| 00:01.52 | *** join/#asterisk lucasb (~lucasb@S0106000c42710923.ok.shawcable.net) |
| 00:09.04 | Schreiber1337 | Can someone school me in SLA and SEA? |
| 00:09.10 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:09.36 | carrar | SEA == Seattle |
| 00:09.42 | carrar | SLA = Service Level Agreement |
| 00:10.10 | Schreiber1337 | Shared Line Appearance / Shared Extension Appearance |
| 00:13.25 | Schreiber1337 | If I am correct Shared Line Appearance means several phones can ring/answer/see if the line is in use on a Trunk line only... Is that correct? |
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| 00:20.53 | Schreiber1337 | Anyone willing to have a conversation on SLA / SEA / BLA ..... |
| 00:21.58 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:21.58 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 00:45.28 | DrDamnit | where is the documentation for exten => same? |
| 00:46.32 | DrDamnit | Foudn it. |
| 00:46.32 | leifmadsen | ~thebook |
| 00:46.32 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ |
| 00:46.41 | leifmadsen | same => n,NoOp() |
| 00:46.53 | leifmadsen | not exten => same,n,NoOp() |
| 00:46.54 | leifmadsen | fyi |
| 00:47.05 | DrDamnit | leifmadsen: Thanks. I was foolishly doing: exten => same for some reason. Brain Fart. |
| 00:47.32 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 00:47.49 | leifmadsen | heh |
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| 00:53.38 | Schreiber1337 | Does 1.8 support "shared extension appearance" or "bridged line appearance" |
| 00:55.55 | leifmadsen | yes |
| 00:56.14 | leifmadsen | it's called Shared Line Appearance (SLA) |
| 00:56.23 | leifmadsen | documented on the OFPS website |
| 00:57.04 | Schreiber1337 | @leifmadsen: From what I've read SLA is only for multiple phones monitoring a single Trunk channel... is that correct? |
| 00:58.35 | Schreiber1337 | @leifmadsen: I'm looking for the ability of multiple phones to ring/answer/see state of an extension number... |
| 01:00.39 | leifmadsen | you'll have to check the documentation -- I didn't write that section |
| 01:04.13 | *** join/#asterisk waterfoul (~chatzilla@67.129.121.92) |
| 01:04.35 | waterfoul | i can make a call through but i hear no audio through google voice, what did i do wrong? |
| 01:06.50 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
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| 01:19.01 | drfreeze | Looks like 1.8.4-rc2 fixes the bug in 1.8.3.2 |
| 01:20.13 | waterfoul | which? |
| 01:20.43 | Schreiber1337 | If more than one phone uses the same SIP User ID, how does Asterisk know which one to send incoming calls to? The last one to register? |
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| 01:47.17 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 02:32.38 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 02:33.23 | a1fa | so i need to build an agi to take an imput and query a remote site with it .. thinking about php |
| 02:34.05 | a1fa | any pre-built frameworks i can look at? |
| 02:36.54 | agroman | better off picking a lang you are familiar with. the last framework I heard of was ragi and it's not been updated in quite some time. |
| 02:37.34 | agroman | could be wrong... |
| 02:37.35 | agroman | http://www.voip-info.org/wiki/view/Asterisk+AGI |
| 02:37.38 | agroman | for more |
| 02:37.53 | seraphie | a1fa: for python, StarPy works for me, and there is the Asterisk::AGI Perl module, but I am not familiar with it. |
| 02:41.00 | a1fa | i am going with php |
| 02:41.02 | a1fa | :) |
| 02:41.11 | a1fa | it looks easy |
| 02:41.17 | a1fa | hopefully i'll be able to make SOP calls with php |
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| 03:05.12 | pabelanger | +1 starpy |
| 03:05.28 | \DSAFEW\ | what's it called when I want my dialplan to forward calls to the one extension I have? |
| 03:05.37 | \DSAFEW\ | is that a transfer? Dial? |
| 03:06.29 | pabelanger | \DSAFEW\: You want to know which application to use? |
| 03:06.59 | \DSAFEW\ | pabelanger, I'm looking for help with my dial plan, the demo has something maybe similar |
| 03:07.10 | \DSAFEW\ | ;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer |
| 03:07.15 | \DSAFEW\ | not sure what that would do |
| 03:07.43 | pabelanger | If you Dial() something, asterisk still have call control, if you Transfer(), asterisk releases call control |
| 03:07.56 | pabelanger | So, depends what you want to do |
| 03:08.26 | pabelanger | Dial() is a bridge |
| 03:10.32 | \DSAFEW\ | okay, so if I wanted to use a one-line FXO card for PSTN and one connected SIP phone, making asterisk transparent |
| 03:10.42 | \DSAFEW\ | I would use Dial to change the protocol? |
| 03:11.04 | pabelanger | yes |
| 03:11.24 | pabelanger | because you are bridging to 2 different protocol |
| 03:12.46 | \DSAFEW\ | so how exactly do I catch everything SIP dials and make it Dial out? |
| 03:13.10 | pabelanger | using contexts |
| 03:13.13 | \DSAFEW\ | and how would I configure the incoming calls to go to that extension? |
| 03:13.23 | pabelanger | When you define a sip peer, you will assign it a context. |
| 03:13.37 | \DSAFEW\ | right |
| 03:13.37 | pabelanger | same goes for your incoming FXO |
| 03:13.41 | \DSAFEW\ | and in its context I put... |
| 03:13.42 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 03:13.54 | \DSAFEW\ | one second I'll pastebin my dialplan |
| 03:15.45 | \DSAFEW\ | here's the extensions.conf http://paste.pocoo.org/show/371958/ |
| 03:17.33 | \DSAFEW\ | my dahdi is configured to context=from-pstn and sip is from-sip |
| 03:17.59 | *** join/#asterisk cVsup (~cVsup@189.83.184.198) |
| 03:18.33 | pabelanger | Okay, looks fine. So if you wanted your sip phone to dial 10 digits, you add exten => _NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN}) |
| 03:18.40 | pabelanger | in from-sip |
| 03:19.57 | \DSAFEW\ | pabelanger, so the _ counts as a number or not a number? |
| 03:20.48 | \DSAFEW\ | is this magic _NXXNXXXXXX a catch-all pattern for 9-10 digit phone numbers? or does it need to be a 10 digit? |
| 03:20.54 | *** join/#asterisk Quant (~QuantB@ool-45765d5c.dyn.optonline.net) |
| 03:21.20 | \DSAFEW\ | err, I guess what I meant was 10-11 |
| 03:21.49 | pabelanger | \DSAFEW\: Yes, that will only match a 10 digit pattern. X is 0-9, N is 2-9 |
| 03:21.58 | \DSAFEW\ | it's not going to do the 1 in a 1800? I can add another string to match them then? |
| 03:22.25 | pabelanger | \DSAFEW\: Correct, just add a new pattern |
| 03:22.36 | pabelanger | if you look at extension.conf.sample it shows you how |
| 03:22.46 | \DSAFEW\ | pabelanger, you're a real help, I don't think there's any documentation on this simple syntax, it's been driving me nuts groping in the dark like this |
| 03:22.47 | pabelanger | you can also #include other contexts |
| 03:22.57 | pabelanger | ~book |
| 03:22.57 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 03:23.04 | pabelanger | \DSAFEW\: ^ Great resource |
| 03:23.30 | \DSAFEW\ | been reading this here http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-4-SECT-7.html |
| 03:23.45 | \DSAFEW\ | maybe I somehow missed that part |
| 03:24.28 | pabelanger | \DSAFEW\: http://ofps.oreilly.com/titles/9780596517342/ |
| 03:24.33 | pabelanger | the latest and greatest release |
| 03:24.38 | pabelanger | Just release, this month |
| 03:28.20 | \DSAFEW\ | in case you're wondering why the monkeys sound is there, Answer() wasn't working for some reason, possibly the dahdi config defaulting to the wrong context before |
| 03:28.27 | \DSAFEW\ | testing, brb |
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| 03:34.53 | \DSAFEW\ | this is a good book |
| 03:34.57 | \DSAFEW\ | thanks pabelanger |
| 03:35.27 | pabelanger | \DSAFEW\: send beers to russellb and leifmadsen |
| 03:35.36 | \DSAFEW\ | I will. |
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| 04:27.12 | teathsch | i got a voip onion network up and running (for making anonymous calls) with 4 whole nodes lol... http://jailcity.com/voiponion/ .. looking for feedback |
| 04:27.44 | *** join/#asterisk izhak (~izhak@snms.e-tagil.ru) |
| 04:32.06 | pabelanger | annonymous to a point, your originating IP information would be in the SIP INVITE message, no? |
| 04:32.32 | teathsch | anonymous in the sense that you can't link sender and receiver |
| 04:32.55 | pabelanger | how is RTP established? |
| 04:33.35 | teathsch | you can only use sip to an entry node or out an exit node.. between relays it is all iax |
| 04:33.54 | teathsch | reinvites are fatal to anonymity |
| 04:34.59 | Wiretap | this is a cool idea |
| 04:35.42 | *** join/#asterisk waterfoul (~chatzilla@67.129.121.92) |
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| 04:36.06 | waterfoul | I keep getting presence packet errors, how can i handle these? |
| 04:36.57 | Wiretap | with some duct tape, rubber gloves and pliers? |
| 04:37.11 | Wiretap | that is insufficient information to solve your problem |
| 04:37.20 | Wiretap | pastebin logs and relevant sections of config |
| 04:37.47 | kaldemar | waterfoul: errors or notices? |
| 04:37.53 | waterfoul | notice |
| 04:38.02 | waterfoul | [Apr 14 22:37:27] NOTICE[6641]: res_jabber.c:2282 aji_handle_presence: Got presence packet from msn.jabber.hot-chilli.net, someone not in our roster!!!! |
| 04:38.21 | waterfoul | the address varies, i knowwhere they are comming from i just on't know how to handle them |
| 04:43.46 | teathsch | i say this with no experience with jabber, but the message seems self-explanatory.. but how is it problematic? |
| 04:44.21 | waterfoul | i just want to handle the packets but don't know how to configure it |
| 04:44.55 | waterfoul | its not necessary i would just like to handle them |
| 04:46.30 | *** join/#asterisk felimwhiteley (~quassel@109.255.104.145) |
| 04:47.19 | waterfoul | is there a full list of jabber.conf settings? |
| 04:50.02 | waterfoul | found it, you need buddy=.... lines |
| 04:51.12 | waterfoul | where does it get the info when calling getaddrinfo at the startup |
| 04:53.23 | *** join/#asterisk Maxus2 (~Maxus@59.191.225.49) |
| 04:54.48 | Maxus2 | hi Asterisk people, is there a way of call multiple devices one after if they fail in one dial plan line? similar to the dial(sip/X&sip/Y) but not failing over rather than calling all at once? |
| 04:56.05 | Maxus2 | never mind, i think retry dial does it. |
| 04:58.15 | *** join/#asterisk benngard (~mabe@213.88.138.230) |
| 04:59.42 | Maxus2 | nope apprently it doesn't :( |
| 05:03.24 | teathsch | Maxus2: Dial(...||G) .. then check for ${DIALSTATUS}=CONGESTION |
| 05:03.30 | teathsch | ||g rather |
| 05:03.48 | Maxus2 | Hi teathsch, but will that work ina single line? |
| 05:04.18 | teathsch | no you'd have to make a macro or a custom context |
| 05:04.44 | teathsch | you also want to check for CHANUNAVAIL |
| 05:05.10 | Maxus2 | cant do that, im inside realtime, and need to return the result it and the result of an odbc call |
| 05:05.20 | Maxus2 | and = as |
| 05:06.28 | Maxus2 | yeah was hoping for something simple like dial(SIP/X&SIP/Y, param) where if X fails it then dials Y |
| 05:08.07 | Maxus2 | i could do a nested if(Chanunavail(sip/X), dial(SIP/X), if(Chanunavail(sip/Y), dial(SIP/Y))) but i will run out of characters pretty quickly |
| 05:11.21 | Maxus2 | can i do mutiple applications in one like: dial(sip/X)&dial(sip/Y)? |
| 05:23.07 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
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| 05:42.58 | tehrabbitt | i feel stupid but I just noticed sip isn't loaded in my freshly compiled asterisk install.... am I missing something? |
| 05:43.55 | tehrabbitt | when I mean by it's not loaded, I type "SIP" in the cli and hit tab but it doesn't find sip, just sccp, skinny, say and stun |
| 05:44.39 | tehrabbitt | am I missing a step? |
| 05:44.49 | Maxus2 | i had that once, i just rebooted asterisk and it was there |
| 05:45.16 | tehrabbitt | i've tried that :-\ no luck |
| 05:45.58 | tehrabbitt | i'm tempted to see if recompiling / reinstalling will work but i wanna make sure there's nothing I should try first |
| 05:48.09 | tehrabbitt | now asterisk won't even load :-\ |
| 05:48.14 | tehrabbitt | i really screwed something up now :-\ |
| 05:49.20 | kaldemar | tehrabbitt: i was about to ask if "module show like sip" lists chan_sip.so, but since asterisk doesn't start... start it with -vvvvvc and see where it crashes. |
| 05:49.28 | tehrabbitt | alright |
| 05:49.39 | kaldemar | tehrabbitt: did you have a previous version installed on the machine? |
| 05:50.16 | tehrabbitt | nope it's fresh reformat |
| 05:50.41 | tehrabbitt | http://pastebin.com/aprMDgCd |
| 05:50.44 | tehrabbitt | theres where it's crashing :-]\ |
| 05:50.46 | tehrabbitt | :-\ |
| 05:51.06 | *** join/#asterisk cellkill (~kvirc@2001:470:c469:beef:1d46:c5ff:5d32:db30) |
| 05:52.03 | tehrabbitt | <PROTECTED> |
| 05:52.04 | tehrabbitt | Segmentation fault |
| 05:52.20 | tehrabbitt | idk what "ael-demo" is supposed to be but that's not mine 0_o |
| 05:52.42 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 05:54.05 | kaldemar | tehrabbitt: it is a context in the sample extensions.ael. |
| 05:54.17 | tehrabbitt | so delete extensions.ael? |
| 05:54.57 | tehrabbitt | or is it because SIP isn't working its complaining? |
| 05:57.41 | tehrabbitt | I just recompiled to see if that brings SIP back and maybe then everything will work... it's almost done already lol |
| 05:57.49 | kaldemar | no clear indication of an issue in your paste. |
| 05:58.26 | kaldemar | i've never seen extensions.ael cause a crash. |
| 05:58.56 | tehrabbitt | the paste was made using '-vvvvvvc' and the above inline paste i sent in regard to ael-demo was using '-vvvvvvvvvvvvvc' |
| 05:58.57 | tehrabbitt | lol |
| 05:59.10 | tehrabbitt | kaldemar, thats why i'm thinking it's in regard to SIP missing |
| 05:59.28 | tehrabbitt | which maybe it never compiled somehow :-\... I checked make menuconfig this time and it was selected so i'm not sure |
| 06:01.03 | tehrabbitt | should I run "make samples" or just use my actual .conf files I already have made? |
| 06:01.07 | kaldemar | do you have chan_sip.so in /usr/lib/asterisk/modules? |
| 06:01.15 | tehrabbitt | kaldemar, it wasn't there before, no |
| 06:01.25 | tehrabbitt | hence why I think maybe thats what was causing the seg fault |
| 06:01.41 | tehrabbitt | dont know how that would happen though :-\ |
| 06:01.43 | kaldemar | if you had chan_sip selected in the menu and didn't have the module, then there was something wrong with the compilation. |
| 06:02.13 | tehrabbitt | well I did just recompile / reinstall, so hopefully this time all is good :)... should I run "Make samples" or should I just work from scratch? |
| 06:02.54 | kaldemar | configuration files have been known to cause crashes in the past, even the samples. try with your configs first, then with samples if it doesn't start or give a clear indication on what is to blaim. |
| 06:03.24 | tehrabbitt | wait a minute.... still no SIP |
| 06:03.27 | tehrabbitt | and it's there now 0_o |
| 06:03.51 | tehrabbitt | and I checked modules.conf and it's not marked *not* to load so idk |
| 06:04.56 | tehrabbitt | http://pastie.org/1796427 |
| 06:05.05 | tehrabbitt | i copied basically everything on my screen |
| 06:06.18 | kaldemar | what does "module load chan_sip.so" tell you? |
| 06:06.36 | tehrabbitt | kaldemar, looks like a bad config file :-\ |
| 06:06.45 | tehrabbitt | i loaded samples, everything is there |
| 06:07.07 | tehrabbitt | maybe it was a corrupted config from when I compiled samples earlier and it left SIP out or such? |
| 06:07.17 | kaldemar | let me guess, an empty [authentication] in sip.conf? |
| 06:09.13 | tehrabbitt | I'm thinking you might actually be right on target here... I load my SIP.conf back in, and bam, back to messed up :-\ |
| 06:09.56 | tehrabbitt | what do yo umean by empty authetication though? |
| 06:09.58 | tehrabbitt | like a missing username? |
| 06:10.00 | tehrabbitt | or password? |
| 06:10.16 | kaldemar | i was being literal.. |
| 06:11.11 | tehrabbitt | I need to add [authentication] to my sip.conf? |
| 06:11.20 | kaldemar | some version, i don't remember which one, crashed with an empty [authentication] context in the file long time ago. |
| 06:11.21 | tehrabbitt | i've never seen that in my previous conf files before :-\ |
| 06:11.33 | tehrabbitt | hm |
| 06:11.51 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 06:12.30 | tehrabbitt | ah I see now 0_o |
| 06:12.33 | tehrabbitt | i feel stupid :=\ |
| 06:12.34 | tehrabbitt | lol |
| 06:14.50 | tehrabbitt | still nothing :( |
| 06:14.58 | tehrabbitt | kaldemar, do I need to reload the module or something? |
| 06:15.22 | tehrabbitt | ah [Apr 15 02:15:04] WARNING[23698]: config.c:1102 process_text_line: parse error: No category context for line 1 of /etc/asterisk/sip.conf |
| 06:16.00 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 06:16.20 | kaldemar | missing [general]? |
| 06:16.25 | tehrabbitt | nope even worse... |
| 06:16.26 | tehrabbitt | :-\ |
| 06:16.35 | tehrabbitt | used "//" as comments isntead of ";" |
| 06:16.56 | tehrabbitt | soooo it wouldn't read the conf :-\ |
| 06:18.28 | tehrabbitt | so here's my second question, it seems to be working now, for the exception I noticed "sip show registry" doesn't show any registered SIP peers, even though I do have two registration lines within my .conf file |
| 06:19.41 | kaldemar | sip show registry isn't supposed to show any registered peers, it shows where your asterisk tries to register. |
| 06:19.51 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 06:20.00 | kaldemar | i.e. outgoing registrations from asterisk's point of view. |
| 06:20.23 | kaldemar | sip show peers is what you were thinking of. |
| 06:20.49 | tehrabbitt | ah :-D |
| 06:22.34 | tehrabbitt | kaldemar, hm, noticed under status it shows "unmonitored" i'm guessing there's a way to set it up so it monitors if it's online or offline? |
| 06:26.09 | tehrabbitt | kaldemar, ah, "qualify=yes" did it :-D sweet, for once i'm actually taking my time setting this up so that way it's nice and neat and organized .conf files and not a mess to jumble through lol |
| 06:26.52 | kaldemar | see also the other qualify parameters to tune how it works. it can be used as a keep-alive for example. |
| 06:27.34 | tehrabbitt | ah, cool :)... i'm guessing that's for dynamic IP or such? |
| 06:28.47 | kaldemar | more like clients behind NAT routers or in WLAN access points that drop connections too quickly. |
| 06:32.26 | *** join/#asterisk eerie (~mime@gateway/shell/bshellz.net/x-bggsypbynxaxzkci) |
| 06:33.54 | tehrabbitt | Ah that makes sense |
| 06:34.51 | tehrabbitt | kaldemar, if I have two SIP trunks, and i have two SIP phones, I can just assign each device to it's own context (Personal/Business) and it'll use the proper outgoing / incoming trunk, right? |
| 06:35.26 | kaldemar | if a client IP address changes, the client has to re-register. that can be done with expiry settings. |
| 06:36.08 | kaldemar | tehrabbitt: yes. you can do pretty much what ever you want. |
| 06:37.16 | tehrabbitt | so do [personalInbound] and list my incoming route for my personal line within that, correct? |
| 06:37.30 | tehrabbitt | then do a seperate [BusinessInbound] |
| 06:37.39 | tehrabbitt | or can both be inside the same [inbound] 0_o |
| 06:37.39 | tehrabbitt | lol |
| 06:38.55 | kaldemar | be clear on what you want to do first. |
| 06:38.56 | *** join/#asterisk ajkaanbal (~ajkaanbal@189.181.31.58) |
| 06:39.42 | tehrabbitt | I have 3 DIDs... I have two phones (devices). one phone (business) will have just one DID assigned to it, the personal phone will have two. |
| 06:40.16 | tehrabbitt | on outbound, I want the personal calls to be routed via the personal account / pesonal DIDs and vice versa with the business |
| 06:40.31 | tehrabbitt | problem is, my business uses IAX for Outbound, SIP for inbound |
| 06:40.41 | tehrabbitt | will that matter much? |
| 06:41.32 | kaldemar | how is that a problem? |
| 06:42.23 | tehrabbitt | I guess it's not, maybe I just confused myself... I guess what i'm trying to say is should I list both sets of incoming filters under the "incoming" context |
| 06:42.29 | tehrabbitt | well all 3 sets I should say |
| 06:42.51 | *** join/#asterisk cyford (~cyford@96-25-169-243.gar.clearwire-wmx.net) |
| 06:43.13 | tehrabbitt | I guess just make one context "Businessoutbound" and the other "personaloutbound" |
| 06:43.24 | tehrabbitt | with different "Dial(XXXX)" for each context? |
| 06:43.30 | kaldemar | tehrabbitt: what you should do depends on what you configure in sip.conf/iax.conf. |
| 06:44.02 | tehrabbitt | ah true. |
| 06:44.43 | kaldemar | for incoming that is. for outbound calls, make one context that has an extension for the personal DID and one for the business. then include the personal in the personal phone's context and same for the business one. |
| 06:45.27 | kaldemar | by include i mean include statements, as in "include => anothercontext" inside a [context]. |
| 06:46.23 | tehrabbitt | ah true |
| 06:48.24 | *** join/#asterisk imcdona (imcdona@2001:470:e8f1:4:243c:284b:85c0:282c) |
| 06:52.55 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 06:53.03 | kleszcz | morning |
| 06:53.39 | tehrabbitt | kaldemar, what's the best way of transfering a user to voicemail if there is no answer within lets say the default 35 secons I have set for "dial()" |
| 06:54.30 | kaldemar | tehrabbitt: set a timeout in the dial app and the voicemail app as the next priority. core show application dial. |
| 06:54.49 | tehrabbitt | ah |
| 06:56.21 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:57.11 | schmidts | good morning |
| 06:57.25 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 07:14.43 | tehrabbitt | How can I enable T.38 on one of my SIP lines? |
| 07:18.42 | *** join/#asterisk emora (~emora@213.236.9.114) |
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| 07:19.26 | *** part/#asterisk mpe (~mpe@office.ipvision.dk) |
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| 07:32.43 | cjk_ | hi, i am looking for a way to hangup a call from a Local channel. I can hangup the local channel but this does not affect the "parent" |
| 07:32.54 | cjk_ | any hints or ideas are welcome |
| 07:43.59 | *** join/#asterisk emora (~emora@213.236.9.114) |
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| 07:55.43 | puzzled | hi |
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| 08:10.57 | tehrabbitt | stupid question, I have two DIDs from flowroute, I have them configured in my asterisk server, and I have my asterisk server registering to flowroute, when I dial my DID's I get a message saying "you have reached a non-working number". I know i'm missing something in one of my .conf but i'm not sure what |
| 08:12.39 | tehrabbitt | :You can manage your inbound routes below and link any of your DIDs to any route. (Not required if using SIP registration for inbound. DIDs route via SIP Regis |
| 08:12.49 | tehrabbitt | so i'm not sure what i'm doing wrong |
| 08:13.14 | *** join/#asterisk jg1234 (~jan@dslc-082-082-037-188.pools.arcor-ip.net) |
| 08:13.15 | tehrabbitt | register => XXXXXXXX:XXXXXXXXXXXX@sip.flowroute.com ;TeneHawk Account |
| 08:13.34 | jg1234 | hi |
| 08:14.13 | tehrabbitt | nevermind figured it out >_< |
| 08:17.21 | jg1234 | i am trying to delete a line in a context of my extensions.conf via AMI -> UpdateConfig |
| 08:17.39 | jg1234 | but all it lets me do is delete the whole context |
| 08:20.03 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 08:26.41 | kaldemar | jg1234: a single line in an extension or a whole extension? |
| 08:29.41 | jg1234 | one extension would be fine |
| 08:29.59 | kaldemar | jg1234: use Command and dialplan remove exten@context [priority] |
| 08:30.26 | jg1234 | ok |
| 08:32.15 | jg1234 | and do you have any idea why updateconfig wont delete just one line ? |
| 08:33.37 | jg1234 | http://pastebin.com/BQzbXqur |
| 08:38.10 | *** join/#asterisk Ecco (~User@anj75-2-88-162-180-91.fbx.proxad.net) |
| 08:38.13 | Ecco | Hi everyone |
| 08:38.45 | Ecco | Quick question : I have a SIP account I'm paying for, and I'd like to share that account on my LAN. |
| 08:38.57 | Ecco | (IOW, I'd like anyone here to be able to pick up a phone call on their machine) |
| 08:39.20 | Ecco | (Ideally, it would be great if people could also forward a call to each other) |
| 08:39.27 | Ecco | what's the proper way to do this ? |
| 08:40.19 | Faustov | get an asterisk server to route your sip calls |
| 08:40.29 | Faustov | similarly to masquerade in networking |
| 08:41.41 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 08:41.54 | Ecco | Hmm, ok |
| 08:42.07 | Ecco | Is this something difficult to do ? |
| 08:42.23 | Ecco | (This would be my first try at setting up a PBX and I don't want to start with something too hard) |
| 08:43.08 | teathsch | nothing to it.. just focus on understanding sip.conf and extensions.conf |
| 08:43.14 | Ecco | Allright :-) |
| 08:43.16 | Ecco | Thanks |
| 08:44.24 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 08:50.44 | kaldemar | jg1234: it's not supposed to delete one line. |
| 08:51.00 | kaldemar | jg1234: it is for operating on category contexts. |
| 08:51.42 | kaldemar | jg1234: or settings. |
| 08:52.42 | jg1234 | ok thx |
| 08:58.33 | *** join/#asterisk BlackBishop (dexter@2001:470:26:45f::1) |
| 08:58.52 | BlackBishop | any way to check if someone has ringed more than like .. 3 seconds .. to go do a callback somehow ? |
| 08:59.06 | BlackBishop | I've read about the callback by creating a file in a queue somewhere .. |
| 08:59.17 | BlackBishop | but how do I check how long did a person ring ? |
| 08:59.33 | BlackBishop | and make the incoming wait to ring at least X amount of seconds |
| 09:05.43 | *** join/#asterisk davlefou (~david@41.225.9.81) |
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| 09:27.37 | BlackBishop | ok .. so I should do Wait(5) or something .. but how can I find out if the person hanged up in less than 5 seconds ? |
| 09:33.15 | *** join/#asterisk flashdeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 09:34.12 | flashdeluxe | hi! does anybody know if chan capi supports asterisk 1.8? |
| 09:39.15 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 09:39.29 | kaldemar | BlackBishop: take an epoch before dial and compare it to the value after the user has hung up. how to proceed dialplan execution after a hangup is up to your asterisk version. |
| 09:40.54 | BlackBishop | yeah ... good idea |
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| 10:12.04 | BlackBishop | kaldemar: how do I set in a var ${EPOCH}-${START_CALL} ? |
| 10:12.36 | BlackBishop | it currently seems to interpret exten => h,n,Verbose(RING_TIME = ${EPOCH}-${START_CALL}) as RING_TIME = int-int |
| 10:12.39 | BlackBishop | not as a result |
| 10:12.40 | BlackBishop | :/ |
| 10:13.26 | BlackBishop | which should be a number between 1 and 5 |
| 10:14.25 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 10:15.23 | BlackBishop | MATH |
| 10:16.32 | *** join/#asterisk vadmeste (~anis@41.224.36.130) |
| 10:23.03 | vadmeste | Hello everybody. Can someone tell me some keywords that guide me videoconference with asterisk ? |
| 10:34.46 | kaldemar | vadmeste: app_conference or wait until someone implements it in asterisk. |
| 10:36.24 | *** join/#asterisk micols (~ident@rlogin.dk) |
| 10:46.10 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 10:47.31 | z4nD4R | hi guys, i want to implement IM and video to my asterisk, have samo tutorial?... and wich client ( free ) supports this features? thx... |
| 10:50.22 | *** join/#asterisk dimm1 (~appleworm@unaffiliated/dimm) |
| 10:53.22 | nickfennell | What's the deal with asterisk and video |
| 10:53.25 | nickfennell | is it supported now? |
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| 10:57.05 | *** join/#asterisk X-Rob (~Rob@eth2083.qld.adsl.internode.on.net) |
| 10:57.06 | zamba | how do i set the default prompt language? |
| 10:57.35 | *** join/#asterisk alkali147 (~alkali147@109.205.41.244) |
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| 11:03.22 | aberrios | Anyone using the shiny new version of OrderlyStatsse? ( 1.8RC5) |
| 11:12.40 | alkali147 | actualy never tried it, seems to be very heavy |
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| 11:25.07 | *** join/#asterisk nickfennell (~nick@cov1.appliansys.com) |
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| 11:50.24 | atan | nickfennell, it has been for awhile |
| 11:50.36 | atan | nickfennell, I use it every day :) love it |
| 11:51.03 | atan | z4nD4R, I think x-lite has video support... |
| 11:51.03 | nickfennell | Really? |
| 11:51.14 | nickfennell | I'm interested in deploying it |
| 11:51.21 | atan | nickfennell, totally. I've had the family setup with video phones for the last few months. |
| 11:51.32 | nickfennell | Oh nice |
| 11:51.39 | nickfennell | Any pointers/caveats? |
| 11:51.49 | atan | I can't tell you much about how it all works... I'm not that technical. I just plugged it all in and the thing worked really. |
| 11:51.56 | atan | Watch out for the codecs is all I do suppose. |
| 11:52.04 | z4nD4R | atan: and i muss something set on asterisk side? to support video? |
| 11:52.37 | *** join/#asterisk hugorebelo (~hugo@201.81.187.10) |
| 11:52.42 | atan | I'm using h246 for the phones but other devices, like my cellphone, can't connect to the video phones (voice works) because of it. I guess my phone only supports h243 though. |
| 11:53.11 | atan | z4nD4R, maybe. http://www.voip-info.org/wiki/view/Asterisk+video |
| 11:54.14 | atan | nickfennell, the video works great for me. I have one phone in Taiwan and another in Canada. The two talk like they are right there. It's actually better than the POTS line they both have which has a 1 second delay! |
| 11:54.35 | atan | nickfennell, what are you planning anyway? :) |
| 11:55.12 | z4nD4R | atan: nice thx, and have you some info about IM messaging trought asterisk? |
| 11:56.05 | *** join/#asterisk kaushal (~kaushal@182.72.14.170) |
| 11:56.07 | kaushal | hi |
| 11:56.29 | kaushal | is there a way to know the PRI Link is OK or UP in Asterisk ? |
| 12:01.55 | nickfennell | atan, nothing too serious, just a quick and easy video ability |
| 12:03.06 | nickfennell | yeah there's a pri status command I think |
| 12:03.19 | nickfennell | show active or show channels |
| 12:04.17 | kaushal | nickfennell: any example ? |
| 12:04.50 | nickfennell | http://www.voip-info.org/wiki/view/Asterisk+CLI |
| 12:04.52 | nickfennell | There's a few |
| 12:06.04 | *** join/#asterisk Linux4Eric (~chatzilla@24.209.64.104) |
| 12:07.33 | kaldemar | kaushal: dahdi show status |
| 12:08.10 | nickfennell | help them to help themselves kaldemar |
| 12:10.16 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 12:10.16 | Linux4Eric | Anyone else having issues with binary packages via yum. I get kernel dependency problems with dahdi packages |
| 12:11.02 | kaushal | kaldemar: it says command not found ? |
| 12:11.57 | kaushal | kaldemar: I am using asteriknow |
| 12:12.04 | kaushal | 1.7.1 |
| 12:12.16 | grEvenX | anyone else using asterisk-java M3 with asterisk 1.4.x and successfully getting a result from SipShowPeerAction ? |
| 12:12.24 | dimm | zamba: what do you mean ? |
| 12:12.50 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:12.50 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:12.51 | zamba | dimm: i want to change the prompts to norwegian language |
| 12:13.08 | kaushal | kaldemar: got it |
| 12:13.28 | dimm | zamba: did you mean language in system prompt ? |
| 12:13.35 | zamba | yes |
| 12:13.38 | zamba | that's why i wrote that :) |
| 12:13.54 | zamba | with prompts i mean the audio spoken.. like "press 1 for " and so on |
| 12:13.57 | dimm | zamba: LANG=C man vim |
| 12:14.00 | zamba | nono |
| 12:14.11 | dimm | zamba: this command for run 'man vim' at english language |
| 12:14.11 | zamba | voice prompts |
| 12:14.23 | dimm | zamba: aa, i don't know, sorry |
| 12:14.31 | leifmadsen | zamba: after you've recorded the language prompts and put them in /var/lib/asterisk/sounds/<lang>/ (like en, fr, es, etc...) then do Set(CHANNEL(language)=en) |
| 12:14.41 | leifmadsen | zamba: core show function CHANNEL |
| 12:14.43 | zamba | leifmadsen: but i want to do this globally.. for all channels |
| 12:14.49 | leifmadsen | zamba: pretty sure it's language, not 100% sure |
| 12:14.54 | leifmadsen | try asterisk.conf |
| 12:15.02 | leifmadsen | or the channel configuration file |
| 12:15.15 | leifmadsen | you can probably set it per channel type in the [general] section |
| 12:15.41 | zamba | like in sip.conf? |
| 12:15.42 | leifmadsen | zamba: http://ofps.oreilly.com/titles/9780596517342/asterisk-Internationalization.html |
| 12:18.51 | Linux4Eric | Anyone else having problems with a CentOS 5.4 and binary install of dahdi in the last 2 weeks? |
| 12:21.22 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
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| 12:53.39 | anonymouz666 | russellb: are you there? |
| 12:54.08 | leifmadsen | probably not yet |
| 12:54.14 | leifmadsen | it's only shortly past 8am there |
| 12:54.28 | leifmadsen | sorry, shortly before 8am |
| 12:54.49 | anonymouz666 | I'd ask about the thread he posted in 2007.. [asterisk-dev] Application timeouts, Periodic and Scheduled Announcements |
| 12:54.51 | leifmadsen | I don't imagine you'll see him online for at least another hour |
| 12:56.23 | *** join/#asterisk logicwrath (~no@mail.vistitude.com) |
| 12:56.33 | anonymouz666 | I can't follow if this proposal was implemented or changed etc |
| 12:57.50 | logicwrath | are there any projects that will allow me to do nag calling for collections using asterisk? |
| 12:58.05 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 13:10.54 | *** join/#asterisk luke-jr (~luke-jr@ishibashi.dashjr.org) |
| 13:16.12 | psilikon | logicwrath, maybe vicidial |
| 13:16.35 | Linux4Eric | logicwrath, take a look at asterisk call-files |
| 13:17.13 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 13:17.15 | Linux4Eric | it would be more of a do-it-yourself. |
| 13:17.17 | jacc0 | hi all |
| 13:17.23 | jacc0 | I keep getting this error: |
| 13:17.29 | jacc0 | sorry this one : |
| 13:17.40 | jacc0 | [Apr 12 19:06:48] WARNING[4007]: channel.c:6493 ast_do_masquerade: Channel type 'NULL' does not have a fixup routine (for Bridge/SIP/172.20.143.211-0000001a<ZOMBIE>)! Bad things may happen. |
| 13:18.33 | jacc0 | when trying to Bridge(SIP/172.20.143.211-0000001a) from dialplan |
| 13:19.19 | logicwrath | yea, I thought about using call-files. I was hoping someone might have already done the work for a simple nag calling implementation |
| 13:19.37 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 13:19.51 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 13:19.55 | logicwrath | i dont think vicidial has automatic calling, i think its more of a dialer for call centers |
| 13:20.08 | Linux4Eric | what are you planning on using to get the list of deadbeat payers |
| 13:20.24 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 13:20.25 | logicwrath | i was going to manually enter the numbers |
| 13:20.30 | logicwrath | when people owe me money |
| 13:20.36 | logicwrath | take them out when they pay |
| 13:21.36 | Linux4Eric | so you were going to enter everyone as soon as they owe you money and then after so many days if you don't remove them they would get a call? |
| 13:21.48 | jacc0 | I guess nobody can help me |
| 13:22.25 | jacc0 | :S |
| 13:22.32 | logicwrath | I was thinking once they go past due for so long I would manually enter their phone number and maybe their name for TTS and then they would get 1 call a day until they pay automated |
| 13:22.33 | *** join/#asterisk chasing`Sol (~cS@smtp.master-zone.net) |
| 13:23.08 | logicwrath | i figure when i owe credit cards or the bank those turkeys call me 3 times a day until i take care of it |
| 13:25.05 | logicwrath | i am pretty sure i could do it with call files and scripts but it would be better if someone else did something similar already i could just use without programming my own |
| 13:25.43 | kaldemar | jacc0: it thinks your tech is NULL. was Bridge(SIP/172.20.143.211-0000001a) what you saw in CLI? |
| 13:29.07 | psilikon | logicwrath, I don't know if vicidial can do that but I am pretty sure it could but would most likely be overkill for your needs. |
| 13:29.31 | logicwrath | i thought vicidial was just a dialer with a script for call centers |
| 13:29.39 | *** join/#asterisk eerie (hoax@gateway/shell/bshellz.net/x-zlyqgoenzczvchsi) |
| 13:29.40 | logicwrath | and maybe a small CRM |
| 13:29.53 | *** part/#asterisk tasca (~tasca@189.34.27.64) |
| 13:31.25 | jacc0 | @kaldemar: yes, I was trying to bridge from dialplan using de bridge() app |
| 13:33.46 | *** join/#asterisk Faithful (~Faithful@202.189.73.144) |
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| 13:35.08 | jacc0 | @kaldemar: is there anything I can do to avoid this? |
| 13:35.36 | jacc0 | <PROTECTED> |
| 13:36.03 | jacc0 | @kaldemar: how could I check if a channel exists before bridging? (from dialplan) |
| 13:39.53 | jacc0 | <PROTECTED> |
| 13:40.44 | jacc0 | I have some very unstable asterisk installations: asterisk process stops |
| 13:40.55 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 13:40.56 | jacc0 | I'm guessing this could have something 2 do with it |
| 13:42.29 | *** join/#asterisk Cytech (~samuel@187.95.38.156) |
| 13:42.40 | Cytech | Hi guys |
| 13:43.51 | *** join/#asterisk MrTelephone (~MrTelepho@h697179-171.picriverisp.net) |
| 13:44.03 | Cytech | I have a problem with asterisk-gui 2.0 running under Debian 5.0 Lenny, always I put the url http://"myserveraddress":8088/asterisk/static/config/index.html |
| 13:44.24 | Cytech | I receive a message saying: Checking write permission for gui folder |
| 13:48.33 | MrTelephone | Does anyone have experience with both Sangoma and Digium t1 PRI cards? I've been running the sangoma for a long time but it's always a huge hassle to upgrade asterisk/wanpipe/kernel versions. Are the Digium cards more plug and play? Asterisk 2.4.40 with the latest DAHDI and wanpipe drivers work like $*$*#(@. When I tried the old stable wanpipe drivers they wouldn't compile with the kernel headers I had. Today I'm going to place a cisc |
| 13:49.03 | MrTelephone | I know these companies support their cards but for $1200/card the drivers should compile and work properly |
| 13:49.16 | jacc0 | I've some experiance with sangoma a200 and a500 |
| 13:49.35 | Cytech | HAAAAAAa anyone please help-me |
| 13:50.14 | MrTelephone | jacc0, do you run them now? What kernel version asterisk dahdi? |
| 13:50.27 | *** part/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 13:50.30 | jacc0 | I don't |
| 13:50.41 | MrTelephone | Maybe doing a firmware update is a bad idea. Are you using Digium now? |
| 13:50.42 | jacc0 | I'll show you emails from some days ago |
| 13:50.44 | WIMPy | MrTelephone: That's why I used to say that I prefer hardware that is supported by the standard Linux kernel. |
| 13:51.06 | *** join/#asterisk gego (~quassel@b238085.customer.hansenet.de) |
| 13:51.11 | MrTelephone | WIMPy, you are talking about digium? |
| 13:51.21 | WIMPy | But actually I'm using Digium hardware for PRIs. |
| 13:52.00 | MrTelephone | I always lived with the issue that a new call picks up a few milliseconds from another timeslot. |
| 13:52.04 | jacc0 | @MrTelephone: you are using debian 6 (squeeze)? |
| 13:52.09 | WIMPy | Neither the Sangoma nor the Digium cards are supported by standard Linux. |
| 13:53.06 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:53.06 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:53.07 | MrTelephone | whatever http://http.us.debian.org/debian stable main contrib non-free is linked to |
| 13:53.43 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 13:53.57 | MrTelephone | What would you guys try. Different computer systems? Maybe the mainboards I have don't like the kernel PCI device software |
| 13:54.07 | jacc0 | and asterisk 1.8? |
| 13:54.36 | MrTelephone | No I'm using 1.4 series software right now. |
| 13:54.37 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:54.49 | *** join/#asterisk wonderworld (~ww@port-92-201-228-210.dynamic.qsc.de) |
| 13:54.56 | jacc0 | woomera doesn't compile to the new kernel from debian |
| 13:55.01 | MrTelephone | Do you run your asterisk boxes on server mainboards or desktop? |
| 13:55.05 | jacc0 | they are working on an update |
| 13:55.05 | leifmadsen | yes |
| 13:55.09 | WIMPy | For ISDN interoperability, 1.8 is a pretty good idea. |
| 13:55.17 | jacc0 | but have some technical difeculties |
| 13:55.42 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 13:55.59 | jacc0 | I've tryed server and desktop |
| 13:56.18 | MrTelephone | ISDN including PRI? |
| 13:56.19 | leifmadsen | I've installed a few systems using commodity hardware and it works fine |
| 13:56.24 | leifmadsen | PRI is ISDN |
| 13:56.46 | WIMPy | jup |
| 13:56.54 | leifmadsen | ISDN is what PRI signalling travels over |
| 13:57.05 | jacc0 | yes |
| 13:57.07 | serafie | in users.conf, is linenumber the number of lines on the device or the line to ring? |
| 13:57.10 | MrTelephone | ok |
| 13:57.10 | jacc0 | sangoma A500 |
| 13:57.39 | jacc0 | <PROTECTED> |
| 13:57.56 | jacc0 | and complain about wanpipe not compiling on debian 6 |
| 13:58.32 | MrTelephone | well I was receiving a lot of overruns on one t1 port and the calls wouldn't go out. On the other port I have a channel bank and could make calls to that. Hooked to a telco I should have the wanpipe set to NORMAL 0 i guess. It's always been set to that |
| 13:59.04 | MrTelephone | I'm gonna join them and tell them my hardware is crap and their hardware is caca and together it's one big pile of manure |
| 13:59.07 | WIMPy | leifmadsen: I'd put it the other way round. |
| 13:59.13 | MrTelephone | lol |
| 13:59.41 | MrTelephone | leifmadsen, why do you suggest 1.8 for isdn. Long story? |
| 13:59.50 | WIMPy | Overruns as in receive overruns? |
| 14:00.04 | MrTelephone | both ways |
| 14:00.10 | MrTelephone | but then my channel bank port had no errors |
| 14:00.25 | *** join/#asterisk Gugge (~gugge@vlan2.dlxhosting.dk) |
| 14:00.45 | WIMPy | I did. Because Asterisk 1.8 has some changes that improve interoperability, as well as several new features. |
| 14:00.58 | jacc0 | !!!!! alert\ |
| 14:01.05 | leifmadsen | MrTelephone: I didn't suggest it -- WIMPy did |
| 14:01.05 | MrTelephone | I'm scared to use it for production |
| 14:01.07 | jacc0 | mail about sangoma with asterisk 1.8 |
| 14:01.21 | jacc0 | This is an update on our Asterisk 1.8 policy for Sangoma hardware/software using Chan_woomera. |
| 14:01.21 | jacc0 | Sangoma had previously decided to move our Chan_woomera related software to maintenance only mode but with the large number of changes in Asterisk 1.8 we have decided to completely end of life that project. |
| 14:01.21 | jacc0 | All our hardware (PRI, BRI, and analog) is now fully supported in SMGV3. For our BRI line of cards, SMGv3 will also be our only conduit for Asterisk. |
| 14:01.21 | leifmadsen | I've installed 1.8 in several locations without issue |
| 14:01.31 | WIMPy | Receive overruns suggest a hardware near issue like interrupt processing. |
| 14:01.33 | MrTelephone | I just recently switched to DAHDI. that is how far I am behind |
| 14:02.16 | MrTelephone | To be honest I didn't even try to reboot yet after the firmware upgrade |
| 14:02.37 | MrTelephone | People goto IT school for 4 years to get a degree in rebooting |
| 14:03.20 | MrTelephone | I'm about 95% convinced to buy an atlas 550 so I can share timeslots on a single T1 so I can do more testing |
| 14:03.35 | MrTelephone | right now I have to come into work after hours to do any maintenance on the PRI |
| 14:04.41 | WIMPy | An IAD is usually used for the opposite scenario. |
| 14:05.04 | MrTelephone | IAD? |
| 14:05.46 | WIMPy | The Atlas 550 is called an IAD in the product page. |
| 14:06.06 | TobSnyder | tzafrir_laptop? |
| 14:06.35 | MrTelephone | Integrated Access Device? |
| 14:06.48 | WIMPy | yes |
| 14:07.07 | MrTelephone | PRI's are so expensive here we can't afford to have 2 and it makes it very hard to bring it down and test |
| 14:08.22 | WIMPy | So you want the IAD for testing, not as a replacement? |
| 14:10.21 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 14:10.32 | Aut0ExeC | hi everyone |
| 14:20.41 | russellb | anonymouz666: it did not get implemented |
| 14:21.27 | anonymouz666 | russellb: why? it looks so promising |
| 14:21.38 | russellb | i don't remember :-) |
| 14:21.45 | russellb | i just know i didn't end up doing it |
| 14:22.28 | anonymouz666 | russellb: I found that while searching for the feature, I doubt it if it can be done today... scheduled announcents into active call and yet different messages from callee and caller. |
| 14:22.46 | anonymouz666 | "to callee" |
| 14:23.59 | russellb | well, it's possible by being clever |
| 14:24.11 | russellb | the Asterisk Cookbook discusses how to use ChanSpy and Originate to play sounds into a call |
| 14:24.15 | russellb | that's how you would do it |
| 14:24.24 | russellb | you could use AMI originate for example ... |
| 14:24.44 | russellb | Playback() on one side, ChanSpy() on the other in whisper mode to play only to one side of the call |
| 14:26.39 | anonymouz666 | heh, oh well, right. With Asterisk you need to use also your creativity :-) |
| 14:26.52 | MrTelephone | WIMPy, no I want it for production so I can have a failover asterisk server without switching physical t1 lines |
| 14:27.14 | russellb | anonymouz666: yep ... it's creative, but at least it's possible |
| 14:27.36 | WIMPy | MrTelephone: Are you sure it can be used that way? |
| 14:30.05 | MrTelephone | Yeah I talked to their technical support team about it. What happens is that you can actually do dialplans in the atlas 550. So if you have 2 channel banks hooked up and the plan says to go out to the TELCO PRI then it will use the next available timeslot. But in my scenerio I was going to route everything to asterisk for processing before bridging the call back out the TELCO. I can't think of any other solutions :( |
| 14:30.52 | *** join/#asterisk zkn (~zkn@195.222.14.202) |
| 14:31.55 | WIMPy | Maybe you should try your card in another slot? |
| 14:32.05 | MrTelephone | I like the AS530 as an edge router for bridging T1 and SIP but then I still need to have something for the channel bank. I thought of extending some t1's to remote offices over wireless and all I would have to do is plug it into the atlas 550 instead of upgrading both asterisk servers to 4 port T1's |
| 14:32.31 | MrTelephone | Yeah or buy a better server maybe :( |
| 14:32.49 | WIMPy | I didn't say it has to be in the same board ;-) |
| 14:33.19 | WIMPy | Uh, T1 over wireless could become ... interesting. |
| 14:33.29 | MrTelephone | I'm using Asus NLCV-D2 boards in both servers thinking that the PCI bus wouldn't flinch at the thought of a t1 card :) |
| 14:33.46 | *** join/#asterisk antixsuperstar (~antixsupe@201.155.127.36) |
| 14:33.49 | MrTelephone | I guess t1 over wireless works somewhat like TDMoIP |
| 14:34.19 | WIMPy | Yes, but wireless usually means big jitter. |
| 14:35.00 | MrTelephone | Yeah you have to use commercial grade stuff like 24GHZ point to point or something. |
| 14:35.00 | antixsuperstar | hi guys! i don't know where else could I ask. where can i get some info on propietary software (pbxunified; panasonic)? |
| 14:35.20 | \DSAFEW\ | pabelanger, tvc123 you guys helped me out, and I've finally got it working, thanks for the resources I'll be perusing them more when I have the time |
| 14:35.24 | leifmadsen | antixsuperstar: google or the manufacturer |
| 14:35.37 | \DSAFEW\ | pabelanger, tvc123 so thanks |
| 14:35.37 | WIMPy | MrTelephone: Ok, that's another story then. |
| 14:36.05 | antixsuperstar | leifmadsen: i've tried that. just crappy forged search result pages. nothing useful so far. no torrents, no manuals, no tutorials... |
| 14:36.40 | n3hxs | Google gave me this: http://www.google.com/search?client=ubuntu&channel=fs&q=+%28pbxunified%3B+panasonic%29%3F&ie=utf-8&oe=utf-8 |
| 14:36.54 | MrTelephone | I considered the idea because there was a location here that is reachable by wireless. Telco fiber doesn't go that far but we could sell them a t1 and voice access/internet access. But if I can't get things running smoothly here than what is the point. |
| 14:37.50 | WIMPy | MrTelephone: If you want to sell a PRI I'd try to avoid SIP in the path at all cost. |
| 14:38.15 | WIMPy | There are more low level options that work much smoother. |
| 14:38.37 | MrTelephone | I agree. I was looking at proprietory T1 over wireless. Expensive. But RAD makes some equipment that apparently is really good for doing t1 over IP. |
| 14:38.55 | antixsuperstar | n3hxs: any page in particular? all of those results are 'forged'. none of them have anything useful. |
| 14:39.26 | MrTelephone | I'm afraid of using sip from asterisk to a t1 gateway device as it may break the FAX? |
| 14:39.36 | WIMPy | RAD are well known to provide sulutions like that. But there's also ISDNoIP or L1oIP as done by Linux. |
| 14:40.05 | MrTelephone | before i said TDMoIP but i meant to say TDMoe |
| 14:40.10 | MrTelephone | big difference |
| 14:40.15 | WIMPy | It's likely to break a lot more than that. But that depends on the features you get/expect be be available at the other end. |
| 14:40.55 | WIMPy | Yes, the RAD stuff I have seen does E1 over ethernet (no IP). |
| 14:41.08 | MrTelephone | say you used it just for routing calls to the PSTN. I'm sure the quality will be good. The t1 edge device will be on the same network switch so there shouldn't be jitter/loss |
| 14:41.36 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 14:41.58 | WIMPy | If "routing calls" == "basic call setup", yes. But usually you expect more than that. |
| 14:44.03 | MrTelephone | So I setup this as5350 to try it out to replace the PCI t1 cards in the asterisk machines. I goto setup the port for the channel bank and the IOS doesn't support FXO signalling, only FXS. All the documentation online shows that FXO should be an option. Upgraded the IOS and still nothing. You ever ask yourself why you work in the IT field? |
| 14:44.18 | MrTelephone | I'm going back to school to become a lawn mower rider |
| 14:45.04 | zkn | hi, can someone explain the logic behind "setqueuevar" parameter.. it is said that if it's set to yes, variables will be set just prior to the caller being bridged with a queue member and just prior to the caller leaving the queue... |
| 14:45.45 | zkn | but when can I use these variables in the dialplan ? |
| 14:46.01 | WIMPy | tries to avoid Cisco for anything more than switching VLANs :-) |
| 14:47.06 | zkn | if I write: Verbose(0, ${QUEUENAME}) before the queue app then i'm not gettin anything in the console |
| 14:48.36 | MrTelephone | A port died in a fairly new cisco switch three days ago. That is unlike Cisco |
| 14:49.08 | MrTelephone | Swear words fly around here like birds |
| 14:49.25 | serafie | zkn: try another value 0-10. |
| 14:49.58 | serafie | Verbose(10, ${QUEUENAME}) |
| 14:50.08 | WIMPy | No failing hardware is not something they're known for. Unless it's a cutting edge interface. |
| 14:51.35 | Aut0ExeC | where can I find the list of Global variables? |
| 14:51.43 | Aut0ExeC | like QUEUENAME |
| 14:51.47 | Aut0ExeC | for example |
| 14:52.04 | MrTelephone | It's beyond the scope of my knowledge to exercise the idea of a port blowing. Can induction from nearby electrical cause a surge big enough to fail a port? |
| 14:52.27 | drfreeze | Looks like 1.8.4-rc2 defaults to g0 when dialing out |
| 14:52.38 | MrTelephone | Maybe the ground of the remote switch had a different potential and it slowly cooked it like eggs in a frying pan |
| 14:52.45 | zkn | Aut0Exec voip-info has a list of variables |
| 14:52.50 | Aut0ExeC | zkn: thanks |
| 14:54.41 | zkn | serafile: verbosity level did not make any difference: ${QUEUENAME} just doesnt display anything because it has no value. |
| 14:55.00 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 14:55.33 | serafie | Heh, that'll do it. |
| 14:56.49 | *** join/#asterisk mpe_ (~mpe@94.127.50.104) |
| 14:58.52 | zkn | so at what stage in the diaplan will I be able to extract values from the varaibles set by setqueuevar=yes in queues.conf ?? |
| 14:59.11 | zkn | s/varaibles/variables |
| 15:00.02 | *** part/#asterisk mpe_ (~mpe@94.127.50.104) |
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| 15:19.22 | zkn | why does a channel variable lose it's value when the call moves from one context to another ? |
| 15:19.53 | leifmadsen | zkn: one context to another? it doesn't |
| 15:20.15 | leifmadsen | only if you create new channels will the channel variable not follow it unless you tell it to |
| 15:20.25 | leifmadsen | (like via Dial() and whatnot) |
| 15:20.38 | leifmadsen | if channels variables went away between contexts, nothing in asterisk would work |
| 15:20.40 | zkn | it does.. or maybe if not from one to another but when the 3rd is involved then the value set for a variable in the first context is lost |
| 15:21.05 | leifmadsen | no way it does that |
| 15:21.09 | *** join/#asterisk frawd (~francois@133.Red-83-41-197.dynamicIP.rima-tde.net) |
| 15:21.09 | leifmadsen | you have to be doing something to cause that |
| 15:21.16 | zkn | hmm |
| 15:21.20 | leifmadsen | or something is happening that causes other channels to become involved |
| 15:21.49 | leifmadsen | like if you're doing a Goto() between multiple contexts, the channel variables absolutely follow the channel between contexts |
| 15:23.20 | leifmadsen | http://pastebin.com/vV1v1xU4 |
| 15:23.22 | leifmadsen | like that |
| 15:23.29 | leifmadsen | MyVariable will always be available |
| 15:23.45 | zkn | ok i need to investigate my dialplan more thoroughly then, but i'm sure i'm not doing any Dialing or similar at any point, just processing the inbound call in multible contexts before routing it to Queue |
| 15:24.38 | leifmadsen | DING DING DING |
| 15:24.46 | leifmadsen | if you route to a queue, you're going to create new contexts and channels |
| 15:26.17 | zkn | queue is the endpoint, before the caller is sent to queue the variable set in context one should not be lost in context two or three when queue app resides in context four |
| 15:26.23 | zkn | or am i wrong ? |
| 15:28.24 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 15:28.46 | leifmadsen | zkn: it will exist up to the point you call Queue(), but not on the other side of Queue() |
| 15:29.00 | leifmadsen | doing a Goto() into a context which calls Queue(), the variable will absolutely exist |
| 15:29.07 | leifmadsen | you should just show us your dialplan in a pastebin |
| 15:29.31 | leifmadsen | at this point all I can do is speculate and tell you how it works, but you're obviously experiencing something else, which means you're doing something that is causing that |
| 15:29.41 | leifmadsen | it'd be easier to debug and tell you why it might do that with some reference |
| 15:30.02 | leifmadsen | if what you're saying were true, that would be a SERIOUS regression |
| 15:30.05 | *** join/#asterisk tzanger (~tzanger@gromit.mixdown.ca) |
| 15:30.17 | leifmadsen | and then we pink bellied tzanger's mom and we all laughed |
| 15:30.19 | leifmadsen | ohai! |
| 15:31.37 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 15:31.38 | Aut0ExeC | leifmadsen: whassup man? |
| 15:33.16 | zkn | leifmadsen: preparing pastebin |
| 15:34.55 | drfreeze | Anyone know how to get around this problem: Using SIP RTP CoS mark 5 |
| 15:35.06 | drfreeze | was able to make outbound calls, but now can't |
| 15:37.18 | zkn | ok.. so now I spot that typo :D geesh |
| 15:39.03 | zkn | but how to use variables generated by "setqueuevar" ? |
| 15:39.54 | zkn | how to have access to them would be a more precise question |
| 15:40.26 | leifmadsen | drfreeze: that's not a problem - it's just telling you the Class of Service bit was set to 5 |
| 15:40.40 | leifmadsen | if that really was a problem, it's because your router is rejecting anything marked with that bit |
| 15:40.45 | zkn | or even when in the dialplan is it possible to access them would be an even more precise question |
| 15:41.58 | leifmadsen | zkn: you have to use the variables set in the Local channel that the Queue will call |
| 15:42.20 | leifmadsen | if the queue is setting a variable, the only place to access it is in the channels that the queue creates |
| 15:43.27 | leifmadsen | per the docs: |
| 15:43.28 | leifmadsen | ; If set to yes, the following variables will be set |
| 15:43.28 | leifmadsen | ; just prior to the caller being bridged with a queue member |
| 15:43.28 | leifmadsen | ; and just prior to the caller leaving the queue |
| 15:43.48 | leifmadsen | the only place to access those variables is AFTER you've entered the queue |
| 15:44.23 | *** join/#asterisk coppice (~chatzilla@62.166.232.220.dyn.pacific.net.hk) |
| 15:46.49 | zkn | ok, let's try |
| 15:49.20 | *** join/#asterisk Faithful (~Faithful@202.189.73.144) |
| 15:50.37 | zkn | yep, works |
| 15:50.40 | zkn | cheers |
| 15:56.04 | Aut0ExeC | is it ok for telephones to span wirelessly? say thru wireless bridges? etc? |
| 15:56.41 | Qwell | if you're willing to accept the problems, sure |
| 15:56.54 | WIMPy | Aut0ExeC: It's possible but you may not want that. |
| 15:56.57 | leifmadsen | I just tried it over a microwave connection -- it was awful |
| 15:57.09 | Qwell | leifmadsen: Did you leave the doors open? |
| 15:57.12 | leifmadsen | asterisk won't care, and it'll work just fine, but the connection itself will introduce issues |
| 15:57.23 | Aut0ExeC | leifmadsen: thanks |
| 15:57.26 | leifmadsen | Qwell: only when I want to glow |
| 15:57.51 | Qwell | leifmadsen: You're positively glowing. |
| 15:58.06 | Aut0ExeC | leifmadsen: if its a soilid bridged connection then how can there be issues? |
| 15:58.11 | WIMPy | Got gowstick? |
| 15:58.21 | leifmadsen | Aut0ExeC: I don't understand your question |
| 15:58.49 | WIMPy | s/go/glo/ |
| 15:58.51 | *** join/#asterisk c_rat (~mratliff@208.94.89.2) |
| 15:59.16 | Aut0ExeC | leifmadsen: would you suggest connecting phones over wireless bridge is my question |
| 15:59.19 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:59.34 | leifmadsen | Aut0ExeC: never would i recommend that |
| 15:59.44 | leifmadsen | too many possibilities for issues |
| 15:59.44 | Aut0ExeC | k |
| 15:59.53 | Aut0ExeC | like? |
| 16:00.03 | leifmadsen | like any of the possible issues you get with any wireless network |
| 16:00.36 | Aut0ExeC | i guess ur refering to security |
| 16:00.47 | Aut0ExeC | cuz as far as stability... that part is solid |
| 16:01.01 | WIMPy | More like packet loss and LOTS of jitter. |
| 16:01.42 | leifmadsen | WIMPy: +1 |
| 16:01.51 | *** join/#asterisk tzanger (~tzanger@gromit.mixdown.ca) |
| 16:01.54 | Aut0ExeC | kk thanks |
| 16:02.04 | leifmadsen | Aut0ExeC: I'm not referring to security, I'm referring to the inability for a wireless connection to be as solid as a wired connection |
| 16:02.13 | Aut0ExeC | i gatcha |
| 16:02.17 | leifmadsen | when it comes to VoIP your users will know when it's not working |
| 16:02.26 | leifmadsen | and there will be nothing you can do to fix it |
| 16:02.34 | leifmadsen | interference is far too easy to come about |
| 16:02.40 | Aut0ExeC | k |
| 16:02.54 | leifmadsen | goes to lunch |
| 16:03.47 | c_rat | hello mates |
| 16:06.20 | Aut0ExeC | c_rat: hi |
| 16:11.09 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 16:13.31 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 16:14.39 | *** join/#asterisk sled-dog (~luser@adsl-074-165-241-009.sip.msy.bellsouth.net) |
| 16:17.51 | *** join/#asterisk Er00 (er00@i.love.it.when.there.are.girlsinmylan.eu) |
| 16:21.37 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:26.30 | *** join/#asterisk stix (~sorena@200.111.132.195) |
| 16:28.11 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 16:29.43 | dandre | hello |
| 16:30.45 | dandre | I have an fxo port connected to a subscriber line. How can I from a sip phone connected to my asterisk box send a flash hook to the line |
| 16:31.05 | dandre | I have tried *0 but that doesn't work |
| 16:32.01 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 16:32.36 | dandre | I can see in the log: |
| 16:32.38 | dandre | [Apr 15 17:52:51] DEBUG[4633] chan_zap.c: Started VLDTMF digit '*' |
| 16:32.40 | dandre | ... |
| 16:33.42 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 16:38.56 | stix | Hi guys. I'm gonna ask a question about asterisknow, as #asterisknow is almost empty, and many of you probably is using the software. Can anyone tell me how to downgrade asterisk16 from latest (1.6.2.17.2) to 1.6.2.14? When I try to downgrade with yum, I end up with 1.6.2.9. |
| 16:39.25 | Qwell | yum can't downgrade |
| 16:43.38 | stix | Qwell, yes it can |
| 16:44.25 | *** join/#asterisk mawhii (~mawhii@170.220.119.70.cfl.res.rr.com) |
| 16:51.26 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v009-030.mobile.uci.edu) |
| 16:53.22 | WIMPy | just realizes that hist Asterisk doesn't seem to do anyhting sensibnle any more since I updated tonight. |
| 16:54.35 | *** join/#asterisk cj (~cjac@adsl-207-32-169-17.rockisland.net) |
| 16:54.38 | cj | hey folks |
| 16:55.18 | cj | could someone point me to some docs on trunking a couple of asterisk hosts to one another? I want calls initiated by phones registered on the first to reach phones registered on the second |
| 16:56.02 | leifmadsen | ~thebook |
| 16:56.02 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ |
| 16:56.05 | leifmadsen | cj: ^^^ |
| 16:56.15 | leifmadsen | lots of info about that in there |
| 16:56.19 | cj | thanks :) |
| 16:56.23 | leifmadsen | check out the DUNDi chapter |
| 16:56.30 | *** join/#asterisk ttpears (~ttpearso@gw.teamgleim.com) |
| 16:56.31 | cj | cool. will do. |
| 16:56.49 | cj | ooh! New release this year |
| 16:57.22 | cj | is the latest version available in .pdf or should I see if B&N has a copy? |
| 16:59.31 | *** join/#asterisk joesuffceren (~chatzilla@ip68-104-167-226.ph.ph.cox.net) |
| 16:59.51 | joesuffceren | Anyone aware of a good quality conference speaker/mic that I could plug in to an existing SIP enpoint with an RJ11 jack to "make" a decent conference phone. The built-in speaker on my endpoints (cisco 7940)is great for a small office, but not a conference room. I have some polycom Soundstation IP 6000s, which are fantastic, but I need a cheaper solution for this one |
| 17:00.04 | joesuffceren | I've done some looking but haven't come up with anything. Either A. I'm dreaming, and these things don't exist, or B. I don't know the correct name and so my search terms are incorrect. Any advice would be much appreciated. |
| 17:02.51 | _Corey_ | joesuffcern: Polycom has a tiny model called the IP5000 now |
| 17:02.56 | _Corey_ | I think it's cheap |
| 17:03.02 | _Corey_ | (not sure the price though) |
| 17:03.10 | JonathanRose | ~400 on Amazon I think. |
| 17:04.06 | stix | Hi guys. Can someone tell me why my asterisk sometimes completely hangs - not even local SIP is working? I have to restart the process. I see this a lot in the log when it happens: channel.c: Exceptionally long voice queue length queuing to Local/793@from-queue-a0b1;1 |
| 17:06.31 | *** part/#asterisk TobSnyder (~schneider@dslb-088-073-180-175.pools.arcor-ip.net) |
| 17:12.13 | *** join/#asterisk zeropoint46 (~zeropoint@c-24-6-81-186.hsd1.ca.comcast.net) |
| 17:13.40 | zeropoint46 | I was wondering if someone could help me with a probably simple problem with using the asterisk (* star) key incall while dialing into an external conference bridge? seems like asterisk is trapping the key and I can't seem to disable it. Thank you! |
| 17:15.13 | _Corey_ | zeropoint46: Check your features.conf and Dial() arguments... you will note several ways there that * could get trapped |
| 17:18.38 | zeropoint46 | don't see a "Dial()" arguemnt |
| 17:18.41 | zeropoint46 | argument |
| 17:18.49 | zeropoint46 | and I have disabled all features that start with a * |
| 17:19.24 | _Corey_ | zeropoint46: "core show application dial" read about options w,W,x,X etc. |
| 17:21.35 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 17:22.02 | zeropoint46 | so I have the feature code to access voicemail which begins with a star, but shouldn't that no be interpreted after a call has been established? |
| 17:22.50 | serafie | stix: you are getting that error with Asterisk 1.6.2.9? |
| 17:23.11 | stix | serafie, yes and also 1.6.2.17.2 |
| 17:23.17 | stix | tried a bunch of things now |
| 17:23.35 | serafie | I remember a bug related to that, but it was fixed very early in 1.6.2. |
| 17:23.48 | stix | okay, then it's not that then |
| 17:23.54 | serafie | yeah. oh well. |
| 17:26.53 | zeropoint46 | Corey, I do no have w,W,x,X enabled. |
| 17:28.45 | _Corey_ | How about T or t ? |
| 17:29.26 | _Corey_ | (You also didn't say what Asterisk was doing with the call when someone presses *... That would help narrow down what it could be) |
| 17:31.49 | joesuffceren | _Corey_: thanks for the recommendation, but ~400 (if that price is accurate) is a little more than I want to spend on this. I'll look into that model, though, to see if the pricing is better. |
| 17:32.50 | _Corey_ | Yeah, I know of a couple "speakerphone" USB boxes that may work with a softphone but nothing that would plug into a hardphone like you're asking for... |
| 17:32.54 | _Corey_ | good luck |
| 17:33.19 | *** join/#asterisk FeyFre (~panych_y@cpe-109-108-233-12.enet.vn.ua) |
| 17:33.30 | joesuffceren | _Corey_: I thought for sure I had seen something like that before. I must be losing it. I'll report back if I find something. thanks! |
| 17:34.48 | Qwell | joesuffceren: You've clearly insaned. |
| 17:36.11 | JonathanRose | At least it's past tense. |
| 17:36.20 | JonathanRose | There's hope. |
| 17:40.24 | \DSAFEW\ | I have Asterisk Click-to-Dial extension for firefox, it is connected as a manager, and has permission for calls, but I don't see any activity when I try to use it |
| 17:40.47 | \DSAFEW\ | I'm open to suggestions for better click-to-dial interfaces |
| 17:44.16 | zeropoint46 | Corey, I do use T and t but those just reference the use of the # key. * isn't doing anything. |
| 17:47.04 | \DSAFEW\ | I've run asterisk -dvvvgcr |
| 17:47.17 | Qwell | c and r? O.o |
| 17:47.25 | \DSAFEW\ | is that the most detail? I have no idea |
| 17:47.31 | fauxalliance | rdrr |
| 17:47.42 | \DSAFEW\ | fauxalliance, I read that out loud. |
| 17:47.53 | fauxalliance | chuckles again |
| 17:48.10 | _Corey_ | zeropoint46: Again, it'd be easier to offer suggestions if you could provide more detail on Asterisk is doing... (i.e. watch the CLI when someone hits * and pastebin that or something) |
| 17:48.22 | fauxalliance | core set verbosity 10 is the usual deal |
| 17:48.26 | fauxalliance | or -rvvvvvvvvvv |
| 17:48.29 | JonathanRose | c + r is a bit redundant |
| 17:48.41 | \DSAFEW\ | okay, thanks |
| 17:48.52 | fauxalliance | ~collectdebug |
| 17:48.52 | infobot | collectdebug is, like, a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 17:49.01 | JonathanRose | And you should only use r if asterisk is already running. |
| 17:49.11 | JonathanRose | c otherwise if you want a CLI |
| 17:49.12 | fauxalliance | hence, r is for Reconnect |
| 17:49.25 | fauxalliance | s/verbosity/verbose |
| 17:50.34 | zeropoint46 | I'm watching asterisk with "asterisk -rvvvvvvvvvv" when the * key is pressed in call it doesn't show anything. Another thing is I dont really see any dtmf tones in the CLI even when they work. |
| 17:50.34 | *** part/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 17:51.02 | \DSAFEW\ | I'm still not getting anything from the manager when I connect, the client's javascript message is "trying to call:18005551234" |
| 17:51.15 | fauxalliance | zeropoint46, that would probably be in the SIP debugging output... |
| 17:51.41 | zeropoint46 | when there and set "dtmf => debug" |
| 17:51.43 | fauxalliance | zeropoint46, i presume on a SIP trunk, RFC2833 non? |
| 17:51.46 | zeropoint46 | is that what it needs to be |
| 17:51.53 | zeropoint46 | RFC2833 |
| 17:52.02 | zeropoint46 | phones are set to RFC2833 as well |
| 17:52.19 | fauxalliance | then sir, you need to talk to the SIP side of things |
| 17:52.34 | zeropoint46 | thats been in the back of my head |
| 17:52.37 | \DSAFEW\ | when someone's connected to the server via AMI, is there any sort of account configuration which would matter? |
| 17:52.40 | zeropoint46 | I just don't want to go down that route |
| 17:53.00 | zeropoint46 | anybody used "callwithus" and had dtmf issues? |
| 17:53.40 | fauxalliance | zeropoint46, some RETARDED ITSP's insist on inband signaling..... killing kittens makes as much sence |
| 17:53.45 | fauxalliance | s/sence/sense |
| 17:54.02 | fauxalliance | BEEP BOOP BARF |
| 17:55.39 | zeropoint46 | hmm, when I press star in call now, audio drops |
| 17:55.45 | fauxalliance | inband dtmf can take a normal IVR, place it into a pillow case and beat it with a 4"x4" |
| 17:55.51 | fauxalliance | YMMV |
| 17:58.20 | zeropoint46 | faux, in my logger.conf I have dtmf => dtmf,debug,notice,warning,error,verbose... I still can't see any dtmf key pressed on the command line |
| 17:58.23 | zeropoint46 | is this expected |
| 18:00.25 | zeropoint46 | nm got it |
| 18:00.26 | fauxalliance | zeropoint46, depends on what type of signaling you use |
| 18:00.43 | fauxalliance | as aforementioned... start a sip debug... then watch your keypresses |
| 18:01.49 | zeropoint46 | okay, got it |
| 18:01.53 | zeropoint46 | here is pastebin |
| 18:02.09 | zeropoint46 | pastebin.com/7rADYGSm |
| 18:02.19 | fauxalliance | http://pastebin.com/7rADYGSm |
| 18:02.29 | zeropoint46 | seems like it's getting the dtmf and passing it |
| 18:02.47 | fauxalliance | good |
| 18:03.00 | _Corey_ | Where do you lose the call? |
| 18:03.06 | zeropoint46 | I don't |
| 18:03.13 | zeropoint46 | other end doesn't get key presses |
| 18:03.24 | _Corey_ | I thought you were saying you were losing the audio... |
| 18:03.30 | *** join/#asterisk Joe_CoT (~joecot@pdpc/supporter/active/joe-cot) |
| 18:03.33 | zeropoint46 | I think that was a fluke |
| 18:03.36 | zeropoint46 | not happening now |
| 18:03.56 | Joe_CoT | any reason why meetme announce wouldn't work? As in, I pass I into meetme's options, and I just join the room with no attempt at getting my name |
| 18:04.44 | _Corey_ | ah, well if you're just not getting DTMF across there are plenty of articles on that. It would seem Asterisk is getting your tones OK from the endpoint, so check your trunk |
| 18:07.58 | zeropoint46 | hmm, it's weird cause it was working before and now it's not, so thats why I thought it was just with * keys, seems it's all keys now |
| 18:08.19 | zeropoint46 | asterisk is transcoding to ulaw, could that have anything to do with it? |
| 18:09.59 | \DSAFEW\ | I changed my manager permissions and now have more errors (which is good) |
| 18:10.47 | \DSAFEW\ | I'm guessing my originate request is being sent, but it's not connecting to the extension now for some reason |
| 18:11.04 | \DSAFEW\ | I can't do the command from the console either, same error |
| 18:12.49 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 18:14.54 | *** join/#asterisk jong2 (~chatzilla@63.224.204.153) |
| 18:15.16 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 18:15.20 | _Corey_ | zeropoint46: Not typically a transcoding issue because you're seeing those DTMF events in your log |
| 18:15.37 | _Corey_ | I suspect you're not in sync with your carrier (i.e. rfc2833 vs. inband) |
| 18:20.26 | zeropoint46 | Corey, here is the issue, my carrier uses codecs that support out of band on some of their trunks and inband on other trunks, so if I set it one way DTMF fails on half the other calls |
| 18:20.33 | zeropoint46 | how do you recommend I resolved that |
| 18:20.52 | zeropoint46 | sometimes a call goes out ulaw, and my rfc2833 fails |
| 18:21.04 | zeropoint46 | and when I set it to inband, it breaks a bunch of other crap |
| 18:22.25 | cj | is there a preference here toward Asterisk-GUI or FreePBX? |
| 18:23.10 | \DSAFEW\ | in my attempts to get originate to call some where using the dialplan or just the Dial application, I keep getting the notice "__ast_request_and_dial: Unable to request channel SIP/1001" |
| 18:24.09 | fauxalliance | cj, hmmm... |
| 18:24.44 | _Corey_ | zeropoint46: I recommend inband with your trunk config then |
| 18:25.00 | fauxalliance | cj, want a FULL FEATURED PBX? or just a configuration overlay? |
| 18:25.29 | serafie | and cj, will you ever want to manually edit or tweak your config files? |
| 18:25.37 | cj | probably the former. need to test how our hardware load balances SIP, and we'll likely need to push it pretty hard. |
| 18:25.49 | cj | yes, will want to manually edit /etc/asterisk/* |
| 18:26.15 | serafie | if I am remembering correctly, FreePBX overwrites all manual edits |
| 18:26.44 | cj | maybe I should just install it from source on a squeeze system or something. |
| 18:27.36 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 18:36.37 | *** join/#asterisk synthetiq (~User@208.90.33.43) |
| 18:37.34 | synthetiq | an an invite sdp line, what indicated that dtmf is inband? a=fmtp:101 0-15 ? |
| 18:39.50 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 18:44.59 | *** join/#asterisk |Physis| (physisheck@186.213.36.247) |
| 18:50.30 | cj | fauxalliance: so, which one is the FULL FEATURED PBX? |
| 18:52.24 | \DSAFEW\ | cj, my guess is that the asterisk gui is just a shell for the configs |
| 18:58.24 | *** join/#asterisk g-ram (~gsaathoff@12.200.95.45) |
| 18:58.42 | *** part/#asterisk g-ram (~gsaathoff@12.200.95.45) |
| 19:03.18 | *** join/#asterisk |Physis| (physisheck@186.213.36.247) |
| 19:08.24 | *** join/#asterisk Dryanta (dryanta@dev.hockingits.com) |
| 19:08.28 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:08.29 | Dryanta | sup frens |
| 19:08.44 | Dryanta | hey i have a dialplan with my freepbx install from the michigan telephone google voice howto |
| 19:09.48 | Dryanta | and im trying to figure out how to make it ignore the '1' google voice seems to prepend on my droid when i have sipdroid as the call manager |
| 19:09.51 | Dryanta | so wat do? |
| 19:11.20 | Dryanta | 215 people idling? lolol |
| 19:11.34 | Freeaqingme | Dryanta, look at the webcast that was broadcast ~2 weeks ago |
| 19:11.53 | Freeaqingme | it was solely on implementing google voice/talk |
| 19:11.53 | Dryanta | Freeaqingme: no, im rolling my own solution and my setup works perfect |
| 19:12.00 | Dryanta | this is an asterisk call plan question |
| 19:12.08 | Freeaqingme | yes, and in that webcast they say how to handle that 1 |
| 19:12.10 | Dryanta | not a 'hao do i google voice guiz' |
| 19:12.15 | Dryanta | ok url plz? |
| 19:12.18 | Freeaqingme | google.com |
| 19:12.22 | Dryanta | id prefer the straight answer |
| 19:12.25 | *** join/#asterisk marlowe (~marlowe@ip68-100-150-54.dc.dc.cox.net) |
| 19:12.26 | Dryanta | instead of LMGTFY |
| 19:12.38 | Freeaqingme | Dryanta, sure, if you tell me where to send the invoice to... |
| 19:12.48 | *** part/#asterisk FeyFre (~panych_y@cpe-109-108-233-12.enet.vn.ua) |
| 19:12.50 | Dryanta | that kind of pomposity on freenode reminds me of #freebsdhelp on efnet in the 90s :p |
| 19:13.10 | Freeaqingme | Dryanta, I'm telling you how to find it. you're supposed to do a bit of searching youself as well ;) |
| 19:13.15 | Dryanta | and if i was incapable of figuring it out to the point where i was going to pay somebody... it wouldnt be off freenode irc |
| 19:13.16 | Dryanta | no |
| 19:13.24 | Dryanta | i already did weeks of research |
| 19:13.28 | Dryanta | have 99.99% of it done |
| 19:13.32 | Dryanta | i need the last 0.1% |
| 19:13.36 | Freeaqingme | digium has done 1 webcast |
| 19:13.38 | fauxalliance | sighs |
| 19:13.42 | Freeaqingme | that should not take weeks to find |
| 19:13.58 | fauxalliance | depends on how quickly google index's |
| 19:13.59 | Dryanta | you are spending more time and effort being argumentative than answering my question |
| 19:14.30 | Dryanta | which is general enough and specific enough (obviously ive done my fair share of work) |
| 19:14.30 | Dryanta | im not asking stupid n00b questions |
| 19:14.40 | Dryanta | <-- enterprise network architect with multimillion dollar practice |
| 19:14.45 | fauxalliance | <Dryanta> +hey i have a dialplan with my freepbx install from the michigan telephone google voice howto |
| 19:14.49 | Chainsaw | Dryanta: You are failing to put in the work. |
| 19:14.50 | fauxalliance | ^^^take it to #freepbx |
| 19:14.52 | Dryanta | cricket, verizon, facebook... my clients |
| 19:14.58 | Dryanta | fauxalliance: no |
| 19:14.59 | Freeaqingme | ...sure |
| 19:15.02 | Dryanta | dont be a dick |
| 19:15.04 | Chainsaw | Dryanta: Regardless of how much you are paid, you are going to have to share some of that with Freeaqingme if you want your work done for you. |
| 19:15.14 | fauxalliance | Dryanta, i put that blog in the bot... |
| 19:15.18 | Dryanta | i dont want WORK DONE FOR ME I WANNA KNOW DIALPLAN |
| 19:15.22 | fauxalliance | take it to #freepbx |
| 19:15.32 | Freeaqingme | I rest my case :D |
| 19:15.39 | fauxalliance | Freeaqingme, o/ |
| 19:15.57 | Dryanta | fuck most engineers like you never build something like zuck and i do because you are too busy feeling self important and talking down to people |
| 19:16.06 | Freeaqingme | sure |
| 19:16.13 | Freeaqingme | and I like it |
| 19:16.45 | Dryanta | you dont realize you are talking to your better, and if you would help me with my shit and give me what i required... maybe go above and beyond... i totally would not only pay on your invoice but offer you employment |
| 19:16.56 | psilikon | Dryanta, it has been a long time since I used sipdroid. Can you rephrase your question and I'll try to help. |
| 19:16.59 | Dryanta | i came here because i didnt want to bash my head against it |
| 19:17.03 | Dryanta | psilikon: <333333333333333333 |
| 19:17.09 | fauxalliance | Dryanta, try freedoh.. if it still dont' work. |
| 19:17.09 | fauxalliance | <mzb-> [11:02:07] +ValiumMm is on conf to ensure that someone will be on if/when Jeff is able to call in |
| 19:17.09 | fauxalliance | <mzb-> [11:46:39] +http://maps.google.com.au/maps?q=215+carella+street,+howrah&ie=UTF8&hq=&hnear=215+Carella+St,+Howrah+Tasmania+7018&gl=au&ll=-42.900647,147.416782&spn=0.012119,0.023754&z=16&layer=c&cbll=-42.900736,147.416853&panoid=XxoS5mcFnOjU_OlW3qrWKg&cbp=12,263.94,,0,9.32 |
| 19:17.09 | fauxalliance | <mzb-> [11:52:55] +http://maps.google.com.au/maps?hl=en&biw=1488&bih=894&q=28+elliott+road,+glenorchy&um=1&ie=UTF-8&hq=&hnear=28+Elliott+Rd,+Glenorchy+TAS+7010&gl=au&ei=f1OoTdKyNMTlrAeB05CnCA&sa=X&oi=geocode_result&ct=image&resnum=1&ved=0CBYQ8gEwAA |
| 19:17.14 | fauxalliance | <ValiumMm> [11:55:44] -http://maps.google.com.au/maps?hl=en&biw=1488&bih=894&q=28+elliott+road,+glenorchy&um=1&ie=UTF-8&hq=&hnear=28+Elliott+Rd,+Glenorchy+TAS+7010&gl=au&ei=f1OoTdKyNMTlrAeB05CnCA&sa=X&oi=geocode_result&ct=image&resnum=1&ved=0CBYQ8gEwAA |
| 19:17.14 | Dryanta | its not a sipdroid issue, when i use sipdroid to complete calls |
| 19:17.18 | fauxalliance | <ValiumMm> [11:56:41] -http://maps.google.com/maps?f=q&source=s_q&hl=en&geocode=&q=27+Melaleuca+Drive,+St+Ives,+New+South+Wales,+Australia&aq=0&sll=37.0625,-95.677068&sspn=52.372705,97.294922&ie=UTF8&hq=&hnear=27+Melaleuca+Dr,+St+Ives+New+South+Wales+2075,+Australia&ll=-33.737055,151.174278&spn=0.006781,0.011877&t=h&z=17 |
| 19:17.22 | fauxalliance | <***> Playback Complete. |
| 19:17.24 | fauxalliance | * ValiumMm has quit (Read error: Connection reset by peer) |
| 19:17.25 | *** kick/#asterisk [fauxalliance!~pabelange@50.22.5.41-static.reverse.softlayer.com] by pabelanger (use pastebin) |
| 19:17.36 | *** join/#asterisk fauxalliance (~fauxallia@142.163.152.120) |
| 19:17.38 | fauxalliance | * ValiumMm_ is now known as ValiumMm |
| 19:17.40 | fauxalliance | * ValiumMm has quit (Quit: ChatZilla 0.9.86.1 [Firefox 4.0/20110318052756]) |
| 19:17.41 | Dryanta | and the contacts dont have a prepending 1 |
| 19:17.44 | fauxalliance | <fauxalliance> Wie sagt mann 'Please feel free to have a seat with the other bitches waiting for me to give a fuck |
| 19:17.47 | fauxalliance | whoops |
| 19:17.49 | fauxalliance | pabelanger, duly noted ;) |
| 19:17.52 | Dryanta | for example 818.457.6605 (my number) |
| 19:17.52 | Chainsaw | Again, do it again! |
| 19:18.01 | fauxalliance | not likely |
| 19:18.04 | Dryanta | completes fine |
| 19:18.21 | Dryanta | but when im in my google voice inbox and click the 'call' like off a text message or something |
| 19:18.34 | Dryanta | it adds the 1 instead of going off the android contact list |
| 19:18.44 | pabelanger | And lets watch the language, it is unnecessary. |
| 19:18.59 | fauxalliance | pabelanger, duly noted :( |
| 19:19.02 | Dryanta | so i need to change the dialplan field to ignore any 1 and only work off last 8 digits |
| 19:19.06 | Dryanta | i mean 7' |
| 19:19.10 | Dryanta | make sense? |
| 19:19.16 | Dryanta | i can do webex with you if it doesnt |
| 19:20.06 | fauxalliance | pabelanger, the whinging just irritates me. |
| 19:20.22 | Dryanta | 'whining' you mean? |
| 19:20.32 | Dryanta | matter of perspective |
| 19:20.32 | fauxalliance | Dryanta, apt-get install dict |
| 19:20.37 | psilikon | Dryanta, like using ":1" to strip off a digit? as in Dial(SIP/${EXTEN:1}) so if extension was 15555551212 it would dial 5555551212? |
| 19:20.43 | Dryanta | fauxalliance: fuck ubuntu/debian |
| 19:20.50 | Dryanta | Linux irc-staging 2.6.35.4-rscloud #8 SMP Mon Sep 20 15:54:33 UTC 2010 x86_64 GNU/Linux |
| 19:20.51 | fauxalliance | language my child |
| 19:20.57 | Dryanta | lol @ child |
| 19:21.06 | Dryanta | this is what i irc on |
| 19:21.12 | fauxalliance | Dryanta, apparently you have a lot to learn yet 'young fella' |
| 19:21.22 | Dryanta | because if i get gigabits of ddos i can load balance it/null route it without dropping |
| 19:21.27 | Dryanta | fauxalliance: thats where you're wrong |
| 19:21.34 | pabelanger | Dryanta: Do you need to drop the f-bomb to make your point? |
| 19:21.41 | Dryanta | understanding others is wisdom, understanding ones self is enlightenment -lao tsu |
| 19:21.44 | fauxalliance | pabelanger, no one will listen otherwise |
| 19:21.45 | Freeaqingme | did someone already say to ignore the troll? |
| 19:21.49 | fauxalliance | nah... |
| 19:21.51 | Dryanta | pabelanger: um... idk do i? |
| 19:21.51 | fauxalliance | drive him out |
| 19:22.04 | pabelanger | ~ask |
| 19:22.04 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:22.07 | Dryanta | im not a troll, i just dgaf about trivial peoples opinions |
| 19:22.09 | Chainsaw | Just don't assist. Unsociable people have to pay. |
| 19:22.16 | Dryanta | pabelanger: i did exactly that if you review buffer |
| 19:22.17 | Dryanta | lol |
| 19:22.28 | pabelanger | ~patience |
| 19:22.28 | infobot | patience is, like, a Godly attribute, or the solution for most things. |
| 19:22.41 | fauxalliance | whinge whinge whinge.... again... >>#freepbx |
| 19:22.54 | Dryanta | i have unlimited patience for solving problems and doing things perfectly, i have zero patience for disrespect |
| 19:23.04 | fauxalliance | Dryanta, kudos |
| 19:23.04 | Dryanta | respect is met in kind, disrespect as well |
| 19:23.12 | Dryanta | you get what you give in this world xD |
| 19:23.24 | fauxalliance | Dryanta, you reap what you sow.. MY PBX WORKS :) |
| 19:23.31 | Dryanta | psilikon: tyvm <3 |
| 19:23.46 | Dryanta | fauxalliance: my enterprise platform works, this is development/test |
| 19:23.48 | Dryanta | staging |
| 19:23.59 | Dryanta | not even nascent pre-production testing |
| 19:24.30 | Dryanta | which will be fully regression tested in multiple carrier facilities with 10gb datacenter fabric switch intercoonnect, behind two fully populated f5 viprions |
| 19:24.40 | Dryanta | :p |
| 19:24.58 | Dryanta | again, my clients include.... leap/cricket and vzw |
| 19:25.01 | Dryanta | tmo |
| 19:25.02 | fauxalliance | pabelanger ++ |
| 19:25.03 | Dryanta | microsoft |
| 19:25.08 | fauxalliance | Dryanta, get bent |
| 19:25.08 | _Corey_ | uh huh |
| 19:25.09 | Dryanta | et. al |
| 19:25.10 | Freeaqingme | and yet you want your answer to be spelled out here, and when it has, you choose to ignore it? |
| 19:25.28 | Dryanta | fauxalliance: why are you mad? because you got owned? rofl |
| 19:25.41 | *** join/#asterisk jstapleton (~jstapleto@173.15.197.75) |
| 19:25.43 | *** mode/#asterisk [+q Dryanta!*@*] by Qwell |
| 19:25.44 | Qwell | shut up |
| 19:25.49 | Chainsaw | Thank you Qwell. |
| 19:25.53 | _Corey_ | lol |
| 19:25.53 | fauxalliance | Qwell, thanks mate |
| 19:26.02 | _Corey_ | seriously |
| 19:26.10 | fauxalliance | yeah.. seriously! |
| 19:26.17 | leifmadsen | fauxalliance: you're not helping |
| 19:26.23 | pabelanger | fauxalliance: Don't feed the trolls |
| 19:26.43 | fauxalliance | leifmadsen, i am having a bad day, with your respect.. .I'll try to be more helpful |
| 19:26.51 | *** mode/#asterisk [+q fauxalliance!*@*] by leifmadsen |
| 19:26.55 | leifmadsen | you can be more helpful later |
| 19:29.14 | Qwell | Let it be known that I told "The next Zuck" to shut up. |
| 19:29.24 | Qwell | Dryanta: Put me in your novel, would ya? |
| 19:30.00 | *** join/#asterisk Karen_m (~karen@66.222.153.231) |
| 19:30.05 | Karen_m | what is the itu-t code for canada/us |
| 19:30.11 | Chainsaw | Qwell: Let's face it though. This is IRC. It could just be any dude in any basement... |
| 19:30.12 | leifmadsen | 1 ? |
| 19:30.23 | leifmadsen | we're all dudes in a basement :) |
| 19:30.29 | Qwell | leifmadsen: you're thinking NANP |
| 19:30.34 | russellb | I am not in a basement. |
| 19:30.39 | leifmadsen | russellb: either am I |
| 19:30.40 | Chainsaw | I am on a train. |
| 19:30.42 | leifmadsen | Qwell: oh right |
| 19:30.48 | pabelanger | I am :( |
| 19:30.55 | serafie | dibs, I get to play Qwell in The TeleNetwork!!!1! |
| 19:31.12 | russellb | serafie: good call |
| 19:31.15 | Qwell | serafie: ! |
| 19:31.17 | Qwell | wait |
| 19:31.20 | russellb | who do I get to be? :-( |
| 19:31.27 | Qwell | russellb: You can be Dryanta ? |
| 19:31.36 | Qwell | or one of the twins! |
| 19:31.36 | russellb | sure, why not |
| 19:31.41 | leifmadsen | I want to play Russell B |
| 19:31.44 | leifmadsen | or drumkilla |
| 19:31.50 | Karen_m | thank you |
| 19:31.50 | russellb | I'll be blitzrage |
| 19:31.56 | leifmadsen | hawt |
| 19:32.33 | russellb | and i'll wear pink the whole time |
| 19:32.40 | synthetiq | an an invite sdp line, what indicated that dtmf is inband? a=fmtp:101 0-15 ? |
| 19:32.42 | leifmadsen | *shaky fist* |
| 19:33.02 | pabelanger | Can I be frightened inmate number two? |
| 19:33.09 | leifmadsen | lol |
| 19:34.01 | _Corey_ | good one |
| 19:35.55 | voxter | leifmadsen: mr madsen! |
| 19:36.04 | *** join/#asterisk marlowe (~marlowe@ip68-100-150-54.dc.dc.cox.net) |
| 19:37.55 | *** part/#asterisk Dryanta (dryanta@dev.hockingits.com) |
| 19:40.37 | BlackBishop | wonders how to log in .mp3/.gsm/.something the calls incoming/outgoing from his asterisk server |
| 19:41.28 | russellb | BlackBishop: see the Asterisk Cookbook on ofps.oreilly.com |
| 19:41.32 | *** join/#asterisk marlowe (~marlowe@ip68-100-150-54.dc.dc.cox.net) |
| 19:41.35 | sled-dog | an audio log? You could do something silly like festival, I guess |
| 19:41.36 | russellb | it has a recipe (or a couple) on that |
| 19:42.31 | leifmadsen | voxter: ohai! |
| 19:43.08 | Freeaqingme | When setting up a hotdesking setup, how do you make sure your users log out (dont forget)? |
| 19:43.27 | Chainsaw | Freeaqingme: From experience... you pick a time when you know everyone's supposed to have gone and you log them out automatically. |
| 19:43.28 | leifmadsen | you don't? |
| 19:43.30 | Qwell | Freeaqingme: get a cattleprod |
| 19:43.37 | leifmadsen | heh ya :) |
| 19:43.41 | leifmadsen | cronjob |
| 19:43.46 | Chainsaw | Freeaqingme: That and public mocking of whoever forgets. |
| 19:44.03 | russellb | we have a recipe on hot desking, too! |
| 19:44.05 | *** join/#asterisk marlowe (~marlowe@ip68-100-150-54.dc.dc.cox.net) |
| 19:44.11 | Freeaqingme | I see, tnx |
| 19:44.28 | leifmadsen | totally do! |
| 19:44.40 | Chainsaw | Freeaqingme: If you have control of the DHCP server, you could probably log their phones in & out as leases are granted & revoked. |
| 19:44.52 | BlackBishop | russellb: festival as in text-to-speach ? |
| 19:44.56 | *** join/#asterisk wonderworld (~ww@port-92-201-228-210.dynamic.qsc.de) |
| 19:44.57 | Chainsaw | Freeaqingme: I've been wanting to do this, but ISC DHCP does not seem to have the ability to cool hook scripts in a useful way. |
| 19:45.07 | Chainsaw | Freeaqingme: Cool? Call. One day I'll learn to spell. |
| 19:45.35 | Karen_m | after you make a change to 'asterisk.conf', how do you reload it? /etc/init.d/asterisk? or is there a better way |
| 19:45.43 | russellb | restart asterisk |
| 19:45.52 | russellb | i think that's the only way for that file |
| 19:46.02 | Freeaqingme | Chainsaw, nah, the solution I'm working on will be partially hosted, so hooking in to the dhcp server isn't really possible, but I'll work something out |
| 19:46.17 | Chainsaw | Freeaqingme: Ah, quite. |
| 19:46.21 | *** join/#asterisk wonderworld (~ww@port-92-201-228-210.dynamic.qsc.de) |
| 19:46.34 | leifmadsen | russellb: yep restart required |
| 19:46.43 | leifmadsen | core file, so you need to restart -- nothing to reload it |
| 19:47.27 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 19:48.52 | _Corey_ | Freeaqingme: I've found the best solution to be displaying who's logged in on the phone's screen at all times |
| 19:49.08 | _Corey_ | Freeaqingme: if you're using Polycom you can use an idle URL |
| 19:49.21 | Freeaqingme | hmmz, that sounds cool. |
| 19:49.35 | Freeaqingme | got grandstream though, could modify some of its screen perhaps |
| 19:49.37 | _Corey_ | rig something with PHP to pull from Asterisk based on the requestor ip |
| 19:49.48 | *** join/#asterisk marlowe (~marlowe@ip68-100-150-54.dc.dc.cox.net) |
| 19:50.36 | Karen_m | I am trying to do the Hello World example with the oreily book, however when I call my phone number to test it, I keep getting; chan_sip.c:20152 handle_request_invite: call from xxx to extension 's' rejected because extension not found in context 'mycontext', why is that? |
| 19:50.43 | _Corey_ | eh, you may be out of luck with grandstream |
| 19:51.49 | Freeaqingme | _Corey_, you can modify some screens with gxp, but I'm not sure how dynamic that is |
| 19:52.11 | Karen_m | what is extension 's'? |
| 19:52.18 | Karen_m | is that supposed to by my phone number or something? |
| 19:54.35 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 19:55.34 | leifmadsen | Karen_m: explained in the documentation at http://ofps.oreilly.com |
| 19:55.38 | leifmadsen | that's a common question |
| 19:55.53 | leifmadsen | 's' is the default extension for analog phones |
| 19:56.04 | leifmadsen | it may also be what asterisk uses for registration |
| 19:56.14 | leifmadsen | (depending on your configuration or what the other side does) |
| 19:56.16 | Karen_m | I'm on chapter 5, trying to do the hello world, i have never come across a mention of 's' |
| 19:56.25 | *** join/#asterisk b2 (~ion@pdpc/supporter/active/beckb) |
| 19:56.27 | *** join/#asterisk wonderworld (~ww@port-92-201-228-210.dynamic.qsc.de) |
| 19:56.58 | Karen_m | I put my sip.conf, context=testing .. and then in the [testing] i don't even mention 's', I am using their example for hello-world |
| 19:57.53 | leifmadsen | what are you calling? |
| 19:57.57 | leifmadsen | or what is calling you? |
| 19:58.03 | *** join/#asterisk marlowe (~marlowe@ip68-100-150-54.dc.dc.cox.net) |
| 19:58.10 | leifmadsen | 's' is the extension that is being requested |
| 19:58.19 | Karen_m | I am trying to call the phone number i have registered with voip.ms |
| 19:58.36 | leifmadsen | then voip.ms by default is probably trying to send it to the 's' extension |
| 19:58.40 | Karen_m | the book says, 'try it now'. I so cut/paste the voip.ms instructions, changed the context to 'testing', set up the hello world and called the number |
| 19:58.50 | leifmadsen | change your registration line to end in /<number_or_extension_you_want_dialed> |
| 19:59.03 | leifmadsen | you sure did |
| 19:59.26 | Karen_m | Where do I find the registration line? the one in sip.conf ? |
| 19:59.31 | leifmadsen | it'd be the only one :) |
| 20:00.29 | Karen_m | so I have register => <user>:<pass>@host:port , it can optionally take ... /<number_i_want_dialed>? |
| 20:00.51 | Karen_m | so for extension 200, i would do; register=>123:test@host.com:5060/200 |
| 20:03.35 | Karen_m | leifmadsen, so is there a way to print the default extension that will be called? |
| 20:04.52 | *** join/#asterisk marlowe (~marlowe@ip68-100-150-54.dc.dc.cox.net) |
| 20:05.12 | Karen_m | wow it worked, yay |
| 20:05.19 | Karen_m | ;/200 worked |
| 20:05.22 | Karen_m | added on |
| 20:05.55 | Karen_m | I have a weird situation going on with hello world tho, for some reason I don't here the whole thing |
| 20:06.47 | *** join/#asterisk marlowe (~marlowe@ip68-100-150-54.dc.dc.cox.net) |
| 20:06.52 | *** join/#asterisk ]loy[ (~nobody@95.72.26.98) |
| 20:08.13 | *** join/#asterisk Guifort (~Guifort@78.112.90.5) |
| 20:08.52 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 20:09.33 | *** join/#asterisk marlowe (~marlowe@ip68-100-150-54.dc.dc.cox.net) |
| 20:10.26 | Karen_m | leifmadsen, the weird part is the first hello-world gets cut off a bit, but i cut/pasted the line 4 times and i hear it completely the second thru end .. it's awesome!! ok, i can continue reading the book :) |
| 20:14.59 | Karen_m | sounds great and there is no echo/delay though, so it's going to be |
| 20:19.20 | *** join/#asterisk jetlag (~jetlag@pool-173-61-239-190.cmdnnj.east.verizon.net) |
| 20:21.21 | *** join/#asterisk wpvanv (437b78c5@gateway/web/freenode/ip.67.123.120.197) |
| 20:25.05 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 20:36.49 | wpvanv | Can anyone give me a rough estimate on how many days it would take a professional to install/configure an Asterisk PBX with the following setup: Xorcom Asterisk appliance (Elastix distribution), single PRI, 40 DIDs (but only 5-10 used initially), new POE switch, 45 new Aastra phones, 2 fax machines attached to FXS ports, single-site, separate LAN/cabling for voice, basic corporate auto-attendant (dial-by-extension + dial-by-name), |
| 20:37.18 | *** join/#asterisk wonderworld (~ww@port-92-201-228-210.dynamic.qsc.de) |
| 20:39.05 | Freeaqingme | wpvanv, I'm a relative noob, but I'd say less than a week |
| 20:43.33 | _Corey_ | wpvanv: it could be <1 day if everything is ready to go... depends on many things |
| 20:44.15 | _Corey_ | wpvanv: when in doubt, obtain a competitive quote |
| 20:44.53 | Qwell | depends on how much you're paying |
| 20:45.18 | Qwell | $ = a week or more. $$ = a week. $$$ = a few hours. $$$$ = a month. |
| 20:45.24 | Qwell | (not to scale) |
| 20:46.23 | *** join/#asterisk gruvfunk (~gruvfunk@user-160uac8.cable.mindspring.com) |
| 20:46.44 | wpvanv | Thanks for the info. Not much competition around, unfortunately. I don't want to waste someone's time if they're too far away to get the business. |
| 20:47.08 | _Corey_ | Where are you located? |
| 20:48.01 | *** join/#asterisk sahX (~sahX@99-105-56-250.lightspeed.sntcca.sbcglobal.net) |
| 20:48.20 | wpvanv | Middle of nowhere Nevada |
| 20:48.25 | gruvfunk | Having the fight of my life today: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached - Packet timed out after 6400ms with no response. Anyone have any leads on this one? |
| 20:48.43 | gruvfunk | server is on a public IP address |
| 20:48.47 | _Corey_ | wpvanv: Digium's website has a contact form, you could always request a referral there |
| 20:48.59 | _Corey_ | there may be a certified reseller nearby |
| 20:49.51 | wpvanv | _Corey_: Thanks, I'll do that. |
| 20:50.09 | gruvfunk | is it a network / provider issue? there is no NAT in place.. |
| 20:50.13 | _Corey_ | good luck |
| 20:50.15 | *** join/#asterisk ks3 (~ksandy@74.203.195.1) |
| 20:51.03 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 20:51.39 | gruvfunk | outbound works a peach, inbound gets 2 seconds of IVR and nukes |
| 21:03.02 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 21:03.39 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:05.06 | *** join/#asterisk itsbroken (~hater@lolunix.org) |
| 21:05.22 | cj | alright, I've got a somewhat complex question. Say I'm doing round-robin load balancing of incoming registration requests. Client0 makes a registration request and the load balancer sends it to Server0. But at a later time, it could be directed to Server1 or ServerN. I assume that each of these hosts needs to have an entry in its sip.conf channel config file. If Client1 makes a call to Client0 and Server0 .. ServerN all have entries for C |
| 21:06.07 | Qwell | Put something like SER in front |
| 21:06.13 | pigpen | Hi all. I have an audiocodes 4 port fxo. It works great inbound, however, outbound, it picks up the line, and dead air. I get "Invalid RTCP packet SSRC" in the debug on the unit. |
| 21:06.16 | cj | SER? |
| 21:07.34 | cj | ah, SIP Express Router. |
| 21:07.43 | cj | Qwell: what benefit would this provide? |
| 21:07.53 | pigpen | yeah, they split from openser into two forks. |
| 21:08.03 | pigpen | pretty sure that is what he was referring to. |
| 21:10.35 | cj | okay, so the question remains, is Asterisk smart enough to recognize which one of the hosts in the load balanced pool the client is registered to and direct the call there via DUNDi? |
| 21:12.27 | cj | maybe I should be looking at OpenAIS |
| 21:22.45 | cj | looks like XMPP might be more up my alley |
| 21:31.17 | *** join/#asterisk wonderworld (~ww@port-92-201-228-210.dynamic.qsc.de) |
| 21:37.07 | gruvfunk | Anyone?? chan_sip.c:3386 retrans_pkt: Retransmission timeout reached - Packet timed out after 6400ms with no response |
| 21:37.48 | pigpen | Sounds like something is not listening, or like a misconfiguration, ie: wrong ip address or such. |
| 21:37.54 | pigpen | quick guess. |
| 21:38.25 | \DSAFEW\ | I need to see why my AMI script isn't sending the right action, is there any way to be more verbose in the server console? I'm doingasterisk -vvvvvvvvvvddddgr |
| 21:38.36 | \DSAFEW\ | do I need more vs and ds? |
| 21:39.17 | \DSAFEW\ | is there a way to read API actions without parsing sniffer logs? |
| 21:43.03 | \DSAFEW\ | gruvfunk, could it possibly be a NAT issue? UDP or TCP? |
| 21:43.23 | gruvfunk | I'm told there is no NAT, server is on a public IP address, no firewall even |
| 21:43.46 | \DSAFEW\ | gruvfunk, but you are behind a nat? |
| 21:44.18 | \DSAFEW\ | gruvfunk, hard not to be, unless your phone is set on the wrong side of the firewall |
| 21:44.27 | pigpen | for that matter an isp along the way could be blocking some traffic. Especially if it is a residential line |
| 21:44.54 | gruvfunk | that's my thought as well - ISP issue |
| 21:45.00 | \DSAFEW\ | gruvfunk, if outbound works fine, and inbound is testy, I'd really suspect it's a firewall issue |
| 21:45.24 | \DSAFEW\ | perhaps the ISP's firewalls, but maybe your own, is it TCP or UDP? |
| 21:45.44 | gruvfunk | again, no firewall is what I'm being told |
| 21:45.54 | *** join/#asterisk wesphillips (~wphill04@adsl-76-247-249-160.dsl.hstntx.sbcglobal.net) |
| 21:45.57 | pigpen | And, it would be unlikely that this server is not protected in some form or fashion. If not, plan on it being exploited soon. |
| 21:46.25 | gruvfunk | pigpen: well.. it was.. and we had issues, so to prove the firewall was the issue we changed the server ,shipped it to a data center to be hosted |
| 21:46.42 | gruvfunk | now that it's on a public link, same issue |
| 21:47.21 | pigpen | heh, maybe it is already compromised, and they are blocking traffic. :-) |
| 21:47.59 | \DSAFEW\ | gruvfunk, make sure you're provider's not going to charge for calls to Paraguay |
| 21:47.59 | pigpen | so what is the client trying to connect to this hosted server? |
| 21:48.24 | gruvfunk | it's an IVR with information for callers to a toll free |
| 21:48.45 | gruvfunk | simple solution really |
| 21:50.07 | gruvfunk | pigpen: interesting concept |
| 21:50.32 | gruvfunk | if it was compromised wouldn't I see activity? in the logs, the debug? |
| 21:50.48 | *** join/#asterisk wesphillips (~wphill04@adsl-76-247-249-160.dsl.hstntx.sbcglobal.net) |
| 21:52.45 | pigpen | if you know what you are looking for. |
| 21:54.18 | pigpen | Here is the thing about sip over the internet. It sounds easy, and it can be. People do it all the time. |
| 21:54.28 | gruvfunk | I sure have... |
| 21:54.51 | gruvfunk | the only time I've seen this issue and I did not participate in resolving it was when a customer used clear.com 4G USB wireless device as their provider |
| 21:55.07 | pigpen | I have never liked doing that. We have been running our sip phones to a asterisk box at our datacenter for 6 years....but we push the traffic inside a VPN. |
| 21:55.14 | gruvfunk | after beating our heads, after swapping it with a Sprint device, issue was resolved |
| 21:55.29 | pigpen | We were doing this when everyone said not to, because it wouldn't work. Well, they have all changed their tune. |
| 21:56.18 | gruvfunk | it's a good measure, and I applaud your effort to keep voice traffic private, for sure |
| 21:57.18 | *** part/#asterisk wesphillips (~wphill04@adsl-76-247-249-160.dsl.hstntx.sbcglobal.net) |
| 22:00.47 | \DSAFEW\ | so I've got this AMI manager javascript thing called Asterisk Click2Dial Extension for firefox, which I need to work. My problem is that it is reversing the channel and the extension when it originates a call |
| 22:01.48 | \DSAFEW\ | I'm open to any suggestions. |
| 22:02.35 | *** join/#asterisk sahX (~sahX@99-105-56-250.lightspeed.sntcca.sbcglobal.net) |
| 22:03.29 | \DSAFEW\ | looking at this, the example is exactly what my script is trying to do, http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate |
| 22:03.52 | \DSAFEW\ | but in practice, that doesn't work, no matter how I tweak the script settings, I guess I need to fix my dialplans? |
| 22:04.34 | \DSAFEW\ | mine works like the second example |
| 22:18.00 | *** join/#asterisk cyford (~cyford@99-127-135-189.lightspeed.tukrga.sbcglobal.net) |
| 22:19.06 | pabelanger | Look at me, I'm installing AsteriskNOW :) |
| 22:19.51 | _Corey_ | it's just like regular Asterisk, but NOW... |
| 22:24.54 | *** join/#asterisk rdahlin_1 (~rdahlin_1@2001:16d8:cc97:1:21f:5bff:fe37:c2c9) |
| 22:31.31 | Freeaqingme | _Corey_, in that case I'm upgrading as we speak to AsteriskFewWeeksBack |
| 22:46.59 | *** join/#asterisk ariel_ (~chatzilla@unaffiliated/abatista) |
| 22:49.43 | psilikon | \DSAFEW\, can you paste the script somewhere? |
| 22:50.47 | \DSAFEW\ | psilikon, it's not mine, and it's large, you can get it yourself here: https://addons.mozilla.org/en-us/firefox/addon/asterisk-click2dial/ |
| 22:50.54 | psilikon | \DSAFEW\, oh wait nevermind I looked at the log and see that this is a Click2Dial. Yea you need to probably adjust the dialplan. |
| 22:51.05 | \DSAFEW\ | psilikon, I'm looking at the source and it's almost incomprehensible |
| 22:51.06 | *** join/#asterisk jong2 (~chatzilla@63.224.204.153) |
| 22:51.20 | psilikon | \DSAFEW\, yeah I don't know js |
| 22:53.18 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 22:53.57 | \DSAFEW\ | psilikon, well how would I make 'originate SIP/1001 extension 5551234' work like 'originate 5551234 extension SIP/1001' does? |
| 22:54.14 | \DSAFEW\ | as far as extensions.conf is.. |
| 22:54.53 | psilikon | \DSAFEW\, what context is C2D setup with? |
| 22:54.57 | \DSAFEW\ | I notice it says "channel" and "extension", so uhh |
| 22:55.04 | \DSAFEW\ | let me pastebin |
| 22:56.39 | \DSAFEW\ | psilikon, sorry, just read your question, it's default, and I've tried my from-sip, but that only makes it ring for one second and hang up |
| 22:56.56 | *** join/#asterisk bmg505 (~leon@196-209-44-209.dynamic.isadsl.co.za) |
| 22:58.13 | \DSAFEW\ | dialplan is here http://paste.pocoo.org/show/372565/ |
| 23:00.42 | \DSAFEW\ | the part in default about 1001 is just me fiddling around, I don't think it would ever get called |
| 23:00.52 | \DSAFEW\ | I mean the extension |
| 23:01.04 | \DSAFEW\ | <PROTECTED> |
| 23:01.07 | \DSAFEW\ | that's not used |
| 23:02.49 | \DSAFEW\ | I'm not sure what to make of this... |
| 23:02.52 | \DSAFEW\ | dahdi show channels |
| 23:02.52 | \DSAFEW\ | <PROTECTED> |
| 23:02.52 | \DSAFEW\ | <PROTECTED> |
| 23:02.52 | \DSAFEW\ | <PROTECTED> |
| 23:03.13 | carrar | Looks like you've been hacked |
| 23:03.38 | \DSAFEW\ | carrar, haha, I'm sure it looks bad, I'm just in a hurry to make this work |
| 23:04.17 | psilikon | \DSAFEW\, how do you do outbound? |
| 23:04.18 | \DSAFEW\ | carrar, all the security will have to wait for when some dyslexic office worker can click a phone number to call |
| 23:05.15 | \DSAFEW\ | psilikon, there's one sip phone and one fxo card, they both call the other way |
| 23:05.28 | \DSAFEW\ | default is the inbound and from-sip is outbound |
| 23:06.28 | \DSAFEW\ | TRUNK = DAHDI/G1 here '_1NXXNXXXXXX' => 1. Dial(${TRUNK}/${EXTEN}) [pbx_config] |
| 23:07.01 | \DSAFEW\ | do you want any more config files? or suggestions on reading? |
| 23:11.23 | psilikon | \DSAFEW\, ok I just installed it and it worked like a champ. thanks I might start using this. |
| 23:11.46 | \DSAFEW\ | psilikon, lol, care to share some lovely dialplan secrets? :D |
| 23:12.25 | \DSAFEW\ | I probably don't need any sip info, other than the contexts which I can infer |
| 23:12.34 | psilikon | \DSAFEW\, you need to have a matching extension in the context that you specified to perform to oubound dial. So for me to dial a 10 digit # i needed NXXNXXXXXXX. |
| 23:12.58 | psilikon | \DSAFEW\, when I didn't have a matching extension it would ring once and then throw an error in the CLI. |
| 23:13.04 | \DSAFEW\ | oh |
| 23:13.11 | \DSAFEW\ | neat. thanks |
| 23:13.39 | psilikon | \DSAFEW\, so from what I see above you have a nice 11 digit extension ready to handle outbound, but is that in the context that your configured C2D with?? |
| 23:14.01 | \DSAFEW\ | nope |
| 23:14.05 | \DSAFEW\ | it was doing default |
| 23:14.19 | \DSAFEW\ | the from-sip did that strange one ring thing |
| 23:14.26 | psilikon | \DSAFEW\, you need something in that context. |
| 23:15.33 | \DSAFEW\ | rebooting phone, didn't reload dialplan in a soft way |
| 23:16.23 | \DSAFEW\ | okay, it's still ringing the extension and hanging up |
| 23:16.31 | psilikon | \DSAFEW\, configure C2D for 'from-sip'. Disconnect and then reconnect. Issue a 'reload' from the * cl also for good measure. |
| 23:16.49 | psilikon | \DSAFEW\, then try to dial a 7, 10 or 11 digit number. You |
| 23:17.01 | psilikon | your dialplan looks fine. |
| 23:19.22 | \DSAFEW\ | keep getting WARNING[13053] chan_sip.c: Failed to parse contact info |
| 23:19.36 | \DSAFEW\ | with the dialplan from-sip in firefox |
| 23:20.28 | psilikon | \DSAFEW\, paste your extensions.conf |
| 23:20.36 | \DSAFEW\ | just curious |
| 23:20.43 | \DSAFEW\ | Dial,(SIP/${EXTEN},20) is 20 like the seconds? |
| 23:20.58 | psilikon | \DSAFEW\, yes |
| 23:21.01 | \DSAFEW\ | because I have no option like that for the other numbers... |
| 23:21.19 | \DSAFEW\ | so if it were waiting to pick up the extension, it would need that number |
| 23:22.15 | psilikon | \DSAFEW\, not sure what you mean but first paste your actual extensions.conf somewhere and let me see it. |
| 23:22.31 | psilikon | \DSAFEW\, also outbound calls to the PSTN are working ok right? |
| 23:22.51 | \DSAFEW\ | http://paste.pocoo.org/show/372572/ |
| 23:22.55 | \DSAFEW\ | psilikon, yes. |
| 23:24.21 | \DSAFEW\ | moment, rebooting and trying with timer and from-sip context in firefox |
| 23:24.39 | psilikon | \DSAFEW\, comment out line 870 (exten => _1001,1,Dial,(SIP/${EXTEN},20)) then reload asterisk and try it again. |
| 23:24.59 | \DSAFEW\ | that didn't work |
| 23:25.29 | psilikon | \DSAFEW\, C2D is set for 'from-sip'? |
| 23:26.05 | \DSAFEW\ | correct |
| 23:27.11 | \DSAFEW\ | with 870 commented out, there's no change, WARNING[13388] chan_sip.c: Failed to parse contact info |
| 23:27.15 | \DSAFEW\ | rings once |
| 23:27.26 | carrar | ~book |
| 23:27.26 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 23:31.06 | \DSAFEW\ | to give you an idea of what that is doing psilikon, http://paste.pocoo.org/show/372574/ |
| 23:31.32 | \DSAFEW\ | this is the tcpstream of the firefox AMI manager |
| 23:32.48 | \DSAFEW\ | only the first part is sent, the rest is the response |
| 23:32.59 | psilikon | <PROTECTED> |
| 23:33.15 | psilikon | \DSAFEW\, i'll probably be around later and we can try to fix it. |
| 23:33.37 | \DSAFEW\ | kk,thanks for looking at it with me, I'll probably do some more context switching and see if that helps |
| 23:52.35 | *** join/#asterisk josephnexus (~josephnex@75-167-161-60.bois.qwest.net) |
| 23:53.13 | josephnexus | hello everyone, i'm looking to pay someone to help us integrate asterisk with a web app that i've written (the api is developed and just uses http post and reads back xml) if someone is interested, please pm me for details |
| 23:53.39 | psilikon | \DSAFEW\, the tcp stream from AMI actually looks good at first glance. How did you capture it out of curiosity? |
| 23:53.57 | \DSAFEW\ | wireshark |
| 23:54.16 | Freeaqingme | josephnexus, when do you need it done? |
| 23:54.21 | psilikon | \DSAFEW\, ah. I think if you telnet in your can see it as well. |
| 23:54.21 | Nugget | telnet is eeeeeeevil! |
| 23:55.31 | josephnexus | a few wks |
| 23:55.39 | josephnexus | Freeaqingme: a few wks |
| 23:55.41 | Freeaqingme | nope, too busy, sorry |
| 23:55.49 | josephnexus | know anyone i should contact? |
| 23:56.04 | Freeaqingme | you could try contacting digium, apparently they have a list of partners |
| 23:56.16 | \DSAFEW\ | josephnexus, digium for a premium service of course |
| 23:56.29 | josephnexus | yeah |
| 23:56.34 | \DSAFEW\ | probably someone in here if you spam that every 5 hours or so |
| 23:57.07 | Freeaqingme | What do you need to do to become digium partner? |
| 23:57.31 | \DSAFEW\ | I'm guessing an engineering degree and a lot of patience |
| 23:58.11 | marlowe | has anyone picked up O'Reilly's Asterisk Cookbook |
| 23:58.34 | marlowe | I would love to hear what you thought of it? |