to the * machine. and when i say 2 week, i dont mean straigh .. when ever i have spare time i try to get thi solved |
| 18:21.09 | [TK]D-Fender | generalhan: you also need to have the usual pile of sip.conf entries for your * server to work behind NAT. |
| 18:21.20 | [TK]D-Fender | generalhan: and jsut "nat=yes" for the phone's specific entry |
| 18:21.23 | [TK]D-Fender | ~sipnat |
| 18:21.33 | jbot | methinks sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:21.34 | generalhan | [TK]D-Fender: ok i was unaware ... ill take those forwarders off |
| 18:21.50 | codestr0m | generalhan: well. I'd say it pretty clear that if the * box isn't seeing packets and tcpdump isn't showing them you know where to look right? |
| 18:21.52 | generalhan | [TK]D-Fender: yea ive gotten that in |
| 18:22.07 | generalhan | i am not using tcpdump |
| 18:22.21 | *** join/#asterisk step_quasar (n=step_qua@250-171-114-200.fibertel.com.ar) |
| 18:22.30 | generalhan | or rather havent yet |
| 18:22.37 | codestr0m | generalhan: well. try using tcpdump then and see what raw packets your seeing... |
| 18:23.16 | marc\cba | hmm |
| 18:23.16 | marc\cba | does anyone know if there are any cisco phones that support free, low-bandwidth codecs? |
| 18:23.16 | marc\cba | my 7940 only supports g711 and g729a |
| 18:23.21 | [TK]D-Fender | generalhan: pastebin your sip.conf [general] section & the peer entry masking only passwords |
| 18:23.33 | [TK]D-Fender | marc\cba: No. |
| 18:23.39 | Katty | how do i smbclient -M computername with a txt file? |
| 18:23.40 | Mercestes | diclophis-work, Nice, uber helpful. lol. can I ge ta copy of that AGI then? |
| 18:23.43 | *** join/#asterisk svenna_ (n=svenna@p548d0890.dip0.t-ipconnect.de) |
| 18:23.54 | diclophis-work | Mercestes: i have isolated it down the parts i think are relevant |
| 18:23.54 | diclophis-work | http://pastie.caboo.se/48546 |
| 18:24.01 | Mercestes | KATTY!!!! msn me! |
| 18:24.08 | [TK]D-Fender | marc\cba: These are business phone and support codec used by business solutions. G.729 is only beat out by G.723 and thats massively patent encumbered |
| 18:24.19 | diclophis-work | well, not to get all capitalistic on you, but thats not my propertie |
| 18:24.41 | marc\cba | how depressing |
| 18:24.44 | marc\cba | ta fender |
| 18:25.18 | [TK]D-Fender | diclophis-work: Sure olooks like the # is BUSY.... |
| 18:25.21 | Mercestes | diclophis-work, Box2 output?? |
| 18:25.50 | [TK]D-Fender | diclophis-work: This is your box WITH the card thats reporting back... so when it says busy, it MEANS it. |
| 18:25.54 | diclophis-work | Mercestes: that is box2 output (my outbound box) |
| 18:25.59 | [TK]D-Fender | diclophis-work: Call it YOURSELF by hand |
| 18:26.02 | diclophis-work | thats what i have been thinking all along |
| 18:26.05 | diclophis-work | when i call it its not busy |
| 18:26.15 | diclophis-work | could it be super coincedence? |
| 18:26.18 | Mercestes | it looks like circuit busy to me... |
| 18:26.27 | Mercestes | was this box idle before it sent the call? |
| 18:26.30 | diclophis-work | when the call goes through there using the click-to-talk system it always returns busy |
| 18:26.36 | diclophis-work | the box has 23 channels |
| 18:26.41 | diclophis-work | this one 1 of 2 active channels |
| 18:26.55 | Mercestes | yea, I know that. Why did it go out on 2-1? Was 2-1 in use at th etime??? |
| 18:27.19 | diclophis-work | well .. why would asterisk try to send out a call on a used channel? |
| 18:27.32 | diclophis-work | i am using the "group" syntax for the dial command |
| 18:27.38 | Mercestes | diclophis-work, The answre usually is "Becauseyou told it to." and I can see that. |
| 18:27.39 | diclophis-work | Dial(Zap/g1/blah) |
| 18:27.55 | Mercestes | diclophis-work, and IIRC lowercase "g" is the one you want for outbound. |
| 18:28.08 | diclophis-work | yea... |
| 18:28.10 | diclophis-work | wait |
| 18:28.14 | diclophis-work | back up the bus here |
| 18:28.20 | [TK]D-Fender | diclophis-work: Listen this is not zaptel telling you there is no free channel. The person you are CALLING is busyt. |
| 18:28.29 | diclophis-work | ok |
| 18:28.34 | diclophis-work | that is somewhat good news |
| 18:28.45 | diclophis-work | how do i convince the customer their line was busy when they swaer its not |
| 18:28.51 | diclophis-work | and everytime we try to test it |
| 18:28.55 | diclophis-work | it always returns busy |
| 18:28.57 | [TK]D-Fender | diclophis-work: Get off your butt and dial it YOURSELF! :) |
| 18:29.03 | diclophis-work | but when i call via my cell it goes through |
| 18:29.08 | Mercestes | He did, it rings through |
| 18:29.12 | aptura | :) |
| 18:29.18 | Mercestes | only fails on his click to dial app. |
| 18:29.29 | diclophis-work | i mean.. is it super duper coincedence |
| 18:29.30 | Mercestes | works if he dials from box2 using a softphone, or from his cell, etc. |
| 18:29.33 | [TK]D-Fender | diclophis-work: turn on PRI debug for more backup |
| 18:29.38 | aptura | yea work is work. How many here are always on there butts and can loose a few pounds? |
| 18:29.43 | Mercestes | only fails on this one number gonig from AGI box 1 to box 2 to zap channel |
| 18:29.43 | diclophis-work | [TK]D-Fender: haha i had that on |
| 18:29.44 | *** part/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 18:29.57 | diclophis-work | it goes from SETUP->PROCEEDING->DISCONNECT (when dialing using my app) |
| 18:30.12 | diclophis-work | when dialing using my softphone its SETUP->PROCEEDING->ALERT->CONNECT |
| 18:30.30 | [TK]D-Fender | care to share that pastebin? |
| 18:31.00 | *** join/#asterisk fbcit (n=cnighswo@nc-71-0-121-24.sta.embarqhsd.net) |
| 18:31.08 | Mercestes | <PROTECTED> |
| 18:31.12 | Mercestes | There you Fender |
| 18:31.15 | diclophis-work | ... its been changed |
| 18:31.17 | *** join/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 18:31.22 | diclophis-work | oh yea |
| 18:31.34 | diclophis-work | that one is with all the debug |
| 18:32.30 | Mercestes | gets interesting around 124. |
| 18:32.37 | Mercestes | Cause unknown, network congestion. |
| 18:32.41 | [TK]D-Fender | <PROTECTED> |
| 18:32.42 | [TK]D-Fender | < Ext: 1 Cause: Unknown (34), class = Network Congestion (2) ] |
| 18:32.49 | diclophis-work | yea... |
| 18:32.52 | diclophis-work | whats that all about |
| 18:33.00 | diclophis-work | "< |
| 18:33.08 | diclophis-work | damnit: "<" means from the telco right? |
| 18:33.09 | [TK]D-Fender | diclophis-work: their phone = unluckily BUSY <- |
| 18:33.12 | Juggie | exactally what it says |
| 18:33.15 | *** join/#asterisk shinux__ (n=shinux@80.89.187.101) |
| 18:33.31 | Mercestes | oh hey, if i hit "viwe" it quits being retarded! |
| 18:33.57 | diclophis-work | pastie.caboo.se is the best pastie system to date |
| 18:33.59 | diclophis-work | IMHO |
| 18:34.20 | [TK]D-Fender | pastebin.ca > all |
| 18:34.22 | Mercestes | diclophis-work, only if you hit view. |
| 18:34.27 | diclophis-work | i did make the mistake of setting to highlight my pri debug as ruby code |
| 18:34.43 | diclophis-work | pastebin.ca sold out |
| 18:38.39 | [TK]D-Fender | diclophis-work: Sold out how? |
| 18:38.48 | diclophis-work | its covered in ads |
| 18:39.29 | [TK]D-Fender | diclophis-work: Single add, big deal. Far more readable and I can EDIT what you paste to help correct it! |
| 18:39.53 | [TK]D-Fender | diclophis-work: pastie KILLS content, pastebi.ca lets me even edit it and send back. |
| 18:39.53 | SoftIce | anyone clued up with iptables here? |
| 18:39.56 | [TK]D-Fender | pastebin.ca > ALL. |
| 18:40.51 | diclophis-work | haha |
| 18:40.53 | diclophis-work | anyhow |
| 18:41.13 | [TK]D-Fender | :D |
| 18:41.20 | diclophis-work | so, the consensus is... the number i am trying to dial just happens to be in fact busy, every time i try to dial it through my system, but unbusy when i dial directly |
| 18:43.11 | *** join/#asterisk jart (n=user@ool-43509aa5.dyn.optonline.net) |
| 18:44.55 | Mercestes | I don't think that. =/ |
| 18:45.08 | diclophis-work | damnit |
| 18:45.14 | diclophis-work | what is your best guess? |
| 18:45.20 | [TK]D-Fender | diclophis-work: ISDN 24 = busy.... |
| 18:45.23 | [TK]D-Fender | 34* |
| 18:47.39 | *** join/#asterisk ToyMan (n=Stuart@pool-72-84-23-73.pghk.east.verizon.net) |
| 18:49.00 | *** join/#asterisk NormSteel (n=nathank@69.17.44.81) |
| 18:49.28 | Mercestes | 34 is No Circuit Channel Available, not busy |
| 18:50.39 | NormSteel | any one ever seen a SPA-3000 kill a dsl line when it hangs up a call? |
| 18:51.09 | Mercestes | http://www.cisco.com/univercd/cc/td/doc/product/software/ios11/dbook/disdn.htm |
| 18:51.27 | gambolputty | how could it? |
| 18:51.58 | kRutOn | it diverts 120VAC to the Ethernet in frustration |
| 18:52.10 | Mercestes | gambolputty, send out any disruption that resides within the frequency range that DSL occupies. |
| 18:52.17 | diclophis-work | Mercestes: would that be caused by my machine, or my telcos? |
| 18:52.24 | Mercestes | it's just a higher frequency carrier signal. |
| 18:52.26 | gambolputty | does your dsl line have a splitter? |
| 18:52.34 | uwe | should bugs/problems related to building mpg123 using make mpg123 on amd64 be reported to asterisk ? |
| 18:52.34 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:52.37 | Mercestes | diclophis-work, Yes. |
| 18:52.55 | Mercestes | uwe: No. we dont' use mpg123 |
| 18:53.03 | diclophis-work | so, somehow my machine is attempting to make a call on a channel that is used... |
| 18:53.14 | diclophis-work | then maybe its related to the way I am dial()ing ? |
| 18:53.20 | Mercestes | diclophis-work, Or unavaible.....or nonexistant |
| 18:53.28 | diclophis-work | non-existant... hmm |
| 18:53.36 | *** join/#asterisk CrashSys (n=kumba@bartleby.crashsys.com) |
| 18:53.55 | Mercestes | could have something to do with dial/g1 taking you to zap/2-1 *shrugs* |
| 18:54.34 | diclophis-work | ... oh Zap/2-1 is "trunk 2" "channel 1" ? |
| 18:54.42 | uwe | hmmm, AsteriskTFOT states that asterisk works with mpg123, and doing make mpg123 will try to install mpg123 |
| 18:54.47 | diclophis-work | i only have one trunk plugged into this machine |
| 18:54.51 | [TK]D-Fender | Mercestes: A few deprecated ISDN implementations lin NI1 use 34 for Busy still... |
| 18:54.55 | Mercestes | diclophis-work, Not 100% sure on the channel syntax but it sounds reasonable. |
| 18:55.07 | Mercestes | [TK]D-Fender, Oh.... |
| 18:55.24 | Mercestes | [TK]D-Fender, then it could be busy. =/ |
| 18:55.27 | diclophis-work | but in my zapata.conf i have group=1, channel 1-23 |
| 18:55.30 | [TK]D-Fender | Mercestes: Just FYI, but I had to face this >HERE< |
| 18:55.39 | CrashSys | I always figured Zap/2-1 meant Channel 2, Stream 1 |
| 18:55.50 | anonymouz666 | when calling a macro from a gotoif i should use macro-dial-blah ou just dial-blah |
| 18:55.50 | GreyFoxx | Can anyone here recommend a tool for monitoring channel usage? Right now we've got a shell script on a crontab but are looking for something a little "realtime" |
| 18:56.02 | CrashSys | I've never seen a call to Zap/2-1 |
| 18:56.02 | Mercestes | damnit. File?? What does zap/2-1 mean? lol |
| 18:56.05 | CrashSys | err Zap/2-2 |
| 18:56.15 | *** part/#asterisk jart (n=user@ool-43509aa5.dyn.optonline.net) |
| 18:56.16 | [TK]D-Fender | CrashSys: You don't get multiple streams on FXO or on digital ZAP interfaces. |
| 18:56.28 | [TK]D-Fender | CrashSys: So it'll always be -1 |
| 18:56.30 | CrashSys | But you do on sip... so maybe it's there for consistency? |
| 18:56.35 | CrashSys | Yeah |
| 18:56.38 | Mercestes | Ah, okies. |
| 18:56.54 | Mercestes | so channel-stream? Cool. |
| 18:57.01 | CrashSys | You only have 1 stream/session on Zap.. so it's always -1 |
| 18:57.03 | Mercestes | diclophis-work, That eliminates most of my concerns then. |
| 18:57.04 | diclophis-work | so Zap/2-1 is circuit-busy means channel 2 (out of 1-23) |
| 18:57.07 | [TK]D-Fender | CrashSys: Tahts because a SIP device can hold an untold # of calls by the name. thats why it has a 4 digit suffix for "uniqueness" |
| 18:57.17 | [TK]D-Fender | diclophis-work: Correct |
| 18:57.25 | *** part/#asterisk codestr0m (n=asura@ns1.netsyncro.com) |
| 18:57.28 | diclophis-work | there are no warnings on my channels |
| 18:57.35 | diclophis-work | and no other number has been failing that way |
| 18:57.41 | CrashSys | D-Fender: Right, and the -1 added to Zap is just for consistency then... |
| 18:57.55 | Mercestes | diclophis-work, I don't think it's your channels. Otherwise it would be rather intermittent |
| 18:58.03 | [TK]D-Fender | CrashIndeed. |
| 18:59.04 | diclophis-work | so its back to the very unlikly, but appearntly possible scenario that everytime i call this number using my system its busy, and every time i call the number directly its not |
| 18:59.08 | diclophis-work | (keeping in mind whn i say directly i mean over the same wires, just not using the agi stuff) |
| 18:59.37 | CrashSys | just one number? |
| 18:59.43 | CrashSys | if you dial other similar numbers no problems? |
| 18:59.49 | diclophis-work | exactly |
| 18:59.55 | CrashSys | interesting |
| 19:00.04 | Mercestes | CrashSys, Indeed. |
| 19:00.06 | diclophis-work | in fact a nother number with the same XXX-XXX- works |
| 19:00.09 | Mercestes | katty! MSN ME! |
| 19:00.15 | Katty | ummumm |
| 19:00.16 | Katty | how? |
| 19:00.19 | Katty | don't say with a messenger. |
| 19:00.20 | CrashSys | That was my second question :) |
| 19:00.34 | Mercestes | if you really cared you'd know. :( |
| 19:00.35 | diclophis-work | (meaning the last 4 digits are the only difference) |
| 19:00.36 | Mercestes | make a wild guess |
| 19:00.43 | Mercestes | bet you get it right the first time |
| 19:01.07 | CrashSys | is the number being passed to the dial command in the same format? |
| 19:01.17 | Katty | don't make you google stalk you >.< |
| 19:01.26 | diclophis-work | yea |
| 19:01.29 | Katty | ... s/me/you/ |
| 19:01.44 | Mercestes | Katty, LOL...would probably work. |
| 19:01.45 | Katty | my brain hurts. |
| 19:01.58 | Mercestes | Katty, Make a random guess...bet it works. |
| 19:02.00 | Katty | i just google stalked someone...only knowing their first name |
| 19:02.04 | Katty | and that they're a mage |
| 19:02.05 | Katty | in wow |
| 19:02.23 | CrashSys | teh google shall inherit the earth |
| 19:02.43 | Katty | it included cached google pages. |
| 19:02.50 | Katty | and domain registration whois queries |
| 19:02.52 | CrashSys | resistance is futile... teh google shall add your distinctiveness to it'w own... |
| 19:02.53 | Katty | but i found them. |
| 19:03.04 | Mercestes | MSN ME! |
| 19:03.10 | Katty | i don't have your msn address :P |
| 19:03.15 | Mercestes | guess |
| 19:03.16 | Strom_M | hey, here's an uber dumb question |
| 19:03.25 | Strom_M | is it possible to get the b410p working with bristuff? |
| 19:03.34 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 19:03.34 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 19:03.48 | Qwell[] | Qwell: That was fast |
| 19:03.51 | CrashSys | diclophis: You said your using AGI... does it work if you use an extensions.conf dial-cmd? |
| 19:04.17 | CrashSys | or like a .call file? |
| 19:04.25 | Katty | what does \\r mean |
| 19:04.28 | CrashSys | that would rule out AGI or not... |
| 19:04.45 | Katty | more precisely...what do you call commands like \\r |
| 19:04.47 | Katty | and are there mroe |
| 19:04.50 | Katty | also, more. |
| 19:05.14 | uwe | eh, if i shouldnt use mpg123 ! what should i be using to play mp3s for MoH ? |
| 19:05.35 | JunK-Y | uwe: native. |
| 19:08.01 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:10.57 | MACscr | anyone know a good way for doing a test incoming call using a sip url? there isnt some type of site out there that will make the call and record a response is there? |
| 19:11.50 | MACscr | or i guess could someone call mine and tell me what response they get? |
| 19:12.10 | MACscr | i dont have a did associated with this account yet |
| 19:14.57 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:15.29 | anonymouz666 | Mar 21 16:15:33 ERROR[18298]: app_macro.c:149 macro_exec: Macro(): possible infinite loop detected. Returning early. - I have one thing to say: LOL |
| 19:16.11 | JunK-Y | now imagine how * is laughing. |
| 19:17.28 | CrashSys | ... that's an actual error message? |
| 19:18.11 | anonymouz666 | I put a global var and broke everything |
| 19:18.16 | *** join/#asterisk nutcase (n=nutcase@i59F56E0E.versanet.de) |
| 19:23.13 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:23.15 | blitzrage | hey all! |
| 19:23.22 | blitzrage | anyone know how I can filter for RTP using tshark? |
| 19:24.04 | anonymouz666 | no idea. |
| 19:24.33 | zoa | udp.prot == rtp or something |
| 19:24.54 | blitzrage | great thanks zoa |
| 19:25.09 | Strom_M | am I correct in assuming the B410P is HFC-based? |
| 19:25.15 | zoa | yes |
| 19:25.22 | Strom_M | woot |
| 19:25.23 | zoa | blitzrage: i think its something similar |
| 19:25.25 | zoa | not sure though |
| 19:25.26 | Strom_M | this is good news :) |
| 19:26.03 | *** join/#asterisk shinux__ (n=shinux@80.89.187.106) |
| 19:26.04 | blitzrage | zoa: yah... doesn't work exactly, but might help me on google |
| 19:26.57 | *** join/#asterisk manopulus (n=manopulu@cable-6-205.cgates.lt) |
| 19:27.47 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 19:28.01 | MrTelephone | what is the max length you can run a pri cat5 cable from a digium card? |
| 19:28.20 | Strom_M | maximum length before you need a T1 repeater: 655 feet |
| 19:28.30 | MrTelephone | I want to go around 300ft with it straight from a pri port to a nortel pbx |
| 19:28.36 | MrTelephone | next door |
| 19:28.49 | CrashSys | What strom said |
| 19:28.55 | Strom_M | see above ^^^^^^^^^^^ |
| 19:29.06 | MrTelephone | should work good with a nortel pbx? |
| 19:29.09 | manopulus | hello. i want to write some data to CDR (myql) in asterisk, together or updating existing, after-call record. i cannou use user column in table. idea is to use UPDATE sql an to use exten => h,nnn, so, will i have record in db at moment of data updating? or not? |
| 19:29.13 | *** join/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net) |
| 19:29.17 | CrashSys | You have to play with gains past 127 feet :) |
| 19:29.21 | MrTelephone | maybe I should put in a fxs breakout box |
| 19:29.21 | CrashSys | or so the doc's say |
| 19:29.31 | Strom_M | breakout box?! |
| 19:29.34 | Strom_M | are you mad? |
| 19:29.45 | CrashSys | break what? |
| 19:29.56 | MrTelephone | how much is apri card ofr a nortel pbx.. probably be cheaper to just buy a pri->fxs channel bank |
| 19:30.08 | MrTelephone | because the fax is seperate from the pbx as well |
| 19:30.20 | Strom_M | MrTelephone: and then you get echo and noise problems |
| 19:30.37 | Strom_M | and there's no such thing as a pri channel bank - T1 channel bank though... |
| 19:30.44 | MrTelephone | do faxes work through the nortel pbx? im pretty sure it should? |
| 19:30.52 | MrTelephone | sorry, t1 channel bank |
| 19:31.12 | tutt9876 | hi, someone knows why when i type sip show peers I see all the peers mentioned in sip.conf and not only those whose status is different from UNKNOWN? |
| 19:31.15 | MrTelephone | so I can hook up port 1 on the digium card to my telco provider, then port 2 to the nortel pbx nextdoor |
| 19:31.25 | CrashSys | Shouldn't get much echo from a channel bank.... I had a set-up like that using a TE205p without echo... |
| 19:31.41 | MrTelephone | i had to buy a pri because I was getting horrible echo from the telco on the analog |
| 19:31.49 | MrTelephone | but pri->channel bank should be good if it is a short run |
| 19:31.59 | MrTelephone | short run from the channel bank to the punch down I mean |
| 19:32.03 | tutt9876 | Can I clear the result of a "sip show peers" command? |
| 19:32.07 | *** join/#asterisk ParaNoir (n=daanb@84.53.99.162) |
| 19:32.12 | blitzrage | !clear |
| 19:32.12 | fakhir | Shutdown PG2 -> Uninstall -> delete C:\Program Files\PeerGuardian2 -> install -> start PG2 |
| 19:32.22 | CrashSys | C: |
| 19:32.24 | CrashSys | !?!?1 |
| 19:32.31 | ParaNoir | Hey, anybody succeeded in connecting Swyx and Asterisk by using SIP? |
| 19:32.35 | MrTelephone | will I have a lot of conflicts with the nortel pbx and a digium pri card? |
| 19:33.18 | CrashSys | MrTelephone: I used a 2-port T1 card, ran the PRI T1 into port 1, and just a regular T1 (CAS) out to a channel bank that my toshiba strata plugged into... worked fine... |
| 19:33.49 | *** part/#asterisk max_______ (i=max__@ts.bestserversllc.net) |
| 19:33.50 | CrashSys | No echo issues on the old Toshiba system... |
| 19:34.02 | CrashSys | echo issues on sip phones, but that was another problem... |
| 19:34.05 | MrTelephone | what did you use for a channel bank? |
| 19:34.09 | CrashSys | faxing/etc all worked fine through that set-up... |
| 19:34.16 | CrashSys | MrTelephone: Adit 600... |
| 19:34.20 | tutt9876 | know some trick with the "sip show peers command"? |
| 19:34.33 | MrTelephone | looks high tech |
| 19:35.05 | CrashSys | I used the Adit cause it was laying around |
| 19:35.20 | MrTelephone | yeah I'm really debating |
| 19:35.28 | CrashSys | I know of people using Zhone channel banks from e-bay without problems for FXS |
| 19:35.29 | MrTelephone | nortel pri card or channel bank |
| 19:35.37 | CrashSys | $200/bank... |
| 19:36.08 | MrTelephone | how are the rhino channel banks? |
| 19:36.14 | CrashSys | $1500 |
| 19:36.27 | CrashSys | I mean, good from what I hear... |
| 19:36.30 | MrTelephone | but hey are good? |
| 19:36.52 | CrashSys | I hear they are good, and that they will somewhat autoconfigure themselves... |
| 19:36.59 | MrTelephone | so your port1 to port2 bridge on the pri card works smoothly? |
| 19:37.32 | CrashSys | It did for me... it was all u-law so it was a native bridge... no problems with faxing/etc on the old toshiba system... |
| 19:38.03 | CrashSys | I had echo issues on sip phones because I didn't use a HW Echo-can... but the bridging worked fine for me... |
| 19:38.17 | MrTelephone | nice |
| 19:38.26 | MrTelephone | what have you heard about adtran? |
| 19:38.51 | CrashSys | good stuff, exspensive... |
| 19:39.00 | CrashSys | or can be... i'm an e-bay fan... |
| 19:39.17 | MrTelephone | i hate the thought of getting faulty equipment |
| 19:39.32 | CrashSys | I hate the thought of being broke :D |
| 19:39.43 | MrTelephone | it is for a production environment and my repuation is on the line.. a reputation that was hacked by very poor analog telco lines and fxo cards :( |
| 19:39.46 | Katty | so i spent 4 hours |
| 19:39.52 | Katty | getting asterisk to make a popup window |
| 19:39.54 | Katty | and pass variables |
| 19:40.00 | Katty | and figuring out the limitations of the window |
| 19:40.03 | Katty | and you know what my boss says? |
| 19:40.09 | Katty | it won't work - the text is too small to read. |
| 19:40.11 | CrashSys | You're fired? |
| 19:40.23 | MrTelephone | boss's are nit picky |
| 19:40.28 | *** part/#asterisk barrys (n=barrys@128.227.123.61) |
| 19:40.29 | MrTelephone | tell him to find a new employee |
| 19:40.53 | CrashSys | MrTelephone: I would recommend Sangoma A100-series cards then |
| 19:41.31 | [TK]D-Fender | Katty: Font++ |
| 19:41.54 | MrTelephone | Yes I have a sangoma card that was 1600 |
| 19:42.01 | MrTelephone | 2 port octasic echo cancel |
| 19:42.02 | MrTelephone | works well |
| 19:42.43 | CrashSys | 2-port T1 for $1600? |
| 19:42.45 | CrashSys | seems high |
| 19:43.03 | CrashSys | Seems more like an $800 card... |
| 19:43.13 | aptura | the digium cards are now on par for reliability over sangoma cards right? |
| 19:43.19 | MrTelephone | telco grade echo canceller |
| 19:43.47 | CrashSys | aptura: I hear good and bad... so I dont know... I hear only good about sangoma... |
| 19:44.05 | aptura | Same here CrashSys |
| 19:44.20 | MrTelephone | http://www.voipsupply.com/product_info.php?products_id=1910 |
| 19:44.33 | aptura | Company credibility is more important then saving a few dollars so reliability is all to important. |
| 19:44.48 | CrashSys | MrTelephone: An acquaitance of mine uses Zhone channel banks that he buys new from e-bay for like $200/pop in a call center that runs 16-hour days... they've been working for 4 years without a failure... |
| 19:45.08 | aptura | wow |
| 19:45.18 | aptura | those are some good numbers |
| 19:45.34 | Qwell[] | http://www.digium.com/en/mediacenter/news/viewpress.php?id=asterisk-appliance |
| 19:45.35 | Qwell[] | fyi |
| 19:46.41 | aptura | Qwell, are you selling those only to resellers? |
| 19:46.45 | marc\cba | <PROTECTED> |
| 19:46.45 | marc\cba | <PROTECTED> |
| 19:46.47 | Katty | so... |
| 19:46.47 | zoa | not any more |
| 19:46.56 | zoa | qwell, send me one for a review! |
| 19:46.57 | Katty | anyone know anything about live communications server? |
| 19:46.57 | Juggie | Qwell, let me know when it does T1 :) |
| 19:46.58 | Qwell[] | well, for the immediate future |
| 19:46.59 | marc\cba | ^^ how can i change this order & allowed codecs? |
| 19:47.04 | marc\cba | that was sip show peer 200 |
| 19:47.06 | zoa | its too expensive for me to buy it for a review |
| 19:47.07 | Qwell[] | Juggie: call sales, and tell them you want T1 |
| 19:47.35 | MrTelephone | how much is an adit 600 |
| 19:47.42 | aptura | I think it is a good idea to sell only though resellers. If somone wants to stay in the business of sales and support it is the best way to go. |
| 19:47.49 | Katty | isn't there a way to setup an internal IM server thingy |
| 19:49.31 | CrashSys | MrTelephone: I believe they can be had for a little less then a Rhino... |
| 19:49.39 | CrashSys | still over $1K for brand-new |
| 19:52.08 | MrTelephone | automatic impedance matching |
| 19:52.10 | aptura | finding a backbrouad wall mount case to contain a atx board and is low profile to hold the hardware is tough. I have one company that makes a case but its a little to big. Saw a Avaya low profile case the other day and had a backup flash ram stick out the side of it. Was interesting to see. |
| 19:53.00 | *** join/#asterisk phillipk (n=pkey@216.248.143.77) |
| 19:55.46 | phillipk | What's the best way to connect a remote user with a softphone, especially if they have a less-than-optimal connection? |
| 19:55.53 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:56.01 | zoa | speex |
| 19:56.07 | zoa | with iax2 |
| 19:56.17 | zoa | and a jitter buffer on both ends |
| 19:57.19 | phillipk | Can you recommend a particular softphone? |
| 19:58.41 | zoa | well i am very closely linked to one of them, called idefisk |
| 19:58.55 | zoa | so i am a little biased |
| 19:58.56 | zoa | :) |
| 19:59.01 | phillipk | That's the one we're currently experimenting with :) |
| 19:59.28 | zoa | if you want i could send you a prerelease for version 2 (with sip support) |
| 19:59.55 | phillipk | That would be cool, thanks. |
| 20:00.09 | zoa | send me a private message with your email addy |
| 20:00.19 | zoa | (something that accepts .exe files please) |
| 20:00.21 | [jwb] | I have a sangoma PRI card, I have a DID that when dialed immediately dials back out the same PRI to an analog POTS line that has a fax hanging off it.. Inbound faxing is very intermittent with transmission errors a good 70% of the time... Can anyone suggest what might be causing these issues? |
| 20:00.21 | zoa | and i will send it |
| 20:02.36 | deeperror | anyone know of a good resource for kernel upgrades on debian to 2.6.10+ |
| 20:03.01 | zoa | kernel.org ? :) |
| 20:03.14 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 20:03.25 | deeperror | zoa: more as in how to perform the operation |
| 20:03.54 | [jwb] | google ;) |
| 20:03.59 | DocHolliday | i am having DTMF issues on a NAT'd Cisco 7940 |
| 20:04.46 | zoa | http://librenix.com/?inode=3175 |
| 20:04.56 | DocHolliday | i am getting 1 way audio (receive), haven't verified send |
| 20:05.04 | deeperror | yea |
| 20:05.10 | deeperror | lame |
| 20:06.48 | Mercestes | anyone use astlinux on the soekris boxes??? |
| 20:07.36 | *** join/#asterisk shinux__ (n=shinux@80.89.187.106) |
| 20:10.25 | Mercestes | I need to know th econsole port settings |
| 20:10.28 | *** join/#asterisk stefmtl (n=stef@stef.istop.com) |
| 20:15.46 | *** join/#asterisk ltd (n=z@202-161-20-161.dyn.iinet.net.au) |
| 20:17.11 | *** join/#asterisk b11d (n=no@234-200-29-134.hcc.mnscu.edu) |
| 20:17.19 | *** join/#asterisk shinux__ (n=shinux@80.89.187.106) |
| 20:17.58 | *** join/#asterisk Voice2 (n=TripleFF@145-27.mc.cite.net) |
| 20:18.30 | *** join/#asterisk ToyMan (n=Stuart@user-0cevdmv.cable.mindspring.com) |
| 20:18.36 | Voice2 | qustion... can an exten have dashes ? like exten -> _809XX-XXX-XX, |
| 20:19.27 | b11d | no |
| 20:19.31 | b11d | the dashes are irrelevant |
| 20:19.46 | b11d | you could have _908X-X-X-XXX-X |
| 20:19.48 | b11d | it's all the same |
| 20:19.50 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 20:20.05 | b11d | _809 that is |
| 20:22.41 | b11d | man it's dead around here today.. |
| 20:24.17 | *** join/#asterisk xezz (n=sad@athedsl-164281.otenet.gr) |
| 20:24.20 | aptura | yes it is |
| 20:24.53 | aptura | looks like fonality came out with a one of a kind pbx like case to contain the electronics for trixbox. |
| 20:25.02 | b11d | enat |
| 20:25.03 | b11d | neat |
| 20:25.04 | Qwell[] | "one of a kind"? |
| 20:25.07 | Qwell[] | ha |
| 20:25.12 | xezz | hello, i have a problem dialing international calls, i am able to call any number except from internationa.. plz check : http://pastebin.ca/405544 |
| 20:25.20 | Qwell[] | it looks like an off-the-shelf server with a custom case |
| 20:25.51 | DocHolliday | afternoon Qwell[] |
| 20:25.53 | aptura | I liked the Avaya case with a recent install. Very flat and flush to the backboard. |
| 20:26.10 | Qwell[] | yes, a pretty case makes it run MUCH better |
| 20:26.11 | zoa | the fonality thing looks like just a normal pc with the normal software |
| 20:26.14 | zoa | it does look nice though |
| 20:26.15 | zoa | :) |
| 20:26.20 | Qwell[] | zoa: That is all it is :) |
| 20:27.36 | Qwell[] | besides, ours is much nicer looking :p |
| 20:27.49 | Qwell[] | oh, and it doesn't take up a whole 2U... |
| 20:28.27 | aptura | Qwell talking about the appliance? |
| 20:29.00 | *** join/#asterisk spanglesontoast (n=edd@eddland.plus.com) |
| 20:29.18 | spanglesontoast | hey does anyone know much about the password and user format for asterisk |
| 20:31.43 | Qwell[] | spanglesontoast: What about it? |
| 20:33.11 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) |
| 20:33.36 | tzafrir | spanglesontoast, password and user for what exactly? sip? voicemail? |
| 20:33.37 | IOscanner | I am looking to add 2-4 PRI lines into an Asterisk server |
| 20:33.44 | spanglesontoast | well just the manager password |
| 20:33.51 | spanglesontoast | wanting to use gastman on asterisk |
| 20:33.54 | IOscanner | How much of oa difference would the DSP make? |
| 20:34.01 | tzafrir | spanglesontoast, have you looked at the smaple config files? |
| 20:34.04 | IOscanner | is it worth the cost difference? |
| 20:35.35 | spanglesontoast | erm didn't know there was one for the manager.conf |
| 20:35.59 | tzafrir | look in configs/ in the source directory |
| 20:36.26 | tzafrir | There is a sample there for everything except asterisk.conf , I believe |
| 20:36.35 | aptura | what is the general rule of cpu utilization for a g711u duplex call per channel? |
| 20:37.47 | spanglesontoast | nope no configs directory |
| 20:38.18 | tzafrir | spanglesontoast, what version of Asterisk do you use? from source or from a package? |
| 20:38.27 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:38.29 | spanglesontoast | ubuntu package... |
| 20:38.31 | tzafrir | aptura, no rule3 |
| 20:38.40 | spanglesontoast | I guess I could compile it though... |
| 20:38.58 | spanglesontoast | find it kinda yucky and more hassle though when it could be a few lines |
| 20:39.01 | IOscanner | I think they have a separate package for the configs |
| 20:39.12 | IOscanner | debian does the same thing |
| 20:39.23 | tzafrir | aptura, it highly depends on your usage pattern. Other things (conferences, echo canceliing, transcoding) may end up consuming much more CPU |
| 20:39.42 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
| 20:39.48 | tzafrir | IOscanner, which version of the debian package? |
| 20:40.09 | tzafrir | in later versions it is in /usr/share/doc/asterisk/examples/configs |
| 20:40.17 | tzafrir | in some earlier ones: |
| 20:40.48 | spanglesontoast | yea they are there |
| 20:40.48 | spanglesontoast | ;) |
| 20:41.00 | tzafrir | <PROTECTED> |
| 20:41.06 | `p4r14h`work | asterisk is sendout voicemails to email addresses fine, but it keeps sending the message and attachment in plain text. is there anyone who has ran into this before? |
| 20:41.31 | aptura | tzafrir say I am putting together a low power embeded solution for a small office and need to know the limits of a soekris board. There boards cpu requirments only go to 266 mhz so need to calculate what is the maximum cpu utilazation for g711u |
| 20:41.32 | diclophis-work | ok i have a new one |
| 20:41.37 | diclophis-work | what does this mean: "SIP message could not be handled, bad request: 171225110485c51c005bb810368cc3b1@192.168.55.175 " |
| 20:41.47 | Qwell[] | aptura: Why not test it yourself? |
| 20:41.48 | diclophis-work | and why does it show up if i have sip debuging disabled? |
| 20:42.05 | tzafrir | aptura, test. Get a stonger system and bombard your system with SIP calls |
| 20:42.50 | aptura | sipp generator if it works well |
| 20:43.16 | tzafrir | aptura, likewise another Asterisk |
| 20:44.14 | MACscr | whats the best format for MoH or any audio prompts in asterisk? |
| 20:44.26 | Qwell[] | MACscr: whatever your endpoints are using |
| 20:44.49 | aptura | the board contains one mini pci slow. Is there a adapter to convert the digium/sangoma pci card. |
| 20:45.34 | *** join/#asterisk dasenjo (n=dasenjo@190.24.176.87) |
| 20:46.51 | spanglesontoast | hmm gastman doesn't work anyone know of some working web interface I can get my hands so i can learn asterisk easier |
| 20:47.08 | Hmmhesays | web interfaces arent' going to help you learn the basics easier |
| 20:47.39 | spanglesontoast | yea but I wanna get it setup quickly :) |
| 20:48.05 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 20:49.29 | *** join/#asterisk dj-fu (n=ajc@unaffiliated/dj-fu) |
| 20:51.44 | [hC] | so, i have a queue set up, and i am not passing the option "h" or "H" (to allow the caller/callee to hang up by pressing *) yet if the agent presses *, the call hangs up. What am i missing? |
| 20:51.57 | [hC] | I have specified in features.conf to allow * to be used for attended transfer. |
| 20:52.02 | [hC] | this obviously does not work out |
| 20:54.32 | anonymouz666 | all afternoon fighting with dialplan things seems to work now, my braincells are dead this moment |
| 20:54.32 | anonymouz666 | lol |
| 20:54.32 | `p4r14h`work | anyone have troubles with voicemail emails getting sent as plain text |
| 20:54.41 | spanglesontoast | wow it works |
| 20:55.43 | Strom_M | hmm |
| 20:55.45 | *** join/#asterisk Dr-Linux|home (n=dont@DSL-202-59-73-131.nexlinx.net.pk) |
| 20:55.50 | Strom_M | zaphfc.ko isn't seeing the b410p card |
| 20:55.53 | IOscanner | tzafrir:I am not sure. I build my own. I installed it once and remember it not having configs |
| 20:56.04 | IOscanner | I am looking to add 2-4 PRI lines into an Asterisk server |
| 20:56.18 | IOscanner | How much of oa difference would the DSP make? |
| 20:57.03 | tzafrir | Strom_M, qozap may be able to see a b410p (maybe after some massaging, haven't tried) |
| 20:57.09 | anonymouz666 | Strom_M: the card is connected to the motherboard? :D |
| 20:57.15 | tzafrir | zaphfc is for hfc-s, not for hfc-4s |
| 20:57.20 | Strom_M | tzafrir: ah, ok |
| 20:57.44 | *** join/#asterisk shodan- (n=shodan@ip179.96-113-216.pppoe1.joliette.intermonde.net) |
| 20:58.09 | Strom_M | i'll give that a shot shortly |
| 21:04.45 | *** join/#asterisk CunningPike_ (n=CunningP@204.239.12.189) |
| 21:06.04 | Strom_M | qozap: no multibri cards found |
| 21:06.14 | Strom_M | curses |
| 21:07.36 | *** join/#asterisk jans0n (n=janson@1-1-2-2a.gam.gbg.bostream.se) |
| 21:07.57 | tzafrir | Strom_M, well, qozap.c has a small table of PCI IDs. But I figure that a few other minor changes will be needed |
| 21:08.24 | Strom_M | what kinds of changes might I need to make? |
| 21:08.30 | Strom_M | add the pci id of the b410p card? |
| 21:10.31 | jans0n | Hmm this is so weird. i have a phone running sip and a client on my computer. Asterisk to Phone, works great. Phone to Software Client works great. Software Client calling Phone doesn't work, the phone never rings.. messages are "is busy/congested" and "status is 'CONGESTION'" |
| 21:11.03 | jans0n | if making a console dial works to that phone, and from that phone to other devices.. then i cannot see why calling the phone wont work |
| 21:11.25 | *** join/#asterisk saftsack (n=oliver@pD9E0719F.dip.t-dialin.net) |
| 21:11.59 | anonymouz666 | var=1,2,3,4,5,6 - whats the best way to parse this values to do for example saynumber(1), saynumber(2) etc. |
| 21:12.00 | Strom_M | jans0n: pastebin the relevant sections of extensions.cof and sip.conf and some console output |
| 21:12.04 | jans0n | Strom_M: ok, will do |
| 21:12.16 | jans0n | ~pastebin |
| 21:12.19 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 21:12.41 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 21:16.00 | *** part/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 21:17.11 | anonymouz666 | whats the differecen between a app_cut and function cut ? |
| 21:17.50 | delmar | Strom_M, so what did you think of my problem with the 'hold' being received all the way from the other customer gear ? |
| 21:18.31 | spanglesontoast | is there a test number to see if I'm logged in ? |
| 21:20.12 | *** join/#asterisk FarrisG (n=lckirk@gateway.wiquest.com) |
| 21:22.04 | Strom_M | delmar: i havent given it much thought |
| 21:22.18 | Strom_M | anonymouz666: app_cut is deprecated |
| 21:23.33 | anonymouz666 | after cutting the values and setting into a new var i still dont know how to use saynumber with these values |
| 21:23.35 | anonymouz666 | lol |
| 21:25.10 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 21:25.59 | *** join/#asterisk clive- (n=pirch@dsl-242-176-228.telkomadsl.co.za) |
| 21:26.20 | *** join/#asterisk delphiuk (n=Richard@87.252.35.61) |
| 21:26.39 | *** join/#asterisk Meaty` (n=meaty3@office.abi.ca) |
| 21:27.20 | delphiuk | could someone help me with setting up some king of email transport for my voicemail.conf for sending attachments on an Ubuntu server please? the sendmail -t command does not appear to be working |
| 21:29.14 | Voice2 | heard theres another DOS ona sterisk |
| 21:29.30 | *** join/#asterisk BigCanOfTuna (n=arustad@dsl-mac-66-18-226-119-cgy.nucleus.com) |
| 21:31.20 | BigCanOfTuna | In my dial plan, I'd like to call a script via the system(). Anyone know how I would set the SYSTEMSTATUS in a Ruby script? |
| 21:31.40 | Corydon-w | exit(status) |
| 21:32.41 | anonymouz666 | Set(var=1,2,3,4) anyone has an idea (just an idea) on how I could cut this values and use SayNumber() on these values? |
| 21:33.25 | *** join/#asterisk tkowal (n=nospamto@74.93.82.14) |
| 21:33.44 | Corydon-w | anonymouz666: that won't work |
| 21:33.56 | Corydon-w | You need: Set(var=1\,2\,3\,4) |
| 21:34.05 | *** part/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-171-251.ny325.east.verizon.net) |
| 21:34.31 | Corydon-w | and then you need SayNumber(${CUT(var,\,,1)}) |
| 21:34.32 | anonymouz666 | ok, but how do I split the values into variables? |
| 21:34.52 | anonymouz666 | hmm ok |
| 21:34.56 | Corydon-w | and SayNumber(${CUT(var,\,,2)}) |
| 21:35.03 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 21:35.13 | spanglesontoast | hmm why won't my extensions work |
| 21:35.19 | anonymouz666 | but, and if today I got 4 values, and tomorrow 10 values |
| 21:35.36 | anonymouz666 | it's not fixed length |
| 21:35.52 | Corydon-w | You can even embed another variable inside: Set(i=1) ... ${CUT(var,\,,${i})} |
| 21:35.56 | ars247 | Anyone know what would be the cause if, i put a caller on Hold and I get disconnected with call but didnt get any BYE request |
| 21:36.04 | Corydon-w | anonymouz666: you want FIELDQTY, then |
| 21:36.55 | anonymouz666 | fieldqty ? |
| 21:37.01 | anonymouz666 | i am sorry i don't know that |
| 21:37.21 | Corydon-w | Set(max=${FIELDQTY(var,\,)}) |
| 21:37.38 | Corydon-w | tells you how many fields are inside var, as delimited by the comma |
| 21:38.20 | jans0n | http://pastebin.ca/405665 |
| 21:38.33 | anonymouz666 | and then I use max var inside saynumber as you said ? |
| 21:38.37 | Corydon-w | So you create an iterative loop |
| 21:38.47 | jans0n | i cannot understand why i can't make an internal call to the e65 user, though a call from pstn to e65 user works fine. |
| 21:39.03 | Corydon-w | anonymouz666: are you a programmer? |
| 21:39.17 | anonymouz666 | not really :) |
| 21:39.33 | Corydon-w | anonymouz666: go find a programmer. He or she can explain this better |
| 21:39.39 | wwalker | I want to replace my Netrake nCite SBCs. I want to do so by May 1. Would you 1) use OpenSER 1.2.0 2) use OpenSER 1.1.1 3) stick with the nCite or 4) other? |
| 21:40.02 | anonymouz666 | Corydon-w: I got the point |
| 21:40.28 | ars247 | Anyone know what would be the cause if, i put a caller on Hold and I get disconnected with call but didnt get any BYE request but i get this -- Got SIP response 481 "No Such Call" back from 192.168.1.24 |
| 21:41.39 | Corydon-w | jans0n: it's probably not registered |
| 21:42.10 | jans0n | Corydon-w: show peers says its registered. |
| 21:42.14 | jans0n | and.. |
| 21:42.28 | jans0n | if it werent registered then the call from pstn should't come trough |
| 21:42.37 | jans0n | so, i guess that's not the problem :/ |
| 21:42.49 | anonymouz666 | I learned today a very useful command "show functions" |
| 21:42.54 | anonymouz666 | :d |
| 21:42.57 | jans0n | Corydon-w: ow, take a look at http://pastebin.ca/405665 |
| 21:44.10 | jans0n | console dial to the e65 users works great. |
| 21:44.21 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 21:45.37 | jans0n | but calling it with the client doesn't.. |
| 21:45.37 | jans0n | SIP/e65-081fe790 is circuit-busy |
| 21:46.37 | Corydon-w | jans0n: auto_congest means that the SIP peer did not respond in time, for whatever reason |
| 21:46.52 | jans0n | Corydon-w: respond to asterisk that is? |
| 21:47.04 | Corydon-w | respond to the INVITE asterisk sent, yes |
| 21:47.11 | jans0n | aha ok |
| 21:47.39 | jans0n | Corydon-w: is there a difference between how that is done when i do a console dial to the user or dialing it using a software? |
| 21:48.18 | Corydon-w | There are minor differences in the headers, yes |
| 21:48.35 | jans0n | ok |
| 21:48.43 | Corydon-w | callerid, specifically |
| 21:48.58 | jans0n | hmm, i'm thinking of the best way to debug this |
| 21:49.01 | Corydon-w | One thing it might be is that the e65 might not like the '@' in the callerid |
| 21:49.11 | Corydon-w | Run a 'sip debug' |
| 21:49.12 | jans0n | Corydon-w: ow, may be it.. |
| 21:49.43 | Corydon-w | Different clients have different expectations |
| 21:49.45 | jans0n | yeah, i will see if that gives me any better info |
| 21:50.06 | jans0n | Corydon-w: i will check to see if thats the case, with the @ char |
| 21:50.21 | jans0n | because pstn works, and theres no @ char in that callid.. |
| 21:51.12 | Corydon-w | Correct. That would tend to be the major difference |
| 21:53.26 | *** join/#asterisk burd (n=burd@71-210-51-58.hlna.qwest.net) |
| 21:54.48 | burd | I'm basicly a beginner to this stuff but I have a spa3102 set up and working with inward and outward dialing.. my problem is I want the inward dialing to use an asterisk IVR I created but I don't know how to direct the calls, anyone have any experience doing this? |
| 21:55.50 | [jwb] | gah, I've been on hold with digium now for 20-30min |
| 21:55.54 | jans0n | omg Corydon-w |
| 21:56.13 | jans0n | Corydon-w: i'm ashamed i couldnt figure it out myself :) |
| 21:56.30 | jans0n | Corydon-w: big thanks! works like a charm. |
| 21:56.36 | Corydon-w | yw |
| 21:57.51 | *** join/#asterisk poppo (n=adas@S01060050ba23deca.ed.shawcable.net) |
| 21:58.03 | poppo | I am having problems installing cepstral can somebody help me out? |
| 21:58.48 | *** join/#asterisk CunningPike_ (n=CunningP@204.239.12.189) |
| 21:59.08 | Corydon-w | poppo: please request assistance at www.cepstral.com/support/ |
| 21:59.40 | poppo | oj |
| 22:02.16 | Mercestes | diclophis-work, Did you ever get your thing fixed?? |
| 22:02.21 | *** part/#asterisk MarkWD (n=MarkWD@rrcs-67-78-88-186.sw.biz.rr.com) |
| 22:02.34 | diclophis-work | actually i did manage to get through to that number through the system |
| 22:02.44 | diclophis-work | still don't know what caused all the congestion though |
| 22:02.58 | diclophis-work | maybe there is a dying switch somewhere in the telcos system? |
| 22:03.09 | Mercestes | diclophis-work, A very good possibility. |
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| 22:14.44 | *** join/#asterisk Ng (n=cmsj@mairukipa.tenshu.net) |
| 22:16.38 | Ng | Is there a faq or wiki page or suggestion for a good way of making a given DDI number ring at several phones? I would call that a hunt group, but googling for that I'm finding more stuff about making groups of Zap channels |
| 22:17.05 | Qwell[] | Dial(SIP/6001&SIP/6002) |
| 22:17.23 | Qwell[] | then whoever answers gets the call |
| 22:17.58 | Ng | Qwell[]: I tried that, but if 6002 didn't exist then it failed to produce a remote ringing noise. I ran out of time onsite to test how it would work if 6002 was in the config, but just not registered |
| 22:18.17 | Ng | but I'll be back on it tomorrow (no internet installed yet, annoyingly) |
| 22:18.22 | Mercestes | Ng: Yo ushouldn't try to ring devices that don't exist. |
| 22:18.42 | Ng | that's fair enough :) |
| 22:18.56 | Ng | if the device exists in sip.conf, but just isn't registered will it work correctly? |
| 22:19.06 | spanglesontoast | hmm |
| 22:19.14 | Mercestes | Ng: The syntax is Dial(sip/peer&sip/peer) not dial(sip/peer&sip/some_imaginary_ascii_string) |
| 22:19.21 | Mercestes | Ng: Should. |
| 22:19.23 | spanglesontoast | is there anything needed for the mp3 player ? |
| 22:19.43 | spanglesontoast | is it mpg123 ? |
| 22:19.44 | Cheapneasy | Ng: If you want to set up a hunt group but can't find any help from google, search for Asterisk Queue (a queue and hunt group are the same thing, sort of). Also, I've used the www.voip-info.org website for info - it's a great wiki for asterisk. |
| 22:19.52 | Mercestes | Ng: Everything I've read/seen/done/experienced indicates a yes. Ifyou have a no then please let someone know so they can make it a yes. |
| 22:20.08 | Ng | Mercestes: I like that answer :) |
| 22:20.13 | Mercestes | spanglesontoast, mp3 support is embedded now. |
| 22:20.17 | *** join/#asterisk Fieldy (i=sZg2lCDe@gentoo/contributor/Fieldy) |
| 22:20.19 | Mercestes | Ng: ;) |
| 22:20.21 | Ng | I'm just trying to cover my bases tomorrow so I don't have to trudge off to an internet cafe ;) |
| 22:20.32 | Mercestes | yea. |
| 22:20.34 | spanglesontoast | well I got a poll timed out error |
| 22:20.51 | Mercestes | spanglesontoast, Were you asking about playing Mp3s? |
| 22:21.05 | Strom_M | What are the specific differences between EuroISDN and National ISDN? |
| 22:21.06 | Ng | I have to say I'm seriously liking the pattern matching stuff |
| 22:21.15 | flenders | Ng: if you ring a device that doesn't exist, you'll get a message like this: |
| 22:21.16 | flenders | [Mar 21 17:29:20] WARNING[4645]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 22:21.21 | spanglesontoast | yea must be an old version of asterisk |
| 22:21.25 | spanglesontoast | had to install mpg123 |
| 22:21.26 | Mercestes | Ng: Isn't that t3h 133t? I love it too. |
| 22:21.31 | flenders | but the other devices will ring the way they should |
| 22:21.35 | Mercestes | spanglesontoast, Yea, likely. |
| 22:21.40 | Ng | Strom_M: I'm wondering that too, I suspect national isdn means use the settings for your loadzone country, but that's just a guess |
| 22:21.51 | spanglesontoast | hmm |
| 22:21.55 | Ng | I'm using a BT ISDN line which ought to be euroisdn, but it works with national isdn and loadzone=uk |
| 22:22.05 | spanglesontoast | what about conferencing it seems to kick me out as soon as I jump in |
| 22:22.07 | Mercestes | flenders, That's if the device is configured but offline. These devices he's trying to rign don't even exist in sip.conf. |
| 22:22.19 | Strom_M | Ng: what ISDN equipment and driver are you using? |
| 22:22.32 | Ng | Mercestes: it gets me a hundred phones registering and attached to DDIs with about two lines of config and a couple of macros :D |
| 22:22.34 | Mercestes | spanglesontoast, Use zt dummy |
| 22:22.40 | spanglesontoast | zt ? |
| 22:22.46 | Mercestes | spanglesontoast, ztdummy |
| 22:22.56 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 22:23.12 | Ng | Strom_M: a BT ISDN30, plugged into a sangoma A102d, with sangoma's current stable driver on Ubuntu Edgy (I forget which exact version of asterisk that implies) |
| 22:23.24 | Mercestes | spanglesontoast, emerge zaptel...Oh yea! your probably not using gentoo. In that case you have to download the src, put it in /usr/src, untar it, cd into zaptel_?.?.? ./configure make clean make install and hope for the best. |
| 22:23.24 | Strom_M | oh, is it a PRI? |
| 22:23.29 | Ng | Strom_M: yeah |
| 22:23.32 | Strom_M | ah ok |
| 22:23.32 | flenders | Mercestes: then you'll get a: |
| 22:23.35 | flenders | [Mar 22 09:23:16] WARNING[29947]: chan_sip.c:2719 create_addr: No such host: blah |
| 22:23.38 | spanglesontoast | ah |
| 22:23.46 | spanglesontoast | wondered why it kept saying zaptel |
| 22:23.49 | Mercestes | flenders, Does it ring back under multiple deviecs tho? |
| 22:23.55 | spanglesontoast | it did say this isn't a valid conference number |
| 22:24.08 | Ng | Strom_M: aren't the interface cards the same? |
| 22:24.09 | flenders | Mercestes: yes |
| 22:24.13 | Mercestes | Ng: Your behavior might be worth mentioning in a bug report because I kinda feel that if you have *any* valid devices then ring through should continue. |
| 22:24.35 | JT | Strom_M: what's this silly buggers with a b410p? :P |
| 22:24.51 | Mercestes | spanglesontoast, Conferences require a timing source. Ala zaptel |
| 22:24.52 | Ng | Mercestes: yeah. It did make the real devices ring, but the caller never got a ring signal and answering didn't bridge them. I admit it's a stupid thing to do, but failing gracefully is always good |
| 22:25.00 | spanglesontoast | ah |
| 22:25.05 | spanglesontoast | installing now :D |
| 22:25.10 | Ng | it was a quick test because I was out of time ;) |
| 22:25.22 | Mercestes | Ng: I'm with you there. |
| 22:25.42 | Mercestes | spanglesontoast, I always default to using Ztdummy just in case. |
| 22:26.08 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 22:27.25 | spanglesontoast | hmm it says no zaptel transcoders |
| 22:27.34 | Mercestes | spanglesontoast, You have to uncomment ztdummy somewhere. |
| 22:27.35 | lokkju_wrk | anyone know if the sipura 3000 or 3102 is based on GPL code? |
| 22:27.42 | spanglesontoast | ok |
| 22:27.45 | Mercestes | spanglesontoast, google asterisk wiki ztdummy I think |
| 22:27.50 | Mercestes | in the makefile i believe. |
| 22:28.07 | Mercestes | Ng: YOu *could* have one way audio... |
| 22:28.13 | lokkju_wrk | (I'm trying to find a FXO external device that is running a linux kernal - want to try a custom app on it, have to be able to hack in a serial port) |
| 22:28.18 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-177-150-211.lsanca.fios.verizon.net) |
| 22:29.04 | xezz | http://pastebin.ca/405762, any help appreciated -.- |
| 22:29.06 | Voice2 | can an exten have dashes ? like exten -> _809XX-XXX-XX, |
| 22:29.22 | Mercestes | Voice2, Only if yo uare dialing dashes. |
| 22:29.30 | Voice2 | <PROTECTED> |
| 22:29.38 | Mercestes | Voice2, Or you are goto(809XX-XXX-XX,1) |
| 22:29.39 | Voice2 | i cant hint 1000 since multiple people have that |
| 22:29.43 | Voice2 | yes im gotoing |
| 22:29.57 | Mercestes | hint individually. |
| 22:30.00 | Voice2 | i cant hint 1000 since multiple people have that |
| 22:30.06 | Mercestes | works for goto then but not as a number. |
| 22:30.18 | Mercestes | you have to hint individually. |
| 22:30.29 | Voice2 | darn copy paste brke |
| 22:30.38 | Voice2 | i use goto (Accountcode-exten) then in sql i have exten as accountcode-exten , dia sip/blah |
| 22:30.38 | spanglesontoast | brb ciggy |
| 22:30.50 | [TK]D-Fender | lokkju_wrk, Nope, all proprietary. |
| 22:31.15 | Voice2 | but if i exten , 12345-1000,hintmsip/blah |
| 22:31.31 | lokkju_wrk | [TK]D-Fender, so, other then the SPA-400, any ideas? |
| 22:31.32 | Voice2 | doesnt the subscribing phone need to dial that ? |
| 22:31.50 | DocHolliday | sup [TK]D-Fender |
| 22:32.06 | *** part/#asterisk Cheapneasy (n=chatzill@joshie355.plus.com) |
| 22:32.06 | [TK]D-Fender | lokkju_wrk, Nope. |
| 22:32.16 | [TK]D-Fender | DocHolliday, Tired. Long day, long week... |
| 22:32.18 | lokkju_wrk | hmf, gotta be some linux based ones out there |
| 22:32.38 | [TK]D-Fender | lokkju_wrk, No there doesn't |
| 22:32.50 | DocHolliday | [TK]D-Fender, well you survived hump day so the rest is all down hill |
| 22:32.57 | [TK]D-Fender | ... |
| 22:33.13 | DocHolliday | wednesday = middle of the week = hump day |
| 22:34.22 | [TK]D-Fender | xezz, You either don't have a group 2 or it has no free channels |
| 22:34.38 | [TK]D-Fender | DocHolliday, Yes I am fully aware of the term... its ADDED pessimism... |
| 22:35.48 | *** join/#asterisk deeperror (n=deeperro@ppp-69-215-66-176.dsl.sfldmi.ameritech.net) |
| 22:35.54 | DocHolliday | http://www.urbandictionary.com/define.php?term=hump+day |
| 22:36.04 | deeperror | heard that |
| 22:37.51 | DocHolliday | [TK]D-Fender, heh. the glass is always half full with me |
| 22:38.38 | [TK]D-Fender | DocHolliday, I don't leave glasses half-full. Waste not, want not :) |
| 22:40.41 | DocHolliday | heh thats good, need to remember that |
| 22:45.44 | xezz | [TK]D-Fender i have group 2 |
| 22:45.44 | *** join/#asterisk MrTelephone (n=DeaLER25@bas13-toronto63-1242371764.dsl.bell.ca) |
| 22:45.47 | xezz | and all channels are free, its 1:00 am all channels free -.- |
| 22:45.49 | MrTelephone | who here has an adit 600? |
| 22:46.48 | JT | i combined those lines somewhere between my eyes and my brain as saying "anyone want a free adit 600" my mind can hope ;) |
| 22:47.00 | [TK]D-Fender | xezz, pastebin your zaptel. & zapata.conf, and turn up the verbose to 10 on CLI and try again |
| 22:47.07 | [TK]D-Fender | and pastebin that as well |
| 22:47.10 | MrTelephone | i just don't understand from the pictures where the fxs lines come out? |
| 22:47.18 | xezz | okie w8 a second |
| 22:47.23 | MrTelephone | if you insert an fxs card it looks like a little thing with 8 lights |
| 22:49.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:49.37 | spanglesontoast | hmm |
| 22:50.00 | spanglesontoast | is it easy to bridge sip clients ? |
| 22:51.25 | xezz | [TK]D-Fender, http://pastebin.ca/405789 |
| 22:52.02 | spanglesontoast | is that a pear ? |
| 22:52.38 | [TK]D-Fender | sounds fruity to me.... |
| 22:53.01 | xezz | ? :L |
| 22:54.39 | [TK]D-Fender | xezz, Aside from the blatantly obvious "we don't support FreePBX here" statement, any reason you felt I didn't need to see the files INCLUDED by zapata.conf? |
| 22:54.40 | MrTelephone | on the back of an adit 600 is an amphenol cable? |
| 22:54.51 | MrTelephone | i mean amphenol connector? |
| 22:55.13 | `p4r14h`work | could anyone help me with this particular issue? i have an x100p hooked up to a panasonic PBX on ext 205. i have a rule set to forward all unmatched calles on the interface to ring SIP user 600. when somone calls in to the PSTN that is hooked to the panasonic it rings the SIP 600 phone, but the CID shows up as asterisk. is there any way to get the PSTN CID to show up on the SIP phone |
| 22:55.15 | `p4r14h`work | ? |
| 22:56.07 | xezz | soz you are right, but that prob disturbs me long time... |
| 22:57.43 | xezz | thats the zapata-BIR-STUFF included : http://pastebin.ca/405795 |
| 22:58.37 | *** join/#asterisk oQPa (n=uawename@33.Red-83-34-60.dynamicIP.rima-tde.net) |
| 23:00.51 | DocHolliday | [TK]D-Fender, still around? |
| 23:01.21 | JT | xezz: start from scratch |
| 23:01.29 | JT | xezz: all those configs are a complete mess |
| 23:01.33 | JT | download bristuff |
| 23:01.43 | JT | and run the script that installs it |
| 23:01.56 | JT | if you want to save a lot of trouble anyway |
| 23:02.01 | [TK]D-Fender | xezz, and why is this example using group 0? Your first was group 1. you keep showing non-matching bits & pieces. |
| 23:02.13 | JT | group=0,1 wtf |
| 23:02.17 | xezz | well, let me explait |
| 23:02.18 | [TK]D-Fender | xezz, And this call DID progress and was regected for different reasons |
| 23:02.29 | [TK]D-Fender | JT : Multiple groups. its allowed |
| 23:02.47 | JT | oh ok |
| 23:02.50 | xezz | i am able to make calls everywhere except internationals |
| 23:02.56 | JT | zaptel.conf is wrong |
| 23:02.58 | [TK]D-Fender | `p4r14h`work, X100P have been notorious for having crappy CID ddetection. |
| 23:03.02 | xezz | i can accept calls from everywhere include internationals |
| 23:03.32 | xezz | that cli output is a try to make an international call, i get the 'all busy' message |
| 23:03.35 | *** join/#asterisk ChrisHardie (n=silas@frigga.summersault.com) |
| 23:03.39 | [TK]D-Fender | xezz, you are using 2 groups and exhibiting 2 completely different issues |
| 23:03.48 | JT | it's amazing that even works |
| 23:03.51 | xezz | i can call mobiles local etc everything , just international calls wont work |
| 23:03.57 | JT | it looks like a bastard child of freepbx and bristuff |
| 23:04.01 | JT | i have no idea what you did |
| 23:04.02 | *** join/#asterisk qdk (n=qdk@80.243.125.204) |
| 23:04.19 | ChrisHardie | Is this a good place to ask about an issue in upgrading from 1.2.x to 1.4.x? |
| 23:04.35 | [TK]D-Fender | ChrisHardie, Quite likely. |
| 23:04.41 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 23:04.58 | ChrisHardie | Well, I just did it (upgraded) and everything seems to work fine...except that the system will no longer answer calls on the Digium line ports. |
| 23:05.03 | [TK]D-Fender | DocHolliday, yes? |
| 23:05.23 | ChrisHardie | I confirmed that the zaptel modules were being loaded correctly. Was wondering if there's anything new in the config files that I need to be looking at. |
| 23:05.32 | *** part/#asterisk tg (i=tg@x-net.hu) |
| 23:05.34 | [TK]D-Fender | ChrisHardie, Did you upgrade zaptel as well and enable it in your * ./configure? |
| 23:05.54 | ChrisHardie | I didn't pass anything to ./configure, so it's just the vanilla configure. |
| 23:06.14 | [TK]D-Fender | ChrisHardie, There make be something to check off in there to ensure that * knows about zaptel |
| 23:06.53 | ChrisHardie | I see that there's a "--with-zaptel=PATH" command, but assumed that it would find/did find those files on its own if they were in a standard location. |
| 23:07.45 | [TK]D-Fender | xezz, Channel 0/2, span 1 received AOC-E charging 0 units |
| 23:08.10 | [TK]D-Fender | xezz, AOC = Advice of Charge. I'm wondering if you're allowed to make that call.... |
| 23:08.17 | DocHolliday | [TK]D-Fender, if a customer has a TDM PBX aka Nortel, and you want to route their calls over SIP, whats the best way to achieve that? (in terms of hardware) |
| 23:08.18 | DocHolliday | without redoing their entire phone system |
| 23:08.34 | JT | [TK]D-Fender: what's unusual about those lines? |
| 23:08.55 | Dr-Linux|home | hi |
| 23:08.56 | [TK]D-Fender | DocHolliday, How many lines do they have? ALL calls? |
| 23:09.20 | Dr-Linux|home | http://portal.mmasson.com/asterisk/files/pager2sms-0.2.agi , |
| 23:09.36 | DocHolliday | [TK]D-Fender, for the sake of argument 1 PRI |
| 23:09.37 | Dr-Linux|home | who would be sender? and how will type letters, i mean messages :S |
| 23:10.31 | [TK]D-Fender | DocHolliday, So they have a PRI and you want to move them completely to VoIP? |
| 23:10.37 | xezz | [TK]D-Fender from the standard pstn line i am able to call the same number with same prefix normally |
| 23:11.11 | DocHolliday | correct |
| 23:11.39 | JT | [TK]D-Fender: i don't think there's anything wrong with those AOC lines |
| 23:11.49 | [TK]D-Fender | DocHolliday, Well at that point I'd say get a PRI card for the * box and thats it. |
| 23:12.06 | [TK]D-Fender | DocHolliday, What more is there to say? |
| 23:13.41 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 23:13.43 | JT | <PROTECTED> |
| 23:13.54 | DocHolliday | [TK]D-Fender, so install an asterisk box and interconnect the PRI interface on the TDM PBX to the one on the asterisk box? |
| 23:14.03 | [TK]D-Fender | # |
| 23:14.03 | [TK]D-Fender | -- Channel 0/2, span 1 got hangup request |
| 23:14.04 | [TK]D-Fender | # |
| 23:14.04 | [TK]D-Fender | <PROTECTED> |
| 23:14.11 | JT | messages like that usually indicate you are not subscribed to the AOC service from your telco |
| 23:14.14 | [TK]D-Fender | JT : back-to-back |
| 23:14.24 | JT | therefore it doesn't say how much the charge was |
| 23:14.35 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 23:14.47 | JT | [TK]D-Fender: i always get AOC messages at the end of every digital call to telco |
| 23:14.50 | [TK]D-Fender | DocHolliday, Ummm... yeah... what were YOU thinking? ;) |
| 23:15.07 | JT | maybe only bristuff adds them to the cli :) |
| 23:15.21 | ChrisHardie | [TK]D-Fender, I recompiled with --with-zaptel and there's no change in behavior - still won't pick up the line. |
| 23:15.25 | *** join/#asterisk HockeyInJune (i=HockeyIn@pool-68-161-171-251.ny325.east.verizon.net) |
| 23:15.36 | [TK]D-Fender | ChrisHardie, can you dial out? |
| 23:15.54 | JT | AOC advises you of the charge, not if you are permitted to make the call |
| 23:16.05 | ChrisHardie | [TK]D-Fender: No..." channel.c:3024 ast_request: No channel type registered for 'Zap'" |
| 23:16.08 | DocHolliday | [TK]D-Fender, do you think asterisk is most suitable for playing the 'media converter role' as opposed to say a Cisco 28xx? |
| 23:16.46 | DocHolliday | in terms of reliability |
| 23:16.52 | [TK]D-Fender | ChrisHardie, try "load chan_zap.so" |
| 23:17.01 | ChrisHardie | [TK]D-Fender: Tried that..." /usr/lib/asterisk/modules/chan_zap.so: cannot open shared object file: No such file or directory" |
| 23:17.08 | [TK]D-Fender | DocHolliday, Does the job, and give you more control. I don't know how smart you can make a Cisco for what you want. |
| 23:17.29 | ChrisHardie | So I guess it didn't get built after all... |
| 23:18.19 | DocHolliday | [TK]D-Fender, gotcha.. just worried about a linux box vs hardened cisco voice router |
| 23:18.43 | JT | hardened usually refers to stuff designed with security in mind |
| 23:18.47 | JT | ciscos aren't that secure |
| 23:20.20 | [TK]D-Fender | DocHolliday, Your question are sounding remarkably rhetorical...... |
| 23:20.21 | ChrisHardie | [TK]D-Fender: I see references to chan_zap.c in the menuselect code - was I supposed to be prompted at some point to compile in zaptel support? |
| 23:20.44 | [TK]D-Fender | ChrisHardie, I believe so, but don't have the personal experience to corroborate it |
| 23:21.51 | file | configure detects the presence of zaptel and whether it has the capabilities in order to build chan_zap, if it doesn't then chan_zap won't get built and won't be selectable in menuselect |
| 23:21.57 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 23:22.10 | file | if you are using Asterisk 1.4.2 with Zaptel 1.4.0 it won't work, you must use Zaptel 1.4 from SVN until Zaptel 1.4.1 is released |
| 23:22.39 | ChrisHardie | file: OH |
| 23:22.39 | MrTelephone | whats good in asterisk 1.4? |
| 23:22.49 | MrTelephone | shared line? |
| 23:23.03 | ChrisHardie | Is there an easy way to get that, or does this mean I'm treading in territory that isn't fit for production use? |
| 23:23.32 | file | svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel |
| 23:23.36 | file | requires subversion to be installed |
| 23:23.39 | DocHolliday | [TK]D-Fender, much appreciated |
| 23:23.54 | file | and no... that branch is what will become zaptel 1.4.1 |
| 23:24.32 | ChrisHardie | file: But you think it's safe to use 1.4.2 in production? |
| 23:24.58 | file | depends on what you are doing, the only way to know is to setup a lab environment to test YOUR usage |
| 23:25.05 | file | everyone does something differently |
| 23:25.37 | Ng | is there a way of checking if a local SIP extension exists? |
| 23:25.37 | ChrisHardie | file: once I build zaptel from that tree, do I just use --with-zaptel in the asterisk compile to point to it? |
| 23:25.51 | file | no, just do ./configure again |
| 23:25.55 | file | after installing it |
| 23:26.18 | ChrisHardie | does that mean I'll need to reboot to have the updated zaptel modules take effect? |
| 23:26.36 | file | unload the current zaptel modules, install, load the new zaptel modules |
| 23:26.49 | ChrisHardie | will try that, thank you. |
| 23:27.01 | JT | rebooting is unnecessary |
| 23:27.16 | ChrisHardie | I've yet to find the magic combo of modprobe -r commands on my ubuntu box. |
| 23:27.22 | ChrisHardie | so that's always worked. lazy, I know. |
| 23:28.09 | file | rmmod zttranscode zaptel |
| 23:28.10 | JT | rmmod |
| 23:28.12 | file | plus whatever other modules |
| 23:28.14 | JT | modprobe |
| 23:28.18 | ChrisHardie | anyway to verify that configure found the zaptel drivers before doing a "make"? |
| 23:28.32 | JT | i wonder if ubuntu has modconf, like debian does |
| 23:28.43 | file | well, doing a make menuselect and hopping to Channel Drivers will show you if the dependencies for chan_zap were met |
| 23:29.01 | file | plus at the end of configure when it looks at the zaptel install you can tell |
| 23:29.06 | ChrisHardie | Yes, they show up now whereas before they were "XXX" out. |
| 23:29.13 | ChrisHardie | Phew. |
| 23:31.16 | *** join/#asterisk spanglesontoast (n=edd@eddland.plus.com) |
| 23:31.19 | spanglesontoast | back |
| 23:31.35 | spanglesontoast | so does anyone know how to bridge a outgoing sip to asterisk |
| 23:32.26 | [TK]D-Fender | spanglesontoast, HUH?!?! |
| 23:32.27 | file | spanglesontoast: rephrase that so it makes more sense |
| 23:32.41 | [TK]D-Fender | <Katty> That does not parse |
| 23:32.43 | spanglesontoast | well I have a sip account but I want to be able to call out from it |
| 23:32.52 | ChrisHardie | file, JT: thanks for all of your help. I'm recompiling now. |
| 23:33.13 | [TK]D-Fender | spanglesontoast, Dial(SIP/yourpeerentryfortheguy/1234567890) |
| 23:33.14 | *** join/#asterisk smk (n=scott@cobra.httpd.org) |
| 23:33.33 | spanglesontoast | oh it has to be a peer ? |
| 23:33.46 | ars247 | Anyone have an idea regarding when I put someone on Hold/Mute it randomly drops calls |
| 23:34.11 | [TK]D-Fender | spanglesontoast, Or Friend |
| 23:34.17 | spanglesontoast | how do I know if it's connected though ? |
| 23:34.42 | [TK]D-Fender | spanglesontoast, there IS NO CONNECTION. It is a set of auth credentials used when you PLACE a call or GET a call. |
| 23:34.51 | [TK]D-Fender | spanglesontoast, there is no "constant cnnection". |
| 23:34.55 | spanglesontoast | ah |
| 23:34.59 | spanglesontoast | what about incoming ones ? |
| 23:35.08 | [TK]D-Fender | spanglesontoast, I already answered that |
| 23:35.23 | [TK]D-Fender | spanglesontoast, with the "GET a call" part |
| 23:35.24 | spanglesontoast | so how can someone call you then ? |
| 23:35.35 | spanglesontoast | ah so the server sends a GET ? |
| 23:36.02 | [TK]D-Fender | ~siprfc |
| 23:36.13 | jbot | siprfc is probably http://www.faqs.org/rfcs/rfc3261.html |
| 23:36.13 | [TK]D-Fender | ~rffcsip |
| 23:36.15 | ars247 | answer mine TK |
| 23:36.15 | ars247 | =( |
| 23:36.16 | [TK]D-Fender | ~rfcsip |
| 23:36.17 | ChrisHardie | JT: now I can't get zaptel to load via modprobe |
| 23:36.17 | spanglesontoast | hmm |
| 23:36.21 | [TK]D-Fender | There, go read |
| 23:36.26 | spanglesontoast | so how do i set a prefix so I can dial through it |
| 23:36.37 | *** join/#asterisk aldoenviro (n=ask@206-174-140-082.static.adsl.evenlink.com) |
| 23:36.45 | file | spanglesontoast: you need to learn the basics of Asterisk configuration and concepts... |
| 23:37.14 | [TK]D-Fender | ~book |
| 23:37.15 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:37.15 | JT | ~thebook |
| 23:37.17 | jbot | extra, extra, read all about it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:37.17 | file | using.... maybe... just maybe... |
| 23:37.18 | bulle | [TK]D-Fender: that syntax, for dial with SIP/peerentry/number can that be used to direct a call via a given sip proxy, if i make a peer entry for the sip proxy ? |
| 23:37.21 | file | :D ^^^ |
| 23:37.26 | ars247 | file do you have any solution regarding my issue? where when i put someone on HOLD/MUTE it drops calls |
| 23:37.28 | DocHolliday | [TK]D-Fender, what would happen if a customer has a TDM PBX but wants to keep their origination and just wants to use us for termination? |
| 23:37.34 | [TK]D-Fender | spanglesontoast, And if that fails |
| 23:37.36 | [TK]D-Fender | ~osmosis |
| 23:37.38 | jbot | osmosis is probably the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 23:37.45 | file | ars247: without a pastebin of console output and debug information, no. |
| 23:37.45 | DocHolliday | are we forced to install a secondary PRI on their existing PBX? |
| 23:37.47 | spanglesontoast | lol |
| 23:37.57 | bulle | [TK]D-Fender: so i can have several sip proxies i make calls going out via, and then have a register entry in my sip.conf to register to those proxies, if i need that |
| 23:38.03 | [TK]D-Fender | DocHolliday, No. |
| 23:38.13 | aldoenviro | Ok, new to asterisk.... looking for some install help |
| 23:38.16 | spanglesontoast | yea but what action would I need |
| 23:38.23 | [TK]D-Fender | bulle, You can, but you do not need to register to place calls. |
| 23:38.31 | JT | spanglesontoast: a pattern match |
| 23:38.40 | JT | spanglesontoast: and a dial command |
| 23:38.49 | [TK]D-Fender | spanglesontoast, You don't understand the basics yet. Go read the book. I haven't set up a consulting seminar yet :) |
| 23:38.50 | bulle | [TK]D-Fender: i know, but for incomming calls that comes in via proxies, eg ekiga.net or similar |
| 23:38.55 | flenders | spanglesontoast: I read the book, and I can guarantee it's all there |
| 23:39.00 | spanglesontoast | so 60X |
| 23:39.11 | [TK]D-Fender | bulle, No, you also don't need to register to receive calls.... |
| 23:39.15 | JT | if the number is 600-609 |
| 23:39.15 | spanglesontoast | then any number after will go to that prefix and dial that number |
| 23:39.17 | aldoenviro | I am getting "The configure script must be executed before running 'make'." Didn't mention anything about configure on the website |
| 23:39.28 | JT | only numbers 600 to 609 |
| 23:39.32 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 23:39.32 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:40.11 | flenders | aldoenviro: have you installed anything from source on linux before? |
| 23:40.15 | ars247 | file: this is all i get from the log when i checked:Mar 21 13:56:22 DEBUG[5393] chan_sip.c: Stopping retransmission on '233f3fcd08353c497578b42502250416@192.168.1.1' of Request\ |
| 23:40.15 | ars247 | <PROTECTED> |
| 23:40.15 | ars247 | Mar 21 13:56:22 VERBOSE[5393] logger.c: -- Got SIP response 481 "No Such Call" back from 192.168.1.24 |
| 23:40.34 | JT | aldoenviro: look for a file called INSTALL or README in the root of the source tree |
| 23:40.42 | *** join/#asterisk k-man_ (n=jason@unaffiliated/k-man) |
| 23:40.44 | k-man_ | hello |
| 23:40.46 | spanglesontoast | what's dtmf mode ? |
| 23:40.47 | k-man_ | jt? |
| 23:40.53 | JT | hi k-man_ |
| 23:40.56 | k-man_ | hi |
| 23:41.11 | k-man_ | sorry about last night - i was totaly on a different planet |
| 23:41.12 | JT | spanglesontoast: also answered by the book |
| 23:41.19 | JT | heh no probs |
| 23:41.20 | spanglesontoast | which online book ? |
| 23:41.22 | k-man_ | i didn't realise the meeting was that night |
| 23:41.32 | JT | spanglesontoast: seriously, we've linked you twice now |
| 23:41.34 | flenders | k-man_: I missed it too |
| 23:41.38 | JT | spanglesontoast: one more time |
| 23:41.38 | JT | ~thebook |
| 23:41.39 | jbot | methinks thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:41.44 | DocHolliday | [TK]D-Fender, what would be my solution in that case? |
| 23:41.46 | flenders | jt told me about it at lunch time |
| 23:42.12 | JT | flenders, k-man_: sign up to the openvoip list so you get the announce for the next one in advance |
| 23:42.20 | ars247 | file: would it have something to do with rtptimeout? |
| 23:42.24 | flenders | JT: done it |
| 23:47.19 | Voice2 | can monitor do mp3.. nativeley ? |
| 23:50.07 | JT | no |
| 23:50.14 | aldoenviro | flenders: Sorry for the delay... reading documentation... No I haven't :( I do tool around with other mechanisms though... I have webmin and can play around in shell |
| 23:50.51 | *** join/#asterisk xxi (i=foobar@cpe-70-113-208-133.austin.res.rr.com) |
| 23:51.57 | *** join/#asterisk dj-fu (n=ajc@unaffiliated/dj-fu) |
| 23:52.54 | flenders | aldoenviro: is there a webmin module for asterisk? |
| 23:53.01 | Voice2 | seems not |
| 23:53.41 | *** join/#asterisk Soul (n=Soul@87-196-4-223.net.novis.pt) |
| 23:53.58 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 23:54.09 | flenders | can't remember last time I used webmin... I wouldn't recommend |
| 23:54.35 | aldoenviro | flenders: Not sure... good point... I will check. I still need to get the app installed though. |
| 23:55.37 | flenders | aldoenviro: as JT said, read the README file |
| 23:56.49 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:56.59 | flenders | aldoenviro: and just a quick tip, pretty much all software (source tarballs) you download, will come with a README or INSTALL file |
| 23:57.44 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |