| 00:00.08 | *** join/#asterisk DrDigi (~mmurphy@50-73-49-97-static.hfc.comcastbusiness.net) |
| 00:12.34 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 00:15.10 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:16.34 | nightwalk | I've encountered an asterisk system that keeps filling /var/log/messages with this kind of junk: "kernel: [118572.902699] netjet.1: ISAC XDU no TX_BUSY". Apparently, everything works as intended, but I have two questions: 1) does this constitute a gross misconfiguration or other "not right" condition? 2) Are there any arguments I can pass to either the netjet module or the misdn module on-load to shut them up? |
| 00:19.47 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 00:30.09 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 00:39.48 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 00:43.47 | tyman | My client wants, badly, an outlook (barf) plugin to originate calls from within outlook contacts. My googling shows many dated & scrappy looking options. The best, from what I can tell, is OutCall. But each OutCall client requires an AMI socket open with which i'm concerned about for security. Is anyone doing this or have a better idea? |
| 00:43.58 | tyman | I does not have to be free/oss... |
| 00:50.38 | *** join/#asterisk adeel (~adeel@24.52.206.104) |
| 00:51.55 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 00:53.56 | *** join/#asterisk dan__t (~dant@72.233.89.95) |
| 00:53.59 | dan__t | 'evening. |
| 00:54.30 | dan__t | What do I do in the event that I want to make more than one ODBC query "at the same time"? Can I use multiple writesql statements in a function inside of func_odbc.conf, or do they need to be separate functions? |
| 00:55.04 | dan__t | MOre than one update, insert, delete, some kind of destructive change, not a select. |
| 00:55.33 | *** join/#asterisk Guest8383 (~Geek@unaffiliated/cain) |
| 00:59.38 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 01:00.38 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:02.50 | *** join/#asterisk darkdrgn2k (~darkdrgn2@70.51.25.146) |
| 01:02.52 | darkdrgn2k | hey guys |
| 01:02.55 | darkdrgn2k | having a problem with a dial plan |
| 01:03.40 | darkdrgn2k | exten => 4054807001,n,Dial(local/4054807001@outbound-allroutes,,D(1)r) |
| 01:03.59 | darkdrgn2k | it looks like it works |
| 01:04.18 | darkdrgn2k | but you dont hear anything |
| 01:14.01 | *** join/#asterisk dlublink (~david@75-119-235-201.dsl.teksavvy.com) |
| 01:19.15 | dlublink | I want to disable dtmf detection. It either comes in as RFC2833 or not at all. Any ideas? |
| 01:20.16 | *** join/#asterisk james_zhu (~Administr@183.16.210.41) |
| 01:41.43 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:42.32 | *** join/#asterisk coppice (~coppice@m121-202-76-125.smartone-vodafone.com) |
| 01:45.36 | prgmrchris | tyman: just limit ami access in your firewall? |
| 01:46.59 | tyman | prgmrchris: y, i planned on doing thatâ¦but worried about having the ami manager secret in each client. Should i be concerned about this for either performance or security? |
| 01:47.26 | tyman | diff secret for each i guessâ¦just that many connections to ami with relatively loose privs |
| 01:50.21 | prgmrchris | if its firewalled and only allowed to your client location, are you really worried about an internal threat? |
| 01:50.44 | prgmrchris | someone with that level access could easily sniff sip secrets also |
| 01:52.19 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 01:54.19 | dijib | How would you guys improve this line to add any more info? and should this be set as a variable?? exten => 100,n,MixMonitor(/var/spool/asterisk/monitor/incoming/IN-${EXTEN}-${CALLERID(all)}-${STRFTIME(${EPOCH},,%C%y-%m%d-%H%M%S)}.wav,ab) |
| 01:55.02 | tyman | prgmrchris: what about load? are multi-connections to the ami load intensive? |
| 01:55.31 | tyman | or blocking in any way? |
| 01:56.52 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:02.38 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 02:03.07 | *** join/#asterisk Atriks (~Hatrix@2a01:e34:ee51:4670:c824:1367:21ec:abf9) |
| 02:05.59 | *** join/#asterisk dmz (~dmz@64.203.235.49.dyn-cm-pool-34.pool.hargray.net) |
| 02:07.17 | tyman | I'm using ConfBridge w/1.8.5.0 and wondering why after hanging up, as the sole person in a confbridge, the dialplan does ends without flowing thru the rest of the extension config? |
| 02:08.16 | tyman | looking to change some AstDB entries after the conf ends... |
| 02:21.47 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 02:26.52 | *** join/#asterisk bbryant (~brett@c-174-56-132-225.hsd1.sc.comcast.net) |
| 02:28.35 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 02:34.04 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 02:49.03 | dijib | p3nguin, you alive? |
| 02:49.04 | rawburt | any Hitachi WirelessIP5000 users out there? I don't know what to put in the fields for the SIP configuration.. |
| 02:55.30 | *** join/#asterisk james_zhu (~Administr@183.16.210.41) |
| 02:59.38 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 03:06.02 | ChannelZ | not I but I'd hope it'd be semi self-explanatory |
| 03:21.58 | *** join/#asterisk fireman_biff (~biff@65.48.145.62) |
| 03:23.12 | fireman_biff | I'm getting "PRI got event: Alarm (4) on Primary D-channel of span 1" every 3 - 4 minutes. Could this be anything other than an issue with the provider? |
| 03:23.44 | fireman_biff | the alarm only lasts for about 5 seconds each time |
| 03:25.26 | *** join/#asterisk chazzam (~chazz@173-24-236-90.client.mchsi.com) |
| 03:29.10 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 03:32.43 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 03:33.40 | *** join/#asterisk radic (~radic@dslb-178-002-227-139.pools.arcor-ip.net) |
| 03:46.25 | tyman | I'm using ConfBridge w/1.8.5.0 and wondering why after hanging up, as the sole person in a confbridge, the dialplan does ends without flowing thru the rest of the extension config? |
| 03:46.39 | tyman | looking to change some AstDB entries after the conf ends... |
| 03:50.25 | tyman | Any ideas? The core show application ConfBridge docs say nothing about not falling thru as do other applications |
| 04:03.50 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 04:19.29 | *** join/#asterisk ajkaanbal (~ajkaanbal@189.143.123.29) |
| 04:19.51 | *** join/#asterisk tyman_ (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 04:30.47 | *** join/#asterisk Syrex (~syrex@dsl-146-17-198.telkomadsl.co.za) |
| 04:38.42 | *** join/#asterisk greenwolf (penguin@cpe-74-77-221-5.buffalo.res.rr.com) |
| 04:38.51 | *** part/#asterisk greenwolf (penguin@cpe-74-77-221-5.buffalo.res.rr.com) |
| 04:39.03 | *** join/#asterisk greenwolf (penguin@cpe-74-77-221-5.buffalo.res.rr.com) |
| 04:39.27 | fireman_biff | whats the difference between isdn, pri and t1? |
| 04:40.40 | ChannelZ | Capacity, availability, price? |
| 04:43.01 | ChannelZ | T1 is a PRI.. 24 channels. ISDN is a BRI, only 2 channels I think? |
| 04:43.03 | fireman_biff | but aren't they very closely related from a technical point of view? the intro in wikipedia makes it seem like if youre using a pri then youre also using isdn and t1 (or e1) |
| 04:43.55 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 04:45.16 | ChannelZ | I don't really know protocol-wise how similar or dissimilar they are |
| 04:47.19 | ChannelZ | That said I believe ISDN can 'run on top of' a T1 or E1 |
| 04:48.37 | fireman_biff | thats the impression i'm getting too |
| 04:49.42 | ChannelZ | There are multiple flavors of both in either case, but someone who has more technical understanding of them would have to elaborate |
| 04:51.02 | kaldemar | pri is also isdn |
| 04:54.59 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 05:00.03 | kaldemar | a flavor of isdn that can be run over a T1. |
| 05:06.14 | *** join/#asterisk Syrex (~syrex@dsl-146-17-198.telkomadsl.co.za) |
| 05:09.47 | p3nguin | dijib: Sure I am. |
| 05:09.58 | dijib | you must be in china or something |
| 05:10.01 | dijib | up at this hour |
| 05:10.14 | p3nguin | USA |
| 05:10.21 | p3nguin | Central time |
| 05:10.36 | p3nguin | I just got in for the night. |
| 05:11.10 | p3nguin | I have a headache, I'm tired, and I'm extremely thirsty. |
| 05:12.37 | irroot | p3nguin more tequilla |
| 05:12.55 | irroot | morning from my side 7am |
| 05:29.16 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 05:31.50 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 05:41.24 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 05:50.49 | *** join/#asterisk bbryant (~brett@c-174-56-132-225.hsd1.sc.comcast.net) |
| 05:50.59 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 05:55.55 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 06:00.05 | *** join/#asterisk nix8n82-phone (~AndChat@c-24-8-200-7.hsd1.co.comcast.net) |
| 06:02.00 | *** join/#asterisk james_zhu (~Administr@183.16.210.41) |
| 06:05.02 | *** join/#asterisk gseven (~gseven@host-81-190-234-22.wroclaw.mm.pl) |
| 06:05.27 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:05.50 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:11.50 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 06:13.05 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 06:16.27 | *** part/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 06:19.57 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-wvclxplbonuzexhh) |
| 06:23.14 | ChannelZ | It's definately pants-off time here |
| 06:30.11 | james_zhu | I am in china |
| 06:30.20 | james_zhu | :) |
| 06:34.52 | *** join/#asterisk gg0 (~gg0@unaffiliated/gg0) |
| 06:47.12 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 06:49.10 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 06:52.23 | *** join/#asterisk gravin (~gravin@113.210.124.243) |
| 06:52.39 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 06:59.22 | *** join/#asterisk den512 (~dradon@carbon.gonicus.de) |
| 06:59.24 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 07:11.48 | *** join/#asterisk f2knight (~ben@c-76-115-44-207.hsd1.or.comcast.net) |
| 07:16.12 | *** join/#asterisk StaRetji (~BigAll@46.12.157.16) |
| 07:16.56 | *** join/#asterisk nZw (~rychu@85.11.67.125) |
| 07:17.24 | StaRetji | Hi folks, I've made some extensions for a caller to input his mobile phone number exten => 1,4,Read(digits,,10) |
| 07:18.13 | StaRetji | Is there a way to check if first two numbers he entered are 06 and if not offer exten => 1,4,Read(digits,,10) again? |
| 07:21.02 | *** join/#asterisk iulhk (~carryoff@119.152.10.207) |
| 07:21.06 | *** join/#asterisk Nasga (~Nasga@186.212.10.93.rev.sfr.net) |
| 07:24.06 | *** join/#asterisk Polysics (~luca@host210-142-static.228-95-b.business.telecomitalia.it) |
| 07:24.07 | Polysics | hello |
| 07:24.15 | Polysics | having trouble finishing this |
| 07:24.52 | Polysics | caller dials in, MoH gets started, then I start dialing people in succession, each one can press 1 to accept, i bridge them if they accept |
| 07:24.59 | Polysics | that's the theory |
| 07:25.23 | Polysics | in practice, how do i know what people did in the extension i call with Originate? |
| 07:25.37 | Polysics | the problem is with looping, not the actions themselves |
| 07:30.04 | ChannelZ | StaRetji: GotoIf with an expression of $["${digits:0:2}"="06"] |
| 07:31.12 | ChannelZ | at least I think that's right |
| 07:31.27 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 07:31.28 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 07:31.38 | StaRetji | ChannelZ: thank you so much, I will try it now |
| 07:31.39 | StaRetji | thx |
| 07:32.05 | *** join/#asterisk enoch (~enoch@unaffiliated/enoch) |
| 07:32.11 | enoch | hi all |
| 07:32.16 | enoch | i have a simple question |
| 07:32.27 | enoch | can asterisk work using olds ac97 pci modems? |
| 07:32.33 | ChannelZ | StaRetji: I guess you'd actually want != not = but you get the idea |
| 07:32.51 | tzafrir_laptop | enoch, no (unless someone writes a driver for them) |
| 07:33.22 | ChannelZ | Just get a cheap ATA |
| 07:33.33 | enoch | ChannelZ: like? |
| 07:33.39 | StaRetji | ChannelZ: you mean GotoIf $["${digits:0:2}"="!06"] |
| 07:33.40 | StaRetji | ? |
| 07:34.01 | ChannelZ | Linksys SPA3102 |
| 07:34.08 | enoch | no one wrote a driver before? |
| 07:34.10 | StaRetji | Sorry if question is dumb :/ |
| 07:35.14 | ChannelZ | StaRetji: no, I mean $["${digits:0:2}"!="06"] makes more sense logically for what you asked, to say "if the first two digits is NOT 06 then jump somewhere" |
| 07:35.37 | StaRetji | ChannelZ: exactly |
| 07:35.52 | enoch | so there's not a card actually working? |
| 07:35.55 | *** join/#asterisk ollii (~risker@vpn.nbank.de) |
| 07:36.27 | ChannelZ | Winmodems sucked even as modems |
| 07:38.28 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:40.05 | StaRetji | exten => 1,4,Read(digits,,10) |
| 07:40.05 | StaRetji | exten => 1,5,GotoIf($["${digits:0:2}"!="06"]) |
| 07:40.19 | StaRetji | I did this and typed 05xxxxxxxx |
| 07:40.32 | StaRetji | 10 digits and it ignore 1,5GotoIf |
| 07:41.02 | StaRetji | I must getting this wrong |
| 07:41.08 | StaRetji | must be* |
| 07:42.18 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 07:42.27 | ChannelZ | well that wasn't the whole logic |
| 07:42.59 | ChannelZ | You have to GotoIf somewhere... GotoIf(expression?iftrue:iffalse) |
| 07:43.03 | ChannelZ | core show application GotoIf |
| 07:43.19 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 07:44.35 | StaRetji | oh, thx ChannelZ, will try |
| 07:45.07 | ChannelZ | You have to do the rest of the logic yourself depending on how you have your dialplan setup. Generally you'd make a priority label where you want it to jump to, like exten => 1,4(retry),Read(digits,,10) |
| 07:45.29 | ChannelZ | Then you can GotoIf(whatever?retry) |
| 07:46.40 | ChannelZ | actually that's a lie |
| 07:46.53 | StaRetji | :) |
| 07:48.35 | ChannelZ | the syntax is the same as Goto |
| 07:49.01 | StaRetji | I want to accomplish following. Ask caller to input his phone number. If he inputs 0612345678 he proceed down to the tree, if he puts first to numbers !06, he should get prompt again and again until hr types 06xxxxxxxx |
| 07:49.43 | StaRetji | sorry for the typos, still sleeping :) |
| 07:49.53 | ChannelZ | the label would really be GotoIf(whatever?context,extension,retry) where 'context' is whatever context this exten is in, 'exten' being 1 based on the example you pasted above. But "core show application goto" |
| 07:50.31 | ChannelZ | Yeah I know what you're doing, I just lied a little on the syntax. |
| 07:51.55 | StaRetji | thx so much ChannelZ, you've helped a lot. I will now play with it and hopefully make it work ;) |
| 07:53.24 | ChannelZ | Here's a simple example since I was steering you badly - http://pastebin.com/yV3y3hFd |
| 07:53.29 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85f2.bcn.adamo.es) |
| 07:53.51 | ChannelZ | Obviously you need to adapt it to your own dialplan, using whatever context you're actually using (I used 'incoming' as an example) |
| 07:54.08 | ChannelZ | and testexten being whatever your actual exten is - but you see how the pieces connect |
| 07:54.21 | StaRetji | got it |
| 07:55.32 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 07:59.16 | StaRetji | YAY :) |
| 07:59.23 | StaRetji | ChannelZ: that is it :) thx man |
| 07:59.42 | StaRetji | You're PRO! :) |
| 08:01.17 | ChannelZ | not so much but glad it works :) |
| 08:03.54 | *** join/#asterisk RZero (~Mick_Love@gateway-hq-uk.oxygen8.com) |
| 08:11.58 | *** join/#asterisk Formater (~Formater@217.17.105.246) |
| 08:12.44 | Formater | hi : |
| 08:12.45 | Formater | :) |
| 08:12.53 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 08:13.15 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:14.14 | ChannelZ | ahoy |
| 08:15.22 | irroot | the busy has released me for now |
| 08:15.40 | Formater | question.. in my dialplan i have this: exten => _X.,n,Set(CDR(userfield)=${CDR(userfield)}\;XXX) |
| 08:15.41 | Formater | the problem is the semicolon. if i dont escape it with \, then asterisk thinks comment starts, and throws error about missing closing bracet, but if i escape like this \;, then in userfield i have the escape character as well (\;) and not only ; |
| 08:15.53 | Formater | how should i properly escape the ; ? |
| 08:16.38 | *** join/#asterisk ixyd_ (~denzs@carbon.gonicus.de) |
| 08:16.39 | Formater | or is there anything like char(59) or something like that what i can use instead of ; ? |
| 08:16.53 | ChannelZ | I'm not sure what you're trying to do with the ;XXX in the first place |
| 08:17.23 | ChannelZ | you're trying to actually append that as a string? |
| 08:18.05 | irroot | Formater have you used " " |
| 08:18.12 | wdoekes2 | Formater: try "${CDR(userfield)};XXX" |
| 08:18.15 | ixyd_ | hi! iam doing call forwarding to external numbers by using Transfer() in the dialplan, but in that case i dont get any cdr information, is there any way to get a (pseudo) cdr in that case? |
| 08:18.18 | irroot | _X.,n,Set(CDR(userfield)="${CDR(userfield)};XXX") |
| 08:18.42 | irroot | ^^^ wdoekes2 thx |
| 08:18.43 | Formater | thanks, i will try now :) |
| 08:19.21 | irroot | ixyd_ you can dial a local channel if you want full info |
| 08:20.12 | wdoekes2 | ixyd_: what kind of info would you want? you cannot tell the duration of the transferred call |
| 08:21.17 | wdoekes2 | I would assume a cdr does get written regardless.. but you might need unanswered = yes in cdr.conf |
| 08:21.29 | ixyd_ | i know ;) but it would be nice, to see that the transfer happened... |
| 08:21.47 | *** join/#asterisk MarKsaitis (~MarKsaiti@94-30-69-205.xdsl.murphx.net) |
| 08:23.29 | ixyd_ | ah i didnt knew that option in cdr.conf... i'll give that a try! thank you! :) |
| 08:24.02 | Polysics | can i do something in an Originate call if the receiver hangs up instead of answering? |
| 08:24.27 | *** join/#asterisk remmo (~chatzilla@CPE-120-146-137-254.static.nsw.bigpond.net.au) |
| 08:25.12 | remmo | Hi Guys, has anyone noticed asterisk 1.8.5.0 and agi_callerid not being the caller id but the sip username |
| 08:25.20 | kaldemar | Polysics: not really, unless you use an extension that goes on in the dialplan in such a case. |
| 08:25.32 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:26.12 | Polysics | kaldemar, right now, i Originate a call to an extension that asks the receiver to press 1 and gets bridged on 1 |
| 08:26.23 | Polysics | if he does not press 1, input timeouts and that's it |
| 08:26.31 | ixyd_ | wdoekes2: thank you very much, unanswered=yes did the trickt! :) |
| 08:26.40 | Polysics | but if the receiver just rejects the call? what is the extension syntax? |
| 08:27.12 | *** join/#asterisk bbryant (~brett@c-174-56-132-225.hsd1.sc.comcast.net) |
| 08:28.56 | Polysics | i do not even think it is possible to operate after hangup in AGI, is it? |
| 08:29.25 | kaldemar | remmo: quite the opposite. |
| 08:29.56 | remmo | kaldemar: not from what I have seen. seems like 1.4 is fine, then we went straight up to 1.8.5.0 and have issues |
| 08:30.05 | kaldemar | Polysics: app Dial option g |
| 08:31.01 | Formater | thanks everyone, the "" helped :) |
| 08:31.49 | remmo | Polysics: thats what DeadAGI is for |
| 08:31.51 | kaldemar | remmo: well, i just tried on my 1.8.5.0. agi_callerid had the number i had setup in sip.conf and agi_calleridname had the name i had setup in sip.conf. From header in SIP had something else. |
| 08:32.07 | remmo | interesting we are using sip real time |
| 08:33.39 | kaldemar | remmo: if i remove the callerid option from sip.conf, then it sets both agi variables according to the from header. |
| 08:34.58 | Polysics | i am not using Dial, but i am in an Originate call |
| 08:35.10 | Polysics | by the way, isn't DeadAGI... dead? |
| 08:38.44 | kaldemar | Polysics: that's why i said "unless you use an extension". you can originate to an extension that does what you want. |
| 08:40.09 | Polysics | kaldemar, but i am not using Dial anywhere in my extension - my extension is "press 1 to accept the call", on 1 it uses Bridge, otherwise Hangup |
| 08:40.24 | *** part/#asterisk ixyd_ (~denzs@carbon.gonicus.de) |
| 08:40.32 | Polysics | if the call is rejected, no extension is reached |
| 08:44.51 | kaldemar | Polysics: i'm telling you to use Dial. |
| 08:45.20 | kaldemar | if you originate to an extension, how could the call be going to a phone without a Dial? |
| 08:45.51 | Polysics | Bridge |
| 08:46.08 | Polysics | the caller is already in the system listening to MoH |
| 08:52.53 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 08:56.16 | remmo | kaldemar: seems like sip realtime can not see the callerid. thanks for your help !! now i have a starting point |
| 08:58.11 | Formater | hm, i still have probelm with semicolon escaping. i added the ", now the line looks like this: exten => _.,n(app1),Set(CDR(userfield) = "${CDR(userfield)};"), and i got the following error from asterisk 1.8, |
| 08:58.12 | Formater | [Aug 25 08:32:25] WARNING[18295]: pbx.c:3863 pbx_substitute_variables_helper_full: Error in extension logic (missing '}') |
| 08:58.12 | Formater | [Aug 25 08:32:25] WARNING[18295]: pbx.c:3493 func_args: Can't find trailing parenthesis for function 'CDR(userfiel'? |
| 09:00.01 | Formater | what i'm doing wrong?: |
| 09:00.02 | Formater | :) |
| 09:02.14 | *** join/#asterisk gravin (~gravin@175.138.140.21) |
| 09:04.06 | Polysics | couldn't I use the Originate APPLICATION instead of the AMI command? that does seem to set a channel variable |
| 09:07.48 | *** join/#asterisk carrar (~tim@osburn.com) |
| 09:10.30 | Polysics | uhm |
| 09:10.52 | Polysics | if i set an ActionID for my Originate, don't i get an OriginateResponse with the same ID later? |
| 09:11.02 | Polysics | i get a different ID :-/ |
| 09:11.17 | Polysics | if they had the same ID i could handle the hangup or timeout that way |
| 09:13.28 | *** join/#asterisk xbp (seattle@kingdom.scns.com) |
| 09:22.57 | kaldemar | Formater: semicolons are tricky. try Set(CDR(userfield)=${CDR(userfield)}\;) to add the semicolon and Set(CDR(userfield)=${REPLACE(CDR(userfield),\\\\)}) to remove the backslash. |
| 09:23.10 | kaldemar | Formater: if you find a better way, feel free to share. :) |
| 09:26.41 | *** join/#asterisk droemel (~droemel@p4FCAD139.dip.t-dialin.net) |
| 09:34.38 | *** join/#asterisk chrisjunkie (~Chris@2001:4428:22d:2:208:2ff:fe7e:6312) |
| 09:47.23 | *** join/#asterisk gravin (~gravin@180.74.149.0) |
| 09:51.52 | *** join/#asterisk james_zhu (~Administr@183.16.210.41) |
| 09:59.29 | mandla | Hello gents, can ISDN BRI lines carry fax signals?? |
| 09:59.51 | SunTsu | that's G$ fax |
| 09:59.54 | SunTsu | G4 even |
| 10:00.26 | irroot | mandla yes |
| 10:00.47 | irroot | mandla you have a fax machine on the system or you want to do fax to mai |
| 10:01.05 | mandla | irroot, i want to fax to mail. |
| 10:01.25 | irroot | use res_fax res_fax_spandsp |
| 10:01.52 | irroot | fax will come in as call and get handled by them no problem |
| 10:02.25 | mandla | irroot, Go to private dialog |
| 10:05.19 | zamba | how can i get rid of "doing dnsmgr_lookup for" and "ast_get_srv: SRV lookup for" messages in the asterisk console? |
| 10:05.23 | zamba | i'm running at core set verbose 9 |
| 10:10.10 | irroot | res_fax/res_fax_spandsp |
| 10:28.47 | *** join/#asterisk xnfinite (~xnfinite@62.82.191.170.static.user.ono.com) |
| 10:30.44 | *** join/#asterisk flyman_ (~flyman_@194-024-138-050.dyn.orange.at) |
| 10:32.29 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 10:44.11 | coppice | I think they called it G4 FAX, because 4 was about the number of models that ever made it to market.... or was 4 the total production run? :-\ |
| 10:45.08 | WIMPy | Possible, but thenL Who uses real fax machines anyway? |
| 10:48.46 | coppice | until 5 years ago most people did |
| 10:56.49 | *** join/#asterisk blue (~blue@irc.mmh.dk) |
| 10:57.32 | WIMPy | The big run on CAPI was >10 years ago. |
| 10:58.55 | blue | I'm trying to load test our asterisk system. It has to handle a lot of incomming calls that only does a Playback. Does it make any sense to just use Local channels generated from call files to test? |
| 10:59.10 | *** join/#asterisk Atriks (~Hatrix@2a01:e34:ee51:4670:c824:1367:21ec:abf9) |
| 10:59.21 | Kobaz | local channels wont use the same load as sip/rtp or which ever protocol you're using |
| 11:00.24 | coppice | the number of computer to computer FAXes has always been tiny. At least one end of the link is almost always a real G3 FAX machine |
| 11:00.39 | blue | we are using an iax trunk |
| 11:01.20 | blue | but it will be hard for us to generate real calls |
| 11:02.12 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 11:02.43 | Kobaz | why is it hard? |
| 11:02.56 | Kobaz | just have a local channel call an iax device |
| 11:03.07 | Kobaz | it's the same process, just slightly different dialplan |
| 11:03.18 | WIMPy | I use a real fax, but if you have a Windows machine somewhere you will certainly use that. The required software is bundled with almost any card/modem/router. |
| 11:04.09 | Kobaz | or don't use local channels and use sipp with sip peers |
| 11:05.37 | blue | we have no hardware besides the asterisk server, so the iax calls have to to through the trunk |
| 11:05.59 | coppice | it always amazed me how few calls into FAX servers ever appeared to originate from windows machines |
| 11:07.03 | blue | we want to see if the server can handle 500 simultaneous calls |
| 11:07.19 | Kobaz | blue: but that's what you're testing isn't it? |
| 11:07.34 | Kobaz | what better way to test than to use the links that will actually be used |
| 11:07.36 | blue | yes, thats what we are trying to test :) |
| 11:07.40 | blue | true |
| 11:08.44 | blue | but wont using our own line to call ourselves cause problems? |
| 11:09.19 | Kobaz | it's not a "line" in the traditional sense of the word |
| 11:09.28 | Kobaz | it's an account that you can place calls on |
| 11:09.48 | Kobaz | it'll support as many calls as it's configured to support |
| 11:09.52 | blue | ok |
| 11:10.17 | blue | so i could try making call files that uses iax to call our own number |
| 11:10.24 | Kobaz | that's the nice thing about voip |
| 11:10.29 | blue | and we how many we can run through |
| 11:10.32 | Kobaz | it's like serving up web pages |
| 11:10.36 | blue | and if load gets heavy |
| 11:10.41 | Kobaz | you'll have to figure that out yourselv |
| 11:10.56 | Kobaz | depends on server specs and what each call is doing |
| 11:11.12 | blue | i'll give it a try, thanks :) |
| 11:11.41 | blue | the calls arent doing much besides playing a sln file |
| 11:22.40 | *** join/#asterisk vassilux (~vassilux@31.35.29.247) |
| 11:28.55 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 11:29.14 | *** part/#asterisk vassilux (~vassilux@31.35.29.247) |
| 11:43.30 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:46.09 | *** join/#asterisk Arcopix (~Arcopix@78.90.85.118) |
| 11:46.16 | Arcopix | Hello all |
| 11:46.33 | *** join/#asterisk treborsux (~treborsux@75-144-117-117-Jacksonville.hfc.comcastbusiness.net) |
| 11:46.48 | treborsux | What is a good polycom phone with a big screen for an operator? |
| 11:46.54 | treborsux | 560 is too small |
| 11:50.35 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 12:00.25 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 12:07.59 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 12:16.08 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 12:16.30 | *** join/#asterisk mickecarlsson (~Micke@h10n3c1o1101.bredband.skanova.com) |
| 12:19.25 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:25.48 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
| 12:32.08 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 12:33.24 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 12:39.58 | *** join/#asterisk gravin (~gravin@180.74.149.0) |
| 12:53.11 | *** join/#asterisk flyman_ (~flyman_@194.24.158.1) |
| 12:55.00 | *** join/#asterisk dwayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 13:04.15 | *** join/#asterisk TobyRulez (~TobyRulez@66-191-161-122.dhcp.gnvl.sc.charter.com) |
| 13:07.11 | *** join/#asterisk bbryant (~brett@c-174-56-132-225.hsd1.sc.comcast.net) |
| 13:10.05 | *** join/#asterisk serafie (~erin@nat/digium/x-hvxrabdgyvmminan) |
| 13:11.35 | *** join/#asterisk greenwolf (greenwolf@cpe-74-77-221-5.buffalo.res.rr.com) |
| 13:15.08 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-vfvdxtloealnwwiz) |
| 13:17.06 | *** join/#asterisk bent_screwdriver (~bent_scre@74.255.249.66) |
| 13:23.14 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 13:24.38 | bent_screwdriver | i have some polycom ip650's in a ring group. sometimes one of them will keep ringing even after the call has been answered (happens on multiple phones randomly). it seems a reboot of the phones will stop it for several days but then it resurfaces. phones are on same lan/switch with PBX (* 1.6.2) any ideas? thx |
| 13:26.14 | *** join/#asterisk af_ (~getsmart@78.134.21.92) |
| 13:26.40 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 13:40.16 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 13:42.48 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 13:46.23 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:46.55 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 13:47.31 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 13:48.54 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 13:53.07 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:53.30 | *** join/#asterisk bn-7bc (~bjarne-im@pdpc/supporter/active/bn-7bc) |
| 14:00.43 | *** part/#asterisk sorressean (~tyler@tds-solutions.net) |
| 14:03.11 | Arcopix | bye |
| 14:04.00 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-vfvdxtloealnwwiz) |
| 14:04.18 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-hopzaituklykzkaa) |
| 14:07.01 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 14:07.01 | *** join/#asterisk MarKsaitis (~MarKsaiti@94-30-69-205.xdsl.murphx.net) |
| 14:07.02 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 14:07.02 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-wvclxplbonuzexhh) |
| 14:07.02 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 14:07.02 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 14:09.06 | *** join/#asterisk james_zhu (~Administr@116.30.243.167) |
| 14:14.46 | *** join/#asterisk chris613 (~chris@CPE001839c16d35-CM00237453c586.cpe.net.cable.rogers.com) |
| 14:15.59 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:18.28 | chris613 | Having a frustrating inbound call problem. I get the chan_sip message "Call from 'censored' (IP.IP.IP.IP:PORT) to extension 'IP.IP.IP.IP:PORT' rejected because extension not found in context 'censored'. However a 'dialplan show' for the 'censored' context shows one extension labelled '_X.' which should match anything starting with a number, and that IP in the message certainly does start with a number |
| 14:19.20 | chris613 | it's like the wildcard match is broken. It's 1.8.5, and I've tried toggling the "extenpatternmatchnew" on and off, no difference |
| 14:19.57 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 14:21.11 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:21.11 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:23.21 | chris613 | Heck, I just tried 'dialplan show IP.IP.IP.IP:PORT@censored' and it spit out the correct extension config. 'dialplan show bull@censored' returns "There is no existence of bull@censorce extension' since it's not numeric |
| 14:25.06 | *** join/#asterisk coppice (~chatzilla@116.92.16.50) |
| 14:25.21 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:26.02 | Hive | chris, if it's the same IP calling in, you could try making a dial plan specifically for that number |
| 14:26.13 | Hive | to see if it is the pattern matching or something else that is messing up |
| 14:26.48 | chris613 | I will try that, but I'm not to happy with the idea that pattern matching is broken... let me see if it helps, though |
| 14:27.19 | Hive | I doubt it's broken, otherwise i feel like more people would be complaining hehe |
| 14:27.27 | chris613 | Yeah me too |
| 14:27.46 | chris613 | But I can't really explain what I'm seeing |
| 14:28.44 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
| 14:29.06 | *** join/#asterisk zkn (~zkn@193.90.201.218) |
| 14:29.21 | zkn | Hello |
| 14:29.51 | chris613 | Hive, it didn't work. Something fishy is going on... |
| 14:30.04 | chris613 | I have a meeting :( will be back at this soon |
| 14:30.10 | Hive | chris, can you pastebin your command line error |
| 14:30.11 | Hive | ahh okay |
| 14:30.51 | zkn | i've a problem with accepting inbound calls from the ITSP, error message says Failed to authenticate device |
| 14:30.59 | zkn | where do i need to look? |
| 14:31.31 | chris613 | Hive, I will certainly do that upon my return. Thanks for your help |
| 14:31.39 | Hive | No problem |
| 14:32.06 | Hive | zkn, are your devices registered to your * server? |
| 14:32.15 | *** join/#asterisk din3sh (din3sh@41.212.248.197) |
| 14:32.35 | zkn | the device it is referring to is my cellphone number that i'm using to yet the inbound call with |
| 14:32.39 | ironm | hello - is there a possibility to control asterisk writing "/var/log/asterisk/cdr-csv/Master.csv" for the time you want to import this file to database and to "reset" this file (to avo8id double CDR records)? Thank you in advance for any hints. |
| 14:33.25 | zkn | my asterisk server is registered to the ITSP, sip show registry displays Registered |
| 14:33.29 | *** join/#asterisk dre- (~dre@69.38.200.246) |
| 14:33.48 | din3sh | anyone can help me out |
| 14:34.14 | din3sh | https://issues.asterisk.org/view.php?id=11849#c105921 |
| 14:34.36 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:34.45 | din3sh | has this issue of missing cdr on transfer been resolved in 1.8 or 10? |
| 14:35.12 | Hive | din3sh i didnt check your link but |
| 14:35.18 | Hive | cdr and trasnfers are not friends |
| 14:35.25 | Hive | you need to use the cel i believe it is called |
| 14:35.27 | chris613 | Seems I got out of the meeting.... |
| 14:35.33 | Hive | well done, chris! |
| 14:35.53 | chris613 | Hive, I think I found this. With sip debug on I see the invoice looks like this: INVITE sip:IP.IP.IP.IP:PORT |
| 14:36.05 | chris613 | So it has no extension to match! I will try changing my _X. to s |
| 14:36.23 | din3sh | i tried the agi scripts listed on the thread |
| 14:36.33 | din3sh | it works for 1.4.22 |
| 14:36.45 | Hive | din3sh; http://www.voip-info.org/wiki/view/Asterisk+config+cel.conf |
| 14:36.52 | din3sh | hmmm |
| 14:36.57 | Hive | i think this will help you |
| 14:36.58 | din3sh | ok thnx will have a look at it |
| 14:37.05 | Hive | in your quest for tracking transfers |
| 14:37.11 | Hive | if that is what you are trying to do |
| 14:37.26 | din3sh | so the missing transfer cdr bug isnt resolved even in 1.8 and 10 |
| 14:37.36 | Hive | i dont think so |
| 14:37.39 | Hive | i dont use the cdr haha |
| 14:37.42 | din3sh | yes for call billing/accounting purposes |
| 14:37.49 | Hive | i wrote my own stuff to track call information |
| 14:38.03 | Hive | yeah, then the CEL table is most likely what you will need to use |
| 14:39.40 | din3sh | checked the link |
| 14:39.57 | din3sh | how to use the created CEL log for billing purposes? |
| 14:40.10 | din3sh | any info on that? |
| 14:40.42 | Hive | sorry, you'll have to do some digging to find information on that |
| 14:40.46 | din3sh | haha |
| 14:40.48 | Hive | I can merely show you the door :) |
| 14:41.20 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 14:41.24 | din3sh | you wrote your own scripts for log purposes? |
| 14:43.37 | Hive | indeed |
| 14:43.53 | *** join/#asterisk SuperNull (~SuperNull@24-148-101-238.ip.mhcable.com) |
| 14:43.54 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 14:46.07 | *** part/#asterisk james_zhu (~Administr@116.30.243.167) |
| 14:46.54 | SuperNull | Hey guys im using odbc config for sipusers is there a way to setup dtmfmode this way? i dont see the field in the sql table.. |
| 14:48.10 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 14:48.19 | *** join/#asterisk DrDigi (~mmurphy@50-73-49-97-static.hfc.comcastbusiness.net) |
| 14:50.00 | *** join/#asterisk bitbandit (~mojojo200@c-98-202-116-168.hsd1.ut.comcast.net) |
| 14:51.30 | *** join/#asterisk JCCash (JCash@200.205.212.102) |
| 14:52.01 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:52.19 | *** join/#asterisk raj-darkmystery (~suraj@49.248.156.174) |
| 14:52.39 | raj-darkmystery | SIP/2.0 408 Request Timeout need help |
| 14:55.00 | Hive | raj, need a little more information than that |
| 14:55.10 | Hive | you can pastebin a chunk of your CLI |
| 14:55.16 | zkn | ok, seems like the only way to get past the "Failed to authenticate device" error is to set allowguest=yes, which is definitely what i want to allow |
| 14:55.33 | zkn | what i do not what to allow, that is |
| 14:56.42 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 14:57.44 | raj-darkmystery | Hive, I am trying to connect with sip server using twinkle.. from office I can connect to the server (located somewhere else) but from home i am not able to register the peer/sip I am not sure if i how to pastebin but i can paste output here as its of 4-5 lines only |
| 14:59.54 | raj-darkmystery | Hive, here is the output |
| 14:59.57 | raj-darkmystery | +++ 25-8-2011 20:18:59.338905 INFO NORMAL t_tc_non_invite::timeout |
| 14:59.57 | raj-darkmystery | Timer F expired. |
| 14:59.57 | raj-darkmystery | Send internal: |
| 14:59.57 | raj-darkmystery | SIP/2.0 408 Request Timeout |
| 14:59.57 | raj-darkmystery | Via: SIP/2.0/UDP 172.16.0.4;rport;branch=z9hG4bKznyvupkk |
| 14:59.59 | raj-darkmystery | To: "user_name_changed" <sip:1234@xxx.xxx.xxx.xxx>;tag=svhwo |
| 15:00.01 | raj-darkmystery | From: "user_name" <sip:1234@xxx.xxx.xxx.xxx>;tag=klpkx |
| 15:00.03 | raj-darkmystery | Call-ID: user_id@hostname |
| 15:00.05 | raj-darkmystery | CSeq: 264 REGISTER |
| 15:00.07 | raj-darkmystery | Server: Twinkle/1.4.2 |
| 15:00.09 | raj-darkmystery | Content-Length: 0 |
| 15:00.36 | sunfone | that is a few more than 4-5 lines |
| 15:00.52 | *** join/#asterisk n3hxs (~ed@63.68.135.4) |
| 15:01.18 | Hive | haha sorry, i thought you were looking at the asterisk command line on the server |
| 15:01.24 | Hive | thats there the valuable data is :D |
| 15:01.37 | raj-darkmystery | sunfone, sorry for pasting the output here but I wasn't sure about the procedure of pastebin :( |
| 15:01.58 | raj-darkmystery | ohh can't I do anything from client side? |
| 15:02.10 | raj-darkmystery | I don't have any access to the server :( |
| 15:02.25 | Hive | so you have a softphone, and you are trying to connect to an asterisk server without success |
| 15:02.27 | Hive | is that correct? |
| 15:02.34 | raj-darkmystery | yes |
| 15:02.35 | Hive | just trying to understand your problem |
| 15:02.39 | raj-darkmystery | it's correct |
| 15:02.46 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 15:02.56 | raj-darkmystery | from office I can connect but from home 'm not |
| 15:03.02 | Hive | hmm |
| 15:03.05 | raj-darkmystery | and ports are open |
| 15:03.09 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 15:03.18 | Hive | timeout error ehh... |
| 15:03.21 | raj-darkmystery | as I can telnet from other machine out of my home network |
| 15:03.21 | Nugget | telnet is eeeeeeevil! |
| 15:03.22 | din3sh | yup |
| 15:03.27 | din3sh | its a timeout prob |
| 15:03.38 | din3sh | cant get connection to your registrar |
| 15:03.55 | sunfone | what does telnet have to do with anything? |
| 15:04.03 | raj-darkmystery | right din3sh thats the issue but 'm not finding any way out |
| 15:04.06 | sunfone | what ports do you think are "open"? |
| 15:04.20 | raj-darkmystery | sunfone, i was trying to check if the ports are unaccessible at home |
| 15:04.21 | din3sh | sip register? |
| 15:04.39 | sunfone | SIP is a UDP protocol; telnet is TCP |
| 15:04.45 | raj-darkmystery | din3sh, is that a command? sorry but 'm new for this sip thing |
| 15:04.50 | sunfone | telnet won't tell you if SIP is able to pass |
| 15:05.03 | din3sh | check your sip port setting |
| 15:05.05 | raj-darkmystery | ohh so sorry sunfone .. thanks for the info |
| 15:05.07 | din3sh | and firewall |
| 15:05.21 | raj-darkmystery | sip port is 5060 |
| 15:05.23 | din3sh | maybe your firewall allowing telnet but not the sip port |
| 15:05.37 | raj-darkmystery | firewall is disabled |
| 15:05.48 | sunfone | you should be talking to your asterisk server admin |
| 15:06.07 | sunfone | perhaps they are not allowing your network from home to register |
| 15:06.17 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:06.20 | JCCash | WARNING[3416] chan_iax2.c: Max retries exceeded to host xxx.xxx.xxx.xxx on IAX2/OPERADORA_out-13754 (type = 6, subclass = 2, ts=1050069, seqno=209) |
| 15:06.21 | raj-darkmystery | i was able to register two days back |
| 15:06.46 | sunfone | is your Internet connection at home dynamic? |
| 15:06.48 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 15:06.48 | raj-darkmystery | i am facing this issue from last two days only and tried a lot to troubleshoot |
| 15:06.58 | raj-darkmystery | sunfone, yes its dynamic |
| 15:07.14 | sunfone | is your asterisk server an ITSP or run by your company? |
| 15:07.17 | raj-darkmystery | but was working till two days back without any issues |
| 15:07.26 | raj-darkmystery | ITSP |
| 15:07.34 | sunfone | You should call their support department then |
| 15:07.40 | *** join/#asterisk QuantumSchema (~QuantumSc@rrcs-24-227-92-3.se.biz.rr.com) |
| 15:07.48 | QuantumSchema | Good afternon all! |
| 15:07.54 | Hive | Hello QuantumSchema :) |
| 15:07.55 | QuantumSchema | afternoon.. |
| 15:07.56 | din3sh | you checked the connection logs on your router? |
| 15:08.11 | din3sh | if it was working |
| 15:08.26 | din3sh | and not working no, means a change in config |
| 15:08.30 | din3sh | obviously |
| 15:08.45 | QuantumSchema | Gotta quick question... has anyone experienced any issues w/ a Cisco ASA firewall and T.38 before? |
| 15:08.48 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:09.09 | din3sh | not really QuantumSchema..sorry |
| 15:09.11 | raj-darkmystery | din3sh, no I havn't checked over router yet as it is untouched and nothing has changed over computer as well |
| 15:09.23 | QuantumSchema | Thx din3sh |
| 15:09.56 | QuantumSchema | As far as ports are concerned, everything is the same w/ T.38 and RTP right? |
| 15:10.15 | QuantumSchema | The SIP happens on 5060 and then the two specify what port the RTP traffic will happen on. |
| 15:10.23 | QuantumSchema | Then T.38 just rides the RTP ports right? |
| 15:10.40 | *** join/#asterisk spditner (~simon@206-248-134-146.dsl.teksavvy.com) |
| 15:10.56 | din3sh | @raj-darkmystery if it was working and stopped working now, means definitely a change in config |
| 15:11.22 | raj-darkmystery | din3sh, but the same configuration works in office.. and stops working at home :( |
| 15:11.26 | spditner | trying to get app_confbridge going, however it keeps telling me: 'Bridge technology softmix has capabilities 0x34 (ulaw|adpcm|g726aal2) and we want 0x2 (gsm)' |
| 15:11.43 | din3sh | it ever worked at home or never actually did? |
| 15:12.04 | QuantumSchema | I'm banging my head against the wall trying to get a successful fax through. |
| 15:12.11 | raj-darkmystery | din3sh, it was working till 2 days back.. |
| 15:12.14 | QuantumSchema | I've got it working from ext to ext on the same * box. |
| 15:12.20 | spditner | however I have the gsm codec and format loaded -- so not sure where it's going wrong |
| 15:12.33 | *** join/#asterisk DrDigi (~mmurphy@50-73-49-97-static.hfc.comcastbusiness.net) |
| 15:12.34 | QuantumSchema | Then ext on a * to another ext on a different * box. |
| 15:13.13 | QuantumSchema | It only fails when I go from a real fax on PSTN through a SIP trunk w/ T.38 to an ext on my * box. |
| 15:13.44 | zkn | ok, when I use the trunk account details with a SIP client, then all is OK, i can receive calls
but Asterisk server is unable to do the same
even though I have another trunk configured on the same server which works, for some reason calls from the second trunk get rejected
|
| 15:14.08 | zkn | very frustrating |
| 15:15.32 | din3sh | @zkn check the logs for the SIP error you getting |
| 15:15.48 | din3sh | might point you to the nature of the problem |
| 15:15.49 | zkn | what logs? |
| 15:16.01 | din3sh | sip debug logs |
| 15:19.17 | *** join/#asterisk navaismo (~navaismo@187.170.13.36) |
| 15:19.38 | zkn | but what does allowguest=yes change? |
| 15:19.40 | navaismo | morning! |
| 15:19.59 | zkn | why, if i set it to yes, calls work, with "no" it does not |
| 15:20.07 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 15:21.00 | QuantumSchema | Why do I keep reading that T.38 is part of H.323. |
| 15:21.03 | QuantumSchema | Is it really? |
| 15:21.33 | din3sh | @zkn http://www.voip-info.org/wiki/view/Asterisk+sip+allowguest |
| 15:21.41 | p3nguin | If variables are being lost or unset after call forward is performed, that's the fault of the channel driver, isn't it? If I Set(__myVar=${CALLERID(num)}), then Dial() to my phone which has callfwd enabled, which in turn sends the call to a new extension, ${myVar} should retain the value once the call has been redirected, correct? |
| 15:24.02 | p3nguin | zkn: allowguest=yes allowed anonymous calls to be accepted into the context defined in the sip.conf general section, which should never be the context used for non-anonymous calls. If you set allowguest=no and calling from that ITSP no longer works, you aren't matching the peer correctly. Show me full sip debug output of a failed call. |
| 15:24.07 | zkn | din3sh, yep, sip debug showed that inbound call contact was "Anonymous", whereas when I read this voip-info site, i had all the requirements met with allow guest=no |
| 15:24.39 | p3nguin | If you know every peer which you'll accept calls from, you should set allowguest=no. |
| 15:25.25 | p3nguin | If you'll accept calls from various unknown people, for example, on your SIP URI, you'll need allowguest=yes (unless you know everyone who will call you on SIP URI). |
| 15:26.14 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 15:27.44 | zkn | the only sip peer i accept is the ITSP |
| 15:28.23 | zkn | but Asterisk dops calls coming from them |
| 15:28.29 | zkn | drops* |
| 15:30.22 | zkn | p3nguin: http://pastebin.com/FTb4ydkD |
| 15:30.55 | zkn | the 1st 14 lines can be ignored |
| 15:31.52 | p3nguin | No matching peer for '003725013499' <-- create a peer for your phone and then you can handle calls from it. |
| 15:32.57 | zkn | that is my phone number i'm calling from |
| 15:33.05 | zkn | my cell phone number |
| 15:33.17 | Hive | ooo maybe ill give you a call later ;-) |
| 15:33.18 | p3nguin | It says the user agent is sippy. That's surely not what your ITSP is using. |
| 15:33.29 | zkn | :D |
| 15:33.38 | zkn | i'm confused.. |
| 15:33.54 | p3nguin | Are you using something with sippy on it? |
| 15:34.00 | zkn | nope |
| 15:34.13 | zkn | it might what the ITSP is using for all i know |
| 15:34.14 | p3nguin | I'm not familiar with it, but it sounds like a soft phone's name to me. |
| 15:34.17 | p3nguin | maybe |
| 15:34.43 | p3nguin | Did you set trustrpid=yes in the peer entry for the ITSP? |
| 15:35.01 | zkn | nope |
| 15:35.25 | p3nguin | Do you have more than one phone number with that provider? |
| 15:35.38 | zkn | one thing i haven't quite understood is how does the peer context and register => .. know each other's presence |
| 15:35.41 | p3nguin | Do you have a register statement for them which includes your extension? |
| 15:35.52 | zkn | yes, i have a lot of numbers tied to this account |
| 15:35.59 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 15:36.14 | p3nguin | The peer definition and the register statement are actually not related that closely. |
| 15:36.27 | zkn | i did try to have a callbackexten with register statmenet |
| 15:36.36 | zkn | no difference, same error |
| 15:36.40 | p3nguin | The register statement tells your ITSP how to reach you. |
| 15:36.41 | *** join/#asterisk shtrb (~LENOVO1@unaffiliated/shtrb) |
| 15:36.56 | zkn | yep, that makes sense |
| 15:37.15 | p3nguin | The peer definition is used to match calls from them, as well as allows you to make calls to them which will include user account credentials. |
| 15:38.31 | p3nguin | I'd like to see your peer entry for the ITSP. Hide your password only! I don't need the password, but I do need everything else you've written in it. |
| 15:39.37 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 15:39.37 | zkn | http://pastebin.com/bgy3fjcY |
| 15:43.18 | p3nguin | Is the 213.115.26.x IP address yours or theirs? |
| 15:43.34 | zkn | that is the asterisk server |
| 15:43.37 | zkn | mine |
| 15:43.37 | p3nguin | okay |
| 15:45.38 | p3nguin | Stick "trustrpid=yes" in the peer entry, save, then run "sip reload" in asterisk cli. |
| 15:46.40 | zkn | same |
| 15:47.21 | zkn | Failed to authenticate device |
| 15:48.19 | zkn | zippy is some software for VOIP carriers |
| 15:48.24 | zkn | sippy* |
| 15:48.58 | p3nguin | It sounds more like a softphone to me. It would be a good name for one. |
| 15:49.16 | Hive | hahaha |
| 15:49.18 | p3nguin | Are you required to use fromuser and fromdomain in the peer configuration? |
| 15:50.15 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:51.21 | zkn | i do not know |
| 15:51.53 | zkn | i just got the user name, domain and pass and i have built the peer context with what i know |
| 15:53.14 | zkn | when I comment these lines out, are reload sip, still no change |
| 15:53.33 | *** join/#asterisk kerx (~kerx@76-240-161-60.lightspeed.irvnca.sbcglobal.net) |
| 15:54.15 | zkn | i will call it a day, need to run now but will try again tomorrow, probably i will call the ITSP to ask what the monkey |
| 15:54.44 | zkn | thanks for the effort , appreciated |
| 15:57.45 | *** join/#asterisk corretico (~luis@200.12.40.17) |
| 15:58.50 | *** join/#asterisk Micc (~Micc@c-98-232-46-178.hsd1.wa.comcast.net) |
| 15:59.15 | Micc | anyone have any experience using sipsak? |
| 15:59.41 | sunfone | Anyone used Yealink phones with SLA? |
| 15:59.42 | Micc | I can't seem to get it to timeout if I give it a bad server address. |
| 16:00.22 | navaismo | ?SLA |
| 16:01.44 | sunfone | Shared Line Appearance |
| 16:01.56 | sunfone | sla.conf in 1.4+ |
| 16:02.11 | sunfone | doing my best to emulate a key system |
| 16:02.18 | navaismo | Hi i have an stupid basic question, but I cant figure out how set the gotoiftime to work this hourly jobs: 1st October to 15th June Monday to Friday 07:45 till 16:30 Saturday 08:00 till 11:30 if closed play IVR1 |
| 16:02.35 | navaismo | the rest of the year if closed play ivr2 |
| 16:02.39 | sunfone | have had some success with Aastra phones + SLA |
| 16:03.17 | navaismo | thx sunfone i only know BLF |
| 16:03.33 | sunfone | Seems to be our biggest challenge selling to small shops - they all have key systems today and don't want to change! :) |
| 16:03.51 | sunfone | I'm determined to come up with a hosted key system |
| 16:05.55 | Micc | sunfone, I've found with a little guidance they are willing to try something new. Parking can help the transition too. |
| 16:06.24 | sunfone | Ya, that is the stance we have been taking, but it is an impediment to easy sales |
| 16:06.33 | *** part/#asterisk StaRetji (~BigAll@46.12.157.16) |
| 16:06.36 | sunfone | If I can get a hosted key system working, it will be a slam dunk |
| 16:06.48 | sunfone | And I am very close... just looking for the right phone for the job |
| 16:06.56 | sunfone | Need lots of programmable lamped keys |
| 16:07.06 | sunfone | The Yealink T26P has 13 |
| 16:07.25 | sunfone | So for a shop with say 8 or less phones and four "lines", this would be perfect |
| 16:07.49 | sunfone | but I have found that different manufacturers support for asterisk sla.conf is widely varying |
| 16:07.59 | sunfone | just wondering if anyone had tried with Yealink |
| 16:08.04 | sunfone | I've never used them... |
| 16:08.47 | sunfone | The Aastra models I have been playing with lack the number of buttons I need |
| 16:09.02 | sunfone | and most all phones have extra buttons that will just serve to confuse the poor clients :) |
| 16:09.11 | *** join/#asterisk irroot (~irroot@41.115.223.176) |
| 16:09.13 | sunfone | the "line" keys for example will NOT operate as they would expect |
| 16:09.37 | sunfone | I'd love to find a manufacturer that would be willing to create a custom SIP phone |
| 16:12.54 | TobyRulez | we use the aastra 9143i's because of the number of programmable buttons...7 i believe, we use most of them for speed dial park pickup |
| 16:13.24 | sunfone | I just shipped them out, but I think I had 6130's with 8 buttons |
| 16:13.47 | sunfone | The intention, like with most key systems, is to have a lit button for every "line", and another for every "station" (BLF) |
| 16:13.57 | sunfone | so 8 buttons just isn't enough but for very small shops |
| 16:14.04 | sunfone | I figure the minimum to be twelve |
| 16:14.12 | sunfone | twenty would be ideal |
| 16:14.36 | sunfone | I'd like to standardize the service to one phone type that will support up to say 16 stations and four lines |
| 16:14.41 | TobyRulez | there is (at least) one aastra model that can have a "sidecar" attached to it that is full of buttons...but too much imo |
| 16:14.44 | sunfone | anything larger they just have to be a PBX :) |
| 16:14.54 | sunfone | Ya it also has to be fairly inexpensive |
| 16:15.02 | sunfone | Aastra definitely has that going for it |
| 16:15.20 | TobyRulez | lol, definately hard to do for a voip phone vs an analog |
| 16:15.28 | sunfone | right! |
| 16:15.50 | sunfone | We are willing to buy a large volume to start... so trying to convince someone to make a custom phone for us |
| 16:16.13 | sunfone | I think it would be a unique offering today |
| 16:16.24 | TobyRulez | good luck, let us know if you find anyone willing. we usually buy ours by the case |
| 16:16.40 | sunfone | I'm thinking 10,000 to start |
| 16:16.49 | *** join/#asterisk StaRetji (~BigAll@46.12.157.16.dsl.dyn.forthnet.gr) |
| 16:17.04 | TobyRulez | yeah, definately would have to have a very large potential order to get someone to make a model for you |
| 16:17.11 | sunfone | so if anyone has ties to manufacturers please send them my way |
| 16:17.18 | sunfone | right |
| 16:17.26 | TobyRulez | perhaps if the * community can get together and create an ideal phone, we could do a bulk order or something |
| 16:17.47 | sunfone | there is definitely interest if you sift through the forums on SLA |
| 16:17.56 | StaRetji | Folks, need help with setting extension to send 4 of 5 calls to Voicemail, while 1 call listen to a Playback. Any help would be highly appreciated :) |
| 16:19.04 | navaismo | StaRetji can you explain us more |
| 16:19.36 | *** part/#asterisk spditner (~simon@206-248-134-146.dsl.teksavvy.com) |
| 16:20.21 | StaRetji | well |
| 16:20.41 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 16:21.17 | StaRetji | I'm trying to build service where person calls a number, so, of 5 calls, 1 should go directly to exten Playback, while 4 should go to exten Voicemail |
| 16:22.55 | StaRetji | or, let's say 2 of 10, doesn't matter |
| 16:23.05 | StaRetji | navaismo: btw, thx for reply :) |
| 16:24.29 | navaismo | the other day here mentioned the callgrop definition let me see if i found the code |
| 16:24.32 | Micc | sunfone, we've kind of moved away from the small office market and focus more on customers with 6+ lines now. They usually don't have a key system currently, so its an easy transition. And it seems to be more money for our time. Those small offices tend to have lots of network problems and take up a lot of support time. |
| 16:24.57 | Micc | sunfone, but there would certainly be money to be made there if there was a good key system phone. |
| 16:27.26 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 16:32.28 | navaismo | StaRetji still searching |
| 16:32.35 | StaRetji | navaismo: thx! |
| 16:35.26 | *** join/#asterisk leroybuckingham (43350083@gateway/web/freenode/ip.67.53.0.131) |
| 16:37.23 | navaismo | StaRetji not found :( , but basically is the example 2 in this link http://www.voip-info.org/wiki/view/Asterisk+func+group |
| 16:37.52 | navaismo | You use group_count to send the call to voicemail |
| 16:38.37 | StaRetji | navaismo: thx man, will read it |
| 16:39.39 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 16:41.52 | sunfone | Micc: yup, that is the idea - with a plug-n-play key system phone there wouldn't be much training to do, and we intend to do this hosted, so most network issues would be mitigated |
| 16:42.16 | sunfone | We believe the market of 10 phones or less is much bigger (per seat wise) than all the larger shops put together :) |
| 16:42.44 | sunfone | Becomes a matter of direct marketing |
| 16:48.33 | TobyRulez | or if you can convince your target market that you can provide something so worthwhile that it's ok to have to "learn" something new |
| 16:48.43 | leroybuckingham | Hey guys I'm just playing around a bit, I noticed the AGI command from the CLI, but I can't seem to get it to do anything but fauk, and I can't find official documentation on it. http://www.asterisk.org/node/re215.html |
| 16:48.57 | leroybuckingham | fauk = fail* sorry |
| 16:49.40 | TobyRulez | http://www.voip-info.org/wiki/view/Asterisk+AGI |
| 16:49.59 | TobyRulez | good place to start |
| 16:50.08 | leroybuckingham | I know and I'm familiar with AGI |
| 16:50.20 | leroybuckingham | but it's the cli command I'm stuck on |
| 16:50.53 | TobyRulez | your link seems to be broken |
| 16:51.24 | leroybuckingham | exactly ;) http://www.asterisk.org/node/48581 -- a lot of them are. |
| 16:54.42 | TobyRulez | i'm wondering if those are just referring to agi commands you can do on command line (agi debug, agi debug off, agi show, agi dumphtml) |
| 16:55.38 | leroybuckingham | ah i think i noticed something, according to this the channel must already be in the agi async application. http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/ |
| 16:55.47 | ChannelZ | are you trying to use "agi exec" from the console or something? |
| 16:56.21 | leroybuckingham | yeah |
| 16:58.54 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 17:02.41 | TobyRulez | looks like it is supposed to allow you to send AGI commands from AMI...interesting |
| 17:04.06 | leroybuckingham | yeah but you need to start the agi app from the dial plan (i think)... not sure how I could sensibly do that with freepbx |
| 17:04.20 | leroybuckingham | or in what situations |
| 17:04.43 | TobyRulez | have you looked at the actual ticket for it? https://issues.asterisk.org/view.php?id=11282 |
| 17:04.49 | p3nguin | If variables are being lost or unset after call forward is performed, that's the fault of the channel driver, isn't it? If I Set(__myVar=${CALLERID(num)}), then Dial() to my phone which has callfwd enabled, which in turn sends the call to a new extension, ${myVar} should retain the value once the call has been redirected, correct? |
| 17:05.06 | p3nguin | It seems to be true with chan_sip, anyway. |
| 17:05.44 | TobyRulez | looks like you send calls to AGI(agi:async) |
| 17:09.29 | *** join/#asterisk Poincare (~jefffnode@2001:470:cb24:4::2) |
| 17:09.51 | Hive | Lol P3nguin is that a macro? |
| 17:10.57 | Hive | Are you experiencing behavior that is not as you say p3nguin? |
| 17:11.10 | Hive | I had a lot of problems with sharing variables between transfers |
| 17:11.22 | TobyRulez | leroybuckingham: not sure if you've seen this, but looks like it has some good info about halfway down http://ofps.oreilly.com/titles/9780596517342/AGI.html |
| 17:12.08 | leroybuckingham | oh perfect thanks! |
| 17:12.30 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:12.33 | TobyRulez | np, i might have to do some reading myslef |
| 17:12.57 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 17:13.51 | p3nguin | hive: Is what a macro? I said call forwarding. |
| 17:14.21 | p3nguin | hive: You probably have issues with variables over transfers because you aren't putting underscores on your variables when you set them. |
| 17:14.53 | p3nguin | ~brokengtalk |
| 17:14.53 | infobot | i heard brokengtalk is GTalk is a little busted at the moment... see https://issues.asterisk.org/jira/browse/ASTERISK-18301 for more info |
| 17:15.02 | Hive | just curious, what does the _ add? |
| 17:16.56 | p3nguin | Using one underscore allows the variable to be inherited onto one new channel. Using two underscores allows it to be inherited indefinitely. |
| 17:17.06 | p3nguin | Set(__myVar=${CALLERID(num)}) |
| 17:18.09 | *** join/#asterisk DanFromUK (DanFromUK@2.27.26.189) |
| 17:18.15 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 17:18.34 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 17:20.23 | DanFromUK | Hi all. I have a client that wants to be able answer calls, speak to the caller, then transfer the caller to an automated recording (which plays back some terms and conditions), then the call is sent back to the operator where the caller tells the operator that they agree to the terms. The whole thing including the t&cs has to be recorded. |
| 17:20.30 | DanFromUK | Whats the best way of doing this? |
| 17:20.33 | DanFromUK | asterisk 1.4 |
| 17:20.56 | p3nguin | pure dialplan |
| 17:21.37 | DanFromUK | if the dialplan is being used by multiple members of staff, how can i ensure that at the end of the t&c playback, the caller is transfered back to the correct member of staff? |
| 17:23.03 | DanFromUK | is there any way to find out where the caller came from, when a transfer takes place? |
| 17:24.17 | p3nguin | Yes, there is. |
| 17:24.22 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 17:24.39 | DanFromUK | could you give me a clue as to what the variable's called? |
| 17:25.05 | *** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt) |
| 17:25.07 | [sr] | hi |
| 17:25.17 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 17:25.33 | DanFromUK | also, how would I keep the call recording going if the call is transfered? |
| 17:26.45 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 17:27.21 | navaismo | DanFromUK for recording use Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) if you are using mixmonitor |
| 17:28.53 | DanFromUK | thanks. i'll take a look at that. |
| 17:29.40 | DanFromUK | thanks for your help p3nguin |
| 17:30.10 | *** part/#asterisk shtrb (~LENOVO1@unaffiliated/shtrb) |
| 17:30.23 | ChannelZ | or lack thereof |
| 17:32.16 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:32.28 | *** join/#asterisk JCCash (JCCash@200.205.212.102) |
| 17:33.33 | navaismo | <PROTECTED> |
| 17:41.08 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 17:49.27 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 17:52.36 | ChannelZ | navaismo: You might be better off doing your own logic in an AGI, I don't think the time/date ranges aren't flexible enough to do what you want. |
| 17:53.10 | navaismo | Thx |
| 17:54.36 | ChannelZ | At least as a single GotoIfTime anyway |
| 17:56.59 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:57.34 | *** join/#asterisk dogtraineradam (~Adam1983@pool-173-61-159-72.cmdnnj.east.verizon.net) |
| 17:57.48 | dogtraineradam | hello all |
| 17:58.17 | navaismo | "hola" |
| 17:58.23 | dogtraineradam | can anyone help me with getting my cisco 7960 to tell asterisk its available |
| 17:58.25 | dogtraineradam | ? |
| 18:00.14 | ChannelZ | match.com |
| 18:00.44 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 18:01.04 | navaismo | dogtraineradam asterisk see the qualify as OK? |
| 18:02.35 | dogtraineradam | the phones will dial out |
| 18:02.59 | dogtraineradam | but asterisk gives me a chanunavail line in the debug |
| 18:03.21 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 18:03.38 | navaismo | what shows sip show peers |
| 18:03.39 | dogtraineradam | x-lite works out and in (just not with google voice as of yet) |
| 18:04.11 | dogtraineradam | the ips of my two ciscos d for dynamic |
| 18:04.18 | dogtraineradam | 100 and 102 |
| 18:04.18 | ChannelZ | It's too vague a question |
| 18:04.48 | dogtraineradam | the extensions register and are connected |
| 18:04.56 | dogtraineradam | the question is not vague at all |
| 18:05.15 | ChannelZ | Yes it is. What isn't working? What are you trying to do and what is actually happening? |
| 18:05.36 | dogtraineradam | i'm sorry, read above |
| 18:06.09 | ChannelZ | I did, but nevermind, good luck. It could be 50 different things |
| 18:06.29 | dogtraineradam | like what |
| 18:06.37 | dogtraineradam | i think it could only be one thing |
| 18:07.37 | dogtraineradam | and you dont know what setting to change in the sip_notify.conf to tell asterisk to poll the extension cuz its a cisco phone. in the debug it says chanunavail when i call 100 from 102 |
| 18:08.04 | dogtraineradam | so obviously there is a setting to tell asterisk that phone is available |
| 18:08.25 | dogtraineradam | why are people always so snotty when they think they know everything |
| 18:08.28 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 18:08.51 | navaismo | Asterisk see unavailable when not find route to destinattion |
| 18:08.57 | navaismo | destination* |
| 18:09.10 | navaismo | busy and no callwaiting |
| 18:09.27 | dogtraineradam | the route is configured the same as the route for the softphone extension |
| 18:09.35 | navaismo | qualify is activated and the state is unknow or unreacheable |
| 18:09.52 | dogtraineradam | which i can dial from another phone and make it ring |
| 18:10.10 | dogtraineradam | the cisco phones, i cannot call, i get a fast busy right away |
| 18:10.17 | _Corey_ | dogtraineradam: Um, no. qualify != notify |
| 18:11.04 | dogtraineradam | what are you talking about corey |
| 18:11.30 | navaismo | so please make a sip show peers in the asterisk cli and show us the state for 102 or the exten for cisco phone |
| 18:11.45 | _Corey_ | You said you were looking to change sip_notify.conf. I'm suggesting you look elsewhere. |
| 18:12.27 | *** join/#asterisk BenC[UK] (~bcummins@cpc2-lock3-2-0-cust77.6-1.cable.virginmedia.com) |
| 18:12.48 | dogtraineradam | i am trying to perhaps a different place to look, elsewhere is vague |
| 18:12.53 | BenC[UK] | Hi Guys, even though I am specifing Dial(SIP/2345678123,60) in my dial plan.. the call is only ringing for 30 seconds |
| 18:12.57 | *** join/#asterisk mickecarlsson (~Micke@h10n3c1o1101.bredband.skanova.com) |
| 18:13.02 | BenC[UK] | is there somewhere else I should be looking for the max dialtime |
| 18:15.12 | teathsch | BenC[UK]: Dial(SIP/234567812,,60) |
| 18:15.23 | teathsch | or no that's wrong nevermind |
| 18:15.55 | BenC[UK] | was going to say :) |
| 18:16.05 | BenC[UK] | its dial(ext,timeout,options) |
| 18:16.14 | teathsch | are you doing an ami originate into your dialplan and then dialing out? |
| 18:18.05 | ChannelZ | What happens after the 30 seconds? Is it continuing on in the dialplan or..? |
| 18:19.41 | BenC[UK] | teathsch: yeahI am |
| 18:19.51 | BenC[UK] | Channelz - yeah |
| 18:19.52 | teathsch | then you need to set the timeout in the originate |
| 18:19.55 | BenC[UK] | ah |
| 18:20.26 | teathsch | make the timeout in the dial something long like 600 seconds so that it's never shorter than the timeout in the originate |
| 18:21.41 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 18:22.32 | teathsch | or you could do it the other way around and pass a var to the dialplan.. test it both ways to make sure that you get correct responses in OriginateResult or whatever that manager event is called |
| 18:25.07 | ChannelZ | dogtraineradam: No I'm not being snotty because I think I know everything (I do not). Are you getting an 'unable to create channel of type SIP' as well? Have you looked at a SIP debug to see what the device is actually saying? Is DND turned on on the device? There's 3 guesses without having seen any real console info, but you've already told us "it can only be one thing" so I guess I'm wrong. |
| 18:26.21 | BenC[UK] | hmm, I specified it as 600 in the queue |
| 18:26.26 | BenC[UK] | and it hangs calls up right away |
| 18:26.35 | p3nguin | I would have guessed that the phone isn't registered. That makes 4 things. |
| 18:31.34 | Atriks | Is it possible to connect a simple modem card to my server and make it reconized by asterisk ? |
| 18:31.52 | Atriks | and then connect a rj11 between that modem card and my box |
| 18:32.25 | WIMPy | Atriks: Only if you use one of the first Asterisk versions. |
| 18:32.35 | navaismo | i know only the old modems with tiger chipset asterisk view as X100 |
| 18:33.08 | Atriks | I've asterisk 1.6.9.2 |
| 18:33.14 | Atriks | 1.6.2.9* |
| 18:33.52 | Atriks | I saw modem card "for asterisk", wiil it work ? |
| 18:33.58 | Atriks | I'll search that link.. |
| 18:34.05 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 18:34.09 | Atriks | http://cgi.ebay.fr/Authentic-X100P-SE-FXO-PCI-for-Digium-Asterisk-VoIP-PBX-/110645272596#ht_4052wt_1139 here |
| 18:34.51 | Atriks | Not really expensive, but I want to be sure that works |
| 18:35.14 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 18:35.23 | navaismo | maybe works but dont expect a wonderfull card, you may experience alot of echo |
| 18:35.24 | Qwell | Atriks: That is a garbage clone and will not work well at all. |
| 18:35.48 | Atriks | I found some cards, but for 300$... |
| 18:35.58 | Atriks | Like TDM410 |
| 18:36.49 | ChannelZ | You could get a SPA3102 for ~$70 |
| 18:36.58 | Atriks | too expensive for me :x |
| 18:37.16 | ChannelZ | Well you get what you pay for. |
| 18:37.45 | Atriks | :| |
| 18:37.54 | navaismo | nice phrase |
| 18:38.18 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 18:38.32 | navaismo | but its true Atriks, I have one of those modems cost more or less $25USD |
| 18:38.47 | navaismo | but the echo is present even with fxotune |
| 18:39.01 | Atriks | it works ? |
| 18:39.04 | navaismo | sometimes not recognizes the hangups |
| 18:39.51 | navaismo | not for production is for my test in my house |
| 18:40.13 | Tim_Toady | most of echo can be cured with oslec and for the hangups use this in ur conf: busydetect=yes |
| 18:41.04 | navaismo | yep the busydetect is on |
| 18:41.15 | navaismo | and the count in 3 but its a modem |
| 18:41.33 | navaismo | not a fxo card |
| 18:42.12 | Tim_Toady | for a home pbx x100 card is 'okay' |
| 18:42.20 | WIMPy | What kind of an argument is that? Modems used to have very reliable busy detection. |
| 18:42.26 | Atriks | So a simple modem card could work ? |
| 18:42.47 | Tim_Toady | it gets shit done (TM) |
| 18:43.10 | *** join/#asterisk nightrid3r (~kvirc@91.176.74.218) |
| 18:43.19 | navaismo | my simple card modem have tiger chipset im not testing others |
| 18:44.08 | WIMPy | Atriks: It might work to some extent, but it's unlikely to be fun. |
| 18:44.32 | Atriks | erf |
| 18:44.40 | Atriks | and that card "designed for asterisk" ? :/ |
| 18:45.02 | navaismo | I doubt that card was designed for asterisk |
| 18:45.20 | navaismo | designed for asterisk is DIgium or Sangoma |
| 18:45.35 | Atriks | digium is expensive |
| 18:45.39 | WIMPy | POTS is never much fun. |
| 18:45.43 | Tim_Toady | there are some sangoma cards far worse than the x100 xlones :P |
| 18:45.54 | navaismo | in the last case Openvox Chinese clone |
| 18:46.04 | *** join/#asterisk beardy (~beardy@unaffiliated/beardy) |
| 18:46.06 | WIMPy | But at least the right cards should work well with Asterisk. |
| 18:46.19 | navaismo | personally i hate wanpipe |
| 18:46.25 | Tim_Toady | who doesnt |
| 18:46.30 | Atriks | how much cost a good card with one port ? |
| 18:46.37 | WIMPy | Get an ITSP and forget about the POTS stuff. |
| 18:47.03 | Atriks | I've an ITSP.. |
| 18:47.19 | Atriks | But I can't call mobiles with |
| 18:47.30 | Atriks | that's why I'm looking for a pci card |
| 18:47.32 | WIMPy | Then get another. |
| 18:47.35 | Atriks | no |
| 18:47.40 | Atriks | Mine is free |
| 18:47.55 | Atriks | I mean, with my internet subscription |
| 18:47.58 | navaismo | ITSP=sip trunk? |
| 18:48.01 | WIMPy | Then get another free one. |
| 18:48.08 | Atriks | <Atriks> I mean, with my internet subscription |
| 18:48.11 | WIMPy | ~siptrunk |
| 18:48.11 | infobot | siptrunk is probably something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 18:48.16 | WIMPy | ~itsp |
| 18:48.16 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 18:48.27 | ChannelZ | ~boobs |
| 18:48.27 | infobot | boobs are awesome |
| 18:48.49 | ChannelZ | he knows everything |
| 18:48.55 | navaismo | LOL |
| 18:48.58 | navaismo | thx |
| 18:49.04 | WIMPy | ~42 |
| 18:49.04 | infobot | i heard 42 is the answer to life the universe and everything, see also http://en.wikipedia.org/wiki/the_answer_to_life,_the_universe,_and_everything |
| 18:49.25 | Atriks | http://answertolifetheuniverseandeverything.com/ |
| 18:50.25 | [sr] | hi WIMPy |
| 18:51.04 | WIMPy | [sr]: salve! |
| 18:52.40 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-088-065-179-178.pools.arcor-ip.net) |
| 18:58.04 | *** join/#asterisk wizbit (wiz@newelite1.bshellz.net) |
| 18:58.18 | wizbit | ive compiled the dhadi kernel module and tools but cannot get meetme working in 1.8 |
| 18:58.47 | *** join/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 18:58.53 | *** part/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 19:00.57 | ChannelZ | Can you elaborate 'not working'? |
| 19:04.48 | navaismo | ~hungry |
| 19:04.48 | infobot | it has been said that hungry is http://www.thehungersite.com/ or http://www.thehungersite.com/cgi-bin/WebObjects/HungerSite or http://www.hungryminds.com/ |
| 19:05.42 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 19:15.02 | wizbit | ChannelZ: [Aug 25 20:14:36] WARNING[21004]: app_meetme.c:1220 build_conf: Unable to open DAHDI pseudo device |
| 19:15.13 | wizbit | ChannelZ: it worked fine in 1.6 |
| 19:15.48 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v000-154.mobile.uci.edu) |
| 19:16.15 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 19:17.20 | WIMPy | wizbit: 'timing test' |
| 19:18.01 | wizbit | i think i need to start dahdi in the background |
| 19:18.06 | wizbit | WIMPy: ? |
| 19:18.25 | WIMPy | Type that at the *CLI |
| 19:18.37 | wizbit | dahdi_test |
| 19:18.37 | wizbit | Unable to open dahdi interface: No such file or directory |
| 19:18.43 | wizbit | ok |
| 19:19.05 | wizbit | It has been 1000 milliseconds, and we got 50 timer ticks |
| 19:19.16 | WIMPy | Looks like it's not started at all. |
| 19:19.27 | wizbit | hmm |
| 19:19.37 | WIMPy | But "using the" != DAHDI? |
| 19:20.03 | wizbit | god knows how it was working in 1.6 |
| 19:20.07 | wizbit | i didnt have a rc.dhadi |
| 19:20.09 | wizbit | and still dont |
| 19:20.23 | wizbit | im not sure what i need to start |
| 19:21.02 | WIMPy | Without hardware a 'modprobe dahdi' should be enough, I think. |
| 19:21.29 | wizbit | captian obvious! |
| 19:21.35 | wizbit | <--- total idiot |
| 19:43.44 | Karen_m | When I dial someone from exten => 123,n,Dial(SIP/voipms/1403...), is there a way to send a dynamic caller-id to myself? When I receive calls, and forward to myself it sends me the caller id of voip phone, but no idea who is calling |
| 19:44.02 | Karen_m | I want to track who is calling, maybe in the 'name' field, have it show : office - 403.123.1233 |
| 19:44.04 | Karen_m | kind of thing |
| 19:49.44 | jayson_r | i have an astersisk box that when i go to the cli it acts like it's passing "core show" before every command automatically |
| 19:49.48 | jayson_r | ever seen anything like this? |
| 19:56.24 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:59.51 | Qwell | jayson_r: the command is "core show help" |
| 20:01.01 | *** join/#asterisk sulex (~sulex@pdpc/supporter/professional/sulex) |
| 20:06.42 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:08.37 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 20:09.09 | pabelanger | directmedia / canreinvite question people. Assume the following: phoneA <--> AsteriskA <--> AsteriskB <--> phoneB (all SIP). Has anybody ever tried to see if RTP will be establish directly between phoneA and phoneB |
| 20:09.22 | pabelanger | local LAN too |
| 20:10.38 | anonymouz666 | well, why not? |
| 20:10.51 | anonymouz666 | there are some configs that will prevent you from doing this |
| 20:11.23 | pabelanger | anonymouz666: for example? |
| 20:11.48 | anonymouz666 | tTwW, nat=yes ... |
| 20:11.58 | anonymouz666 | MixMonitor etc. |
| 20:12.41 | pabelanger | assume you will not be using any of them.... a simple dial from phoneA to phoneB |
| 20:14.20 | anonymouz666 | already did that |
| 20:14.45 | anonymouz666 | it could be done with re-invite or no. |
| 20:14.51 | anonymouz666 | not |
| 20:15.16 | pabelanger | anonymouz666: Have you tried this setup? |
| 20:15.19 | *** part/#asterisk wizbit (wiz@newelite1.bshellz.net) |
| 20:15.38 | anonymouz666 | exactly this, no. |
| 20:15.57 | anonymouz666 | but I can't find a reason right now that this setup won't work |
| 20:17.05 | pabelanger | That is what I am looking for, the reason; why this will or will not work. |
| 20:21.05 | *** join/#asterisk mocker (~mocker@206.55.118.84) |
| 20:27.35 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:31.07 | *** join/#asterisk devmikey (~irc@96.46.249.230) |
| 20:33.03 | *** join/#asterisk thegoat (~thegoat@c-71-224-170-221.hsd1.pa.comcast.net) |
| 20:36.37 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:45.01 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 20:45.46 | bmoraca_work | has anyone here successfully implemented BLF (hints) across servers? such as, endpoint monitors a hint on server1 which is monitoring a hint on server2 |
| 20:46.30 | anonymouz666 | you mean distribute device state? |
| 20:46.52 | bmoraca_work | yes |
| 20:46.57 | bmoraca_work | devstate |
| 20:46.59 | bmoraca_work | that's what it's called |
| 20:47.02 | bmoraca_work | dur |
| 20:47.15 | anonymouz666 | devstate is one thing, distributed devstate other |
| 20:49.42 | BenC[UK] | ahhh, was in the AMI call.. |
| 20:49.45 | BenC[UK] | I had 60 |
| 20:49.47 | BenC[UK] | and then 600 |
| 20:49.50 | BenC[UK] | but its in MS! |
| 20:50.48 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 20:52.27 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:56.36 | *** join/#asterisk dailylinux (~test@88.87.48.12) |
| 20:57.39 | *** join/#asterisk bbryant1 (~brett@c-174-56-132-225.hsd1.sc.comcast.net) |
| 21:00.27 | *** join/#asterisk JCCash (~JCash@200.205.212.102) |
| 21:00.57 | bmoraca_work | hmm, devstate with AIS looks really straightforward and easy |
| 21:01.12 | bmoraca_work | and should be very scalable as it's based on multicast |
| 21:01.13 | anonymouz666 | you are correct. |
| 21:01.22 | anonymouz666 | but... |
| 21:01.36 | anonymouz666 | res_ais deadlocks here... |
| 21:01.43 | bmoraca_work | oh? |
| 21:02.04 | bmoraca_work | there's no fix? |
| 21:02.17 | anonymouz666 | until now, no. |
| 21:02.27 | anonymouz666 | maybe you have lucky and works just fine for you. |
| 21:02.32 | anonymouz666 | here it does not |
| 21:02.39 | *** part/#asterisk didnot (~didnot@unaffiliated/didnot) |
| 21:02.41 | bmoraca_work | what about XMPP? |
| 21:02.51 | anonymouz666 | I didn't like XMPP approach |
| 21:02.56 | anonymouz666 | too bloated |
| 21:03.02 | anonymouz666 | sux to setup |
| 21:03.10 | anonymouz666 | i think it is slow, etc etc |
| 21:03.28 | bmoraca_work | ahh |
| 21:03.55 | anonymouz666 | I would like to hear if res_ais works for you |
| 21:04.13 | bmoraca_work | well, i'm a ways from implementation |
| 21:16.37 | *** join/#asterisk wizbit (wiz@newelite1.bshellz.net) |
| 21:16.59 | wizbit | what is the quickest way to find out when a new version of asterisk is released or security advisories? |
| 21:18.00 | Kobaz | the mailing list |
| 21:18.06 | wizbit | ace |
| 21:18.32 | wizbit | http://lists.digium.com/mailman/listinfo/asterisk-security <--- here? |
| 21:19.54 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 21:21.40 | pabelanger | wizbit: yes |
| 21:22.04 | pabelanger | that list will only contain security advisories. |
| 21:22.16 | wizbit | ace |
| 21:22.38 | pabelanger | see http://lists.digium.com/mailman/listinfo/asterisk-announce for all releases |
| 21:23.13 | wizbit | ace |
| 21:25.28 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 21:30.52 | *** join/#asterisk mhaddog (~mhaddog@z65-50-116-231.ips.direcpath.com) |
| 21:34.09 | BenC[UK] | guys, is there anyway to list agents who are in "wrapuptime" ? |
| 21:34.14 | BenC[UK] | I have wrapuptime set to 0 |
| 21:34.42 | BenC[UK] | a queue, with strategy least recent, a Queue(7001,n,,,10) |
| 21:34.46 | BenC[UK] | but I am still timing out |
| 21:34.49 | navaismo | so no wait time |
| 21:38.14 | *** part/#asterisk wizbit (wiz@newelite1.bshellz.net) |
| 21:40.58 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 21:41.43 | *** join/#asterisk DrDigi (~mmurphy@50-73-49-97-static.hfc.comcastbusiness.net) |
| 21:53.06 | BenC[UK] | http://pastebin.com/zKWLKiab |
| 21:53.11 | BenC[UK] | this is the config for my queue |
| 21:53.17 | BenC[UK] | I have 10 agents sitting there doing nohting |
| 21:53.41 | BenC[UK] | but keeps missing agents in the queue |
| 21:53.55 | BenC[UK] | calls go into Queue() then carry on after without hitting an agent |
| 21:54.36 | navaismo | the queue show command what show and what is the call flow when calls arrives can you pastebin too? |
| 21:56.28 | BenC[UK] | http://pastebin.com/fkrXJW8M |
| 21:57.57 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-hopzaituklykzkaa) |
| 21:58.33 | BenC[UK] | Sorry, what else did you want to see? |
| 21:58.41 | BenC[UK] | I can show you my crappy dialplan if you like |
| 21:59.37 | navaismo | the output of the cli when a call get into the queue |
| 22:00.48 | BenC[UK] | http://pastebin.com/Bb2Sbhrd |
| 22:01.00 | BenC[UK] | I have a fair bit of agi in there at the moment, logging statuses |
| 22:04.53 | navaismo | In the cli i never see the parto of sip/xxx ringing |
| 22:05.32 | navaismo | part* |
| 22:06.33 | *** part/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 22:07.33 | navaismo | try removong manually one agent per example 2001 with queue remove member SIP/2001 from 7001 |
| 22:07.49 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 22:08.04 | navaismo | and then add manually again with queue add memeber SIP/2001 to 7001 and make a call again |
| 22:09.43 | WIMPy | Hmm. I always thought that SIP was the mature part of Asterisk. |
| 22:13.39 | BenC[UK] | navaismo: ok, if I do that.. what would happen? |
| 22:15.06 | navaismo | I want to see if your device sip/2001 rings |
| 22:21.01 | BenC[UK] | http://pastebin.com/ikRrU2qN |
| 22:21.05 | BenC[UK] | indeed you do see it there |
| 22:22.39 | BenC[UK] | it does find an agent most of the time though |
| 22:23.33 | navaismo | Now is ringing in this http://pastebin.com/Bb2Sbhrd I not see when start ring, and 10 sec its like 3 rings right |
| 22:24.55 | BenC[UK] | I have the agents on auto answer |
| 22:25.26 | BenC[UK] | if they dont answer within 10 minutes, I drop a precorded message and go on to the next call |
| 22:38.50 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 22:41.48 | navaismo | mmmm so i dont have much to say/help i was thinking the problem is the agents in the queue but not |
| 22:44.41 | *** join/#asterisk sunfone (~jeff@c-67-184-36-158.hsd1.il.comcast.net) |
| 22:57.01 | BenC[UK] | nah, they randomly get calls |
| 22:57.24 | BenC[UK] | in the last 10 minutes, I have had two queue timeouts |
| 22:57.35 | BenC[UK] | I have 9 of 20 agents on the phone |
| 22:57.53 | BenC[UK] | makes no sense |
| 22:58.15 | BenC[UK] | I am wondering if I should try a different strategy |
| 22:58.32 | navaismo | maybe rrmemory |
| 23:00.41 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 23:01.04 | BenC[UK] | ok, lets try that :) |
| 23:01.15 | BenC[UK] | nope, 11, then 15 |
| 23:01.25 | BenC[UK] | now 5 |
| 23:01.34 | BenC[UK] | then 9 and 19 |
| 23:01.38 | BenC[UK] | then 12 |
| 23:01.38 | BenC[UK] | :S |
| 23:02.47 | navaismo | :S |
| 23:03.53 | BenC[UK] | is there a way to reload the queue |
| 23:04.40 | navaismo | queue reload |
| 23:04.54 | BenC[UK] | will that reset the agent statses? |
| 23:05.02 | navaismo | i dont know |
| 23:13.19 | *** join/#asterisk Rienzilla (rien@sinas.rename-it.nl) |
| 23:13.49 | Rienzilla | hey, is it possible to dial an extension from the asterisk console? |
| 23:14.02 | navaismo | use originate |
| 23:14.45 | Rienzilla | was that directed to me? |
| 23:14.56 | navaismo | yep |
| 23:15.23 | Rienzilla | ah, thanks |
| 23:15.28 | Rienzilla | that was indeed what I was looking for |
| 23:17.58 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:24.00 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 23:24.15 | navaismo | see you |
| 23:25.09 | *** join/#asterisk penguin (greenwolf@cpe-74-77-221-5.buffalo.res.rr.com) |
| 23:32.09 | *** join/#asterisk drynish (~drynish@modemcable016.169-21-96.mc.videotron.ca) |
| 23:51.50 | *** join/#asterisk passthru (~ramon@201009176059.user.veloxzone.com.br) |
| 23:59.31 | *** join/#asterisk drynish (~drynish@modemcable016.169-21-96.mc.videotron.ca) |
| 23:59.47 | drynish | I have an analog card x100p that I have correctly configured to don't answer on polarity reverse, when I'm answering the call before the answering machine. This is great :) |