| 00:02.10 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 00:02.53 | ketas | i can get some of those for home voip hacking |
| 00:09.57 | *** join/#asterisk darkbasic_ (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 00:12.20 | [TK]D-Fender | ketas: They work. They feel like cheap plastic junk, their featureset isn't the most solid but they do work. |
| 00:17.34 | ketas | what's the top class? |
| 00:17.37 | ketas | cisco? |
| 00:17.49 | [TK]D-Fender | Polycom has been the king for some time. |
| 00:17.57 | talntid | <3 polycom :) |
| 00:18.03 | ketas | heard that from somewhere too |
| 00:18.10 | [TK]D-Fender | Aastra has a few advantages along with tradeoffs in certain scenarios |
| 00:18.24 | ketas | will grandstream gxp2000 be worth of 10 eur for used device? |
| 00:18.31 | [TK]D-Fender | Sure |
| 00:19.06 | [TK]D-Fender | Checkout tim, I'm out for a bit... |
| 00:19.10 | ketas | owner wasn't sure how many work properly and promises money back |
| 00:19.25 | ketas | although why would they not work properly?! |
| 00:30.15 | *** topic/#asterisk by mjordan -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.1.3 (2012/02/23), 1.8.9.3 (2012/02/23), dahdi-linux 2.6.0 (2012/01/04), dahdi-tools 2.6.0 (2012/01/04), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 00:54.17 | *** join/#asterisk Jacke (jacke@valhalla.bofh.org) |
| 00:54.59 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-xabrwmvgvalbpaji) |
| 01:01.31 | butchevans2001 | http://pastebin.com/yyHezRKf Looking for help with asterisk/chan_sccp-b/mysql integration |
| 01:08.10 | *** join/#asterisk jgowdy (~jgowdy@basharteg.com) |
| 01:11.18 | jgowdy | When I bridge a call in Asterisk, it seems like the DSP is disabled, preventing me from detecting inband DTMF. I'm including 'k' in my Bridge command. Is there a way to keep the DSP and inband DTMF detection active during bridged calls? |
| 01:14.41 | snadge | is it worth upgrading to 10? it sounds so much better than 1.8.. like 9.2 better |
| 01:14.44 | snadge | :P |
| 01:15.42 | snadge | im only really interested in the faxing support.. apparently t38 is native now.. and 10 has some gateway mode, which i dont really understand what that means or if i could benefit from it :p |
| 01:24.52 | woleium | lo peeps :-) |
| 01:26.51 | woleium | I have purchased and licensed the g729 codec for asterisk, and I can "g729 show licenses" ad get output, but when I try to connect from a polycom handet that I have limited to G729AB only it doesn't work (chan_sip.c:8918 process_sdp: No compatible codecs, not accepting this offer!) - am I missing something? |
| 01:27.18 | jgowdy | do you have g729 loaded? |
| 01:27.48 | jgowdy | I guess it would have to be loaded to show the licenses |
| 01:27.53 | woleium | I don;t think the g729 command is available if it isn't loaded |
| 01:27.55 | jgowdy | right |
| 01:27.57 | jgowdy | my bad |
| 01:28.07 | woleium | but yes i have run "module load codec_g729a.so" |
| 01:28.47 | jgowdy | did you turn up the log detail and see what the codec negotiation looks like? |
| 01:29.52 | woleium | i've turned on sip debug for the extension, but it doesn't sam to say anything more useful than "SIP/2.0 488 Not acceptable here" |
| 01:30.15 | woleium | is there a codec negotiation debut command? |
| 01:30.21 | woleium | drbug^ |
| 01:30.34 | woleium | ahem, debug |
| 01:31.00 | *** join/#asterisk DavidB2 (57dabd65@gateway/web/freenode/ip.87.218.189.101) |
| 01:31.11 | *** join/#asterisk dijib (~root@bas10-kitchener06-1176001599.dsl.bell.ca) |
| 01:31.19 | dijib | so who wants a sangoma card ??/ |
| 01:31.22 | dijib | :) |
| 01:31.41 | woleium | what sort and how much? |
| 01:33.35 | *** join/#asterisk usrb1n (~usrb1n@101.189.218.87.dynamic.jazztel.es) |
| 01:33.41 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 01:34.27 | usrb1n | Hello everybody, I have a question...kinda stupid I think but I can't find out how to do this |
| 01:34.36 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:34.36 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:34.49 | usrb1n | I saw more than 50 pages, and all of them had a unique way to do this. |
| 01:35.21 | usrb1n | I am start with asterisk (started 2 days ago) I already setup few extensions and tested IP calls |
| 01:35.39 | usrb1n | Works fine, I'm so happy and excited to see that |
| 01:35.40 | usrb1n | but.. |
| 01:35.58 | usrb1n | I have 2 paid sip accounts and I would like to make something like this.. |
| 01:36.40 | usrb1n | To set the asterisk so when I make a call will use some of my sip accounts to make the call |
| 01:36.51 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 01:37.15 | usrb1n | So I login to some asterisk's account and call +1123456 |
| 01:38.06 | usrb1n | That should go to my external sip and call with it, I know it's possible and maybe very very easy but I can't find out how to do this. I always found info about some providers only, but not some "default" configuration for this |
| 01:40.43 | dijib | woleium: sangoma a200 w. 1 fxo board asking a reasonable offer |
| 01:41.26 | woleium | I'm in the market for a card, but i need at least 8 fxo's |
| 01:41.51 | woleium | ebay is your friend :-) |
| 01:42.22 | dijib | this card is the base card then you can get expansion boards.. right now this single card has 2 fxo ports. if you got another 3 fxo daughter board and an expansion that would do 8 fxo ports |
| 01:43.11 | dijib | like this http://www.ebay.ca/itm/Sangoma-A200-R-Analog-IP-gateway-Card-FXO-6-ports-Tested-Working-/350527570996?pt=LH_DefaultDomain_0&hash=item519d124434 |
| 01:43.55 | dijib | its expandable up to 24 ports per base card. with fxo/fxs mixing |
| 01:44.37 | dijib | im looking for any reasonable offer or trade... really looking for an ata or two or three |
| 01:50.17 | butchevans2001 | <PROTECTED> |
| 01:51.03 | *** join/#asterisk Gaiax (~Gaiax@unaffiliated/gaiax) |
| 01:51.32 | butchevans2001 | looks to me like either: 1.) the sccp documentation assumes some other config OR 2.) this feature (realtime) isn't working in either version I've tried OR 3.) I've simply missed some important step |
| 01:53.23 | butchevans2001 | Not sure if there is further documentation, but I'm using this: http://chan-sccp-b.sourceforge.net/doc/setup_sccp.html#sccp_config_rt |
| 02:38.22 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 02:40.44 | *** join/#asterisk odin_ (~Odin@93-97-168-38.zone5.bethere.co.uk) |
| 02:48.23 | *** join/#asterisk Montys (~dmartinez@nat/digium/x-foazhoprojheqkmn) |
| 02:50.51 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 02:51.29 | Montys | Hello Guys, I have been dealing with a problem and I have no idea how to fix it, perhaps you could give me a suggestion, I recently upgraded Astersik from Asterisk-1.4.23/zaptel-1.4.12 to Asterisk-1.4.43/DAHDI-2.6, everything compiles fine and Asterisk and DAHDI now run, the problem is that when I call in, Asteirsk does not see the call, but I'm able to dial out using a callfile |
| 02:51.32 | Montys | any ideas? |
| 02:52.25 | Montys | Initially, I thought that the problem could be located on Asterisk, - Maybe asteirsk was compiling against zaptel instead of DAHDI |
| 02:52.41 | Montys | so I removed all stull related to zaptel in the server |
| 02:52.51 | Montys | modules, firmwares, etc,etc |
| 02:52.55 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 02:52.55 | Montys | and recompiled |
| 02:53.01 | Montys | but the problem still the same |
| 02:53.44 | p3nguin | usrb1n: Did you read the book? |
| 02:53.47 | p3nguin | ~book |
| 02:53.47 | 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://www.asteriskdocs.org/ or see ~buybook |
| 02:53.55 | p3nguin | usrb1n: ^^^^^^^^^ |
| 02:54.05 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:54.05 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:54.14 | p3nguin | usrb1n: You just create an extension using a pattern for whatever you want to call. |
| 03:28.39 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 03:46.08 | *** part/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 03:47.59 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 03:53.04 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 03:55.39 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 04:03.47 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:13.28 | *** join/#asterisk linuxgeek (~uppal@uppal.linux.net.pk) |
| 04:41.12 | Kobaz | okay so |
| 04:41.31 | Kobaz | i have two sip peers defined, with the same host= |
| 04:42.20 | Kobaz | i get a call from A and it works fine... i get a call from B and it complains 2012-02-23 23:40:12] WARNING[2079]: chan_sip.c:13927 check_auth: username mismatch, have <A>, digest has <B> |
| 04:42.40 | Kobaz | so how do i get the sip channel driver to only match on username and not ip address |
| 04:42.53 | Kobaz | because i know by default it matches the first defined peer with the source ip |
| 04:46.01 | Kobaz | i've never really played with that |
| 04:46.05 | Kobaz | i think maybe if you define just a sip user |
| 04:52.54 | *** join/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 04:52.58 | Kobaz | asterisk is supposed to match against users first, but it doesn't always |
| 04:56.01 | *** join/#asterisk RickCogley (~anonymous@114.179.8.185) |
| 05:00.30 | p3nguin | type=peer matches on IP/port, never a username. |
| 05:01.51 | p3nguin | And I'm not sure what you mean by supposed to but doesn't always. It's only coded one way, so it works how it works. |
| 05:02.13 | Kobaz | well |
| 05:02.17 | Kobaz | this is in the docs |
| 05:02.49 | Kobaz | When naming devices, make sure you understand how Asterisk matches calls that come in. |
| 05:03.01 | Kobaz | <PROTECTED> |
| 05:03.14 | Kobaz | 2. Asterisk checks the From: addres and matches the list of devices with a type=peer |
| 05:03.21 | Kobaz | etc |
| 05:03.27 | Kobaz | so, if i define two users |
| 05:03.33 | Kobaz | say A and B |
| 05:03.38 | Kobaz | and two peers, say A and B |
| 05:04.02 | Kobaz | it appears asterisk will not match against the users first, it tries to match against the peers instead, based on ip address |
| 05:04.08 | Kobaz | this is not expected based on what's in the docs |
| 05:04.39 | Kobaz | and another thing to add, the peers, A and B, have the same host= |
| 05:05.25 | Kobaz | if you define only two users, A and B, then calls to A are matched against A and calls to B are matched against B |
| 05:05.44 | Kobaz | as soon as you define a peer, the user matching seems to get skipped, which is what i mean by "doesn't always" |
| 05:06.03 | Kobaz | this has always given me problems, i think there's a problem with ordering when doing the matching |
| 05:07.11 | Kobaz | i don't have conclusive testing this is the case, but it seems to be |
| 05:09.21 | p3nguin | Are you using any insecure settings in either the user or peer? |
| 05:10.08 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 05:10.30 | Kobaz | if i turn on insecure |
| 05:10.41 | Kobaz | then asterisk will pick the first peer that matches the ip |
| 05:10.45 | Kobaz | which isn't what i want |
| 05:11.02 | Kobaz | if i don't turn on insecure, then i get check_auth: username mismatch, have <A>, digest has <B> |
| 05:11.32 | Kobaz | which means asterisk is trying to authenticate against A when really it should be following the digest, and using B |
| 05:13.29 | p3nguin | This seems like something you should talk to qwell about. |
| 05:14.02 | Kobaz | i think it's a matter of the sip matching isn't really flexible as i would like it to b |
| 05:14.03 | Kobaz | e |
| 05:14.23 | p3nguin | It should be able to match just like you are trying to match. |
| 05:14.29 | Kobaz | yeah it should |
| 05:14.37 | Kobaz | but it would be nice to have a ruleset |
| 05:14.48 | Kobaz | this user, go here, this ip go here, this ip and user, go this other place |
| 05:14.57 | p3nguin | If the From: is A and there is a user defined with name A, it should match it. |
| 05:15.06 | Kobaz | like the way you set up a sip proxy |
| 05:15.12 | Kobaz | yeah it should |
| 05:15.15 | p3nguin | If the From: is B and there is a user defined with name A, it shouldn't match it... and it should try to find a match for B. |
| 05:15.57 | p3nguin | If there is no match for B, *THEN* complain about a host match with username mismatch. |
| 05:16.21 | Kobaz | yeah |
| 05:16.39 | Kobaz | it should exhaust all options before failing or trying to match by ip on the peers list |
| 05:19.27 | Kobaz | yeah, i'm reading it right |
| 05:20.03 | Kobaz | check_peer_ok() finds the peer and then sends it to check_auth |
| 05:20.39 | Kobaz | so if the peer that it thinks it matched, is different than the sip header, then it bails |
| 05:22.15 | Kobaz | i think i'll add some matching options to make my life easier |
| 05:23.01 | Kobaz | i remember talks of a whole new sip matching infrastructure |
| 05:23.10 | Kobaz | sounded pretty sexy |
| 05:27.50 | [TK]D-Fender | Kobaz: type=friend <---------- |
| 05:28.04 | [TK]D-Fender | And insecure=no |
| 05:28.39 | Kobaz | actually i'm studying the sip debugs a bit more |
| 05:29.46 | Kobaz | i think the problem is that type=user doesn't take into account match_auth_username |
| 05:30.20 | Kobaz | find_peer() is finding A because the sip invite is actually coming from A@ip |
| 05:30.27 | Kobaz | but the auth digest is B |
| 05:30.53 | Kobaz | it's my sip gateway that's not behaving, kind of |
| 05:31.18 | Kobaz | it's sending the right digest auth, which match_auth_username=yes should enforce the match against that |
| 05:31.19 | Kobaz | but it's not |
| 05:31.39 | Kobaz | what i want is to match against the digest auth no matter what the from address is |
| 05:31.42 | Kobaz | that's actually the issue |
| 05:31.59 | Kobaz | the user/peer matching is doing what it's supposed to (although ignoring match_auth_username) |
| 05:32.29 | Kobaz | fixing that would actually solve a lot of my problems in certain areas that i have stupid workarounds |
| 05:33.22 | p3nguin | I hate that I can't use two different user or peer entries for two different accounts from my ITSP because they never send a username. Ever. |
| 05:33.41 | p3nguin | All calls are from your-cid-number@the-ITSP-host. |
| 05:33.48 | Kobaz | ah |
| 05:34.12 | p3nguin | So unless I made a user entry for every single number which would ever call me, and only if each of the numbers can only call one of the accounts... |
| 05:34.22 | p3nguin | Well, it's just a bother. |
| 05:34.25 | Kobaz | heh |
| 05:34.48 | Kobaz | your itsp needs to behave and send a username |
| 05:34.50 | *** join/#asterisk linuxgeek (~uppal@uppal.linux.net.pk) |
| 05:34.50 | p3nguin | I have no way to use two inbound accounts from the ITSP. |
| 05:35.07 | p3nguin | I can use two separate outbound accounts with no problem. |
| 05:35.12 | Kobaz | yeah |
| 05:35.17 | Kobaz | so by your-cid-number |
| 05:35.20 | Kobaz | is it the dnid |
| 05:35.23 | Kobaz | or the callerid of the caller |
| 05:35.31 | p3nguin | caller id of the caller |
| 05:35.31 | *** join/#asterisk goathouse774_ (~goathouse@20.160.251.216.lan.static.cptelecom.net) |
| 05:35.38 | Kobaz | oh that's annoying |
| 05:35.47 | Kobaz | they should send that in remote party id or in p-asserted-identity |
| 05:35.49 | p3nguin | In other words, impossible to match. |
| 05:35.52 | linuxgeek | hi , i am trying to integrate with a soft switch , the problem is that the soft switch has a number of IPs ( 10 in total ) where the calls come from , do i have to define a seperate peer or can i somehow define a subnet / range? |
| 05:36.36 | Kobaz | linuxgeek: if your soft switch can send a username then you just need that |
| 05:36.54 | Kobaz | no subnet matching |
| 05:36.58 | p3nguin | Maybe I have missed something? I looked for any distinguishing factors, but I never found any. |
| 05:37.06 | Kobaz | p3nguin: see that's why a whole new matching infrastructure would be nice |
| 05:37.12 | Kobaz | match on subnets and etc |
| 05:37.13 | linuxgeek | Kobaz, nope , cant do that , it a R4 switch from nokia |
| 05:37.57 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-bjokvvxrfwpbmney) |
| 05:38.02 | goathouse774_ | I'm staring down the long labor ahead of me of setting up voip for my small business. How -- in light of my ripeness -- should I approach this cryptic task? |
| 05:38.27 | Kobaz | p3nguin: if you convince your itsp to send remote party id or p-asserted-identity instead of cid@ip, and then use username@ip instead... then you can match |
| 05:38.43 | kaldemar | ~book |
| 05:38.44 | 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://www.asteriskdocs.org/ or see ~buybook |
| 05:38.57 | goathouse774_ | I have that book. |
| 05:39.08 | Kobaz | then you're all set |
| 05:39.17 | jpsharp | goathouse774_: Determine what you need to support as far as users & features. |
| 05:39.30 | *** join/#asterisk bmg505 (~leon@196-209-123-82.dynamic.isadsl.co.za) |
| 05:39.40 | Kobaz | oh |
| 05:39.49 | Kobaz | p3nguin: match_auth_username is a global sip option |
| 05:40.07 | Kobaz | i'm trying to use it on a per-user or per-peer context |
| 05:40.28 | Kobaz | i should make some doc changes for leif |
| 05:40.30 | goathouse774_ | but... I don't want to... READ. |
| 05:40.39 | goathouse774_ | I want to talk |
| 05:40.54 | Kobaz | goathouse774_: hire an asterisk guru at $250 an hour then |
| 05:40.56 | kaldemar | goathouse774_: then you're in a world of pain. |
| 05:41.11 | Kobaz | most of us take paypal payments |
| 05:41.42 | goathouse774_ | well, I'm sure you all deserve to paid handsomely. |
| 05:41.48 | goathouse774_ | *be paid |
| 05:41.53 | Kobaz | goathouse774_: or spend a week, read through the book twice, and try some experiments... seriously, you'll learn most of what you need to know by the book + playing |
| 05:42.18 | Kobaz | and then by all means, stop by here and ask questions |
| 05:42.31 | Kobaz | but for someone here to educate you from scratch, well... be prepared to pay |
| 05:42.39 | linuxgeek | Kobaz, here's the thing ,i dont think that it even matches the one IP being mentioned in the HOST section , kblmss03 172.21.42.4 N 5060 OK (1 ms) |
| 05:42.39 | linuxgeek | <PROTECTED> |
| 05:42.54 | Kobaz | linuxgeek: paste some logs |
| 05:42.57 | Kobaz | pastebin |
| 05:42.58 | linuxgeek | sure |
| 05:43.01 | linuxgeek | ~pastebin |
| 05:43.01 | infobot | [~pastebin] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 05:43.25 | goathouse774_ | I never really doubted that I'd have to read that damn book anyway. |
| 05:43.35 | Kobaz | it's an excellent book |
| 05:44.03 | goathouse774_ | I know, I know |
| 05:44.15 | goathouse774_ | I'm just so LAZY |
| 05:44.22 | Kobaz | lazy = hire someone |
| 05:44.30 | Kobaz | industrious = do it yourself |
| 05:44.37 | Kobaz | it's how the world works |
| 05:44.46 | goathouse774_ | did you start from scratch? |
| 05:44.46 | [TK]D-Fender | Cheap + lazy = fail |
| 05:44.53 | Kobaz | [TK]D-Fender: haha, yes |
| 05:45.30 | Kobaz | of course we started from scratch |
| 05:45.34 | Kobaz | you have to start somewhere |
| 05:45.34 | linuxgeek | Kobaz, http://pastebin.com/Mbfbjjfe , even though i have the ip defined in the peer settings , asterisk still looks in default context in the dialplan |
| 05:45.40 | Kobaz | when you start you don't know anything, none of us do |
| 05:45.55 | Kobaz | goathouse774_: but yes, i got hired as an head of IT at a startup developing for asterisk 5 years ago, i knew nothing about asterisk |
| 05:46.06 | goathouse774_ | I mean, did you start without a college course of some relation under your belt? |
| 05:46.16 | Kobaz | i asked some questions to coworkers, i was handed the book and said "have fun" |
| 05:46.35 | [TK]D-Fender | What I started .... there was no book |
| 05:46.37 | Kobaz | goathouse774_: i have a college degree, but i was programming since I was 6 years old |
| 05:46.55 | [TK]D-Fender | Kobaz: same here.... |
| 05:47.01 | Kobaz | goathouse774_: the startup fizzled after two years, but I had learned a lot, now I run a company |
| 05:47.03 | goathouse774_ | you got hired as an IT head and THEN had to learn asterisk? |
| 05:47.11 | Kobaz | goathouse774_: yeap |
| 05:47.23 | Kobaz | i knew linux and programming and nothing about phone systems |
| 05:47.45 | Kobaz | they needed a sysadmin and architect, and i was it |
| 05:48.08 | goathouse774_ | hm; I don't have linux experience and I only know BASIC, and not much of it. |
| 05:48.22 | Kobaz | you'll need to start with linux then |
| 05:48.31 | [TK]D-Fender | goathouse774_: Sorry... you're overqualified. Go hire an underling |
| 05:48.37 | Kobaz | working with asterisk is going to be a mystery otherwise |
| 05:49.00 | goathouse774_ | linux command line, or linux gui? |
| 05:49.08 | Kobaz | command line of course |
| 05:49.17 | goathouse774_ | damn. :) |
| 05:49.20 | Kobaz | you won't get anywhere with just the gui, trying to set up asterisk |
| 05:49.34 | Kobaz | goathouse774_: what you're asking us... |
| 05:49.48 | jpsharp | Or you'll just be buying a Switchvox appliance box. |
| 05:49.51 | Kobaz | goathouse774_: is how to put together a car... but you've never used a screwdriver |
| 05:50.10 | goathouse774_ | I know! Isn't it cute!? |
| 05:50.39 | Kobaz | asterisk is a pile of parts. every part you need to build that car |
| 05:50.55 | Kobaz | but you need to glue and bolt them all together for anything to work |
| 05:51.29 | goathouse774_ | ok, I read you, but don't I need the 3rd edition of a book about Linux then? |
| 05:51.42 | Kobaz | asterisk is the body, the engine, the steering.. and it's your job to put in the trim and upholstery |
| 05:52.36 | Kobaz | goathouse774_: you really need to spend a good 2-3 months learning linux and the command line |
| 05:52.58 | Kobaz | and then dive head first into asterisk and really be a sponge and absorb the book |
| 05:53.22 | goathouse774_ | I need an accelerated path |
| 05:53.26 | Kobaz | $$ |
| 05:53.27 | linuxgeek | Kobaz, anyluck with the log i pasted? |
| 05:53.30 | goathouse774_ | I'm short on time and money. :) |
| 05:53.38 | Kobaz | linuxgeek: umm, which one |
| 05:53.48 | Kobaz | oh that |
| 05:53.50 | Kobaz | sec |
| 05:53.58 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 05:54.00 | p3nguin | There is no accelerated path other than paying someone else for their time to do it faster than you can learn it. |
| 05:54.13 | linuxgeek | even though i have the peer with the ip defined , the call still lands in default in the dialplan |
| 05:54.21 | goathouse774_ | I'm not asking for hand-holding or inordinate free help, but isn't there a good way to learn Linux command line quickly? |
| 05:54.38 | *** join/#asterisk dijib (~root@bas10-kitchener06-1176001599.dsl.bell.ca) |
| 05:54.38 | jpsharp | Set up a linux box, plow into it. |
| 05:54.47 | goathouse774_ | box is computer? |
| 05:54.48 | p3nguin | There could be a Linux For Dummies book, but there's no substitute for experience. |
| 05:54.58 | Kobaz | linuxgeek: there's gotta be more than that, asterisk doesn't just accept calls to default on its own |
| 05:54.59 | dijib | experience!!! |
| 05:55.16 | dijib | hows everyone |
| 05:55.19 | dijib | ? |
| 05:55.24 | jpsharp | Yep. Go get a cheap computer, put linux on it, then asterisk. |
| 05:55.29 | Kobaz | goathouse774_: box = some sort of computer equipment... server.. router... etc |
| 05:55.41 | Kobaz | but in this case, yes, box = computer/server |
| 05:55.51 | goathouse774_ | I have a brand-spanking new Linux computer that I'm using to talk with you now |
| 05:55.56 | Kobaz | good good |
| 05:55.58 | Kobaz | you're almost there |
| 05:56.02 | dijib | desktop? |
| 05:56.06 | dijib | or server |
| 05:56.07 | [TK]D-Fender | linuxgeek: Show us the actual call attempt with SIP DEBUG enabled |
| 05:56.08 | goathouse774_ | desktop |
| 05:56.17 | Kobaz | linuxgeek: that too |
| 05:56.19 | dijib | <PROTECTED> |
| 05:56.19 | goathouse774_ | with xubuntu |
| 05:56.20 | dijib | :D |
| 05:56.23 | linuxgeek | [TK]D-Fender, Kobaz , sure |
| 05:56.27 | goathouse774_ | what!? |
| 05:56.27 | dijib | xubuntu is lame |
| 05:56.35 | goathouse774_ | the garbage!? |
| 05:56.36 | Kobaz | heh |
| 05:56.40 | Kobaz | my dad uses xubuntu |
| 05:56.42 | Kobaz | he gets by |
| 05:56.44 | dijib | yes the garbage... so i can grab it |
| 05:56.45 | dijib | :D |
| 05:56.46 | p3nguin | What I see in that pastebin is a call that didn't match the peer entry shown and an unnecessary Answer() |
| 05:57.15 | goathouse774_ | I was proud of my shiny little linux refurb |
| 05:57.17 | dijib | so i think i have installed a url hook/. |
| 05:57.17 | dijib | eff |
| 05:57.26 | dijib | refurb? |
| 05:57.33 | [TK]D-Fender | linuxgeek: the IP matches so it clearly failed auth |
| 05:57.35 | Kobaz | p3nguin: actually in modern asterisk versions the Answer() is necessary if you want the audio path open and ready when you play your tracks |
| 05:57.35 | p3nguin | ~refurb |
| 05:57.39 | goathouse774_ | well, it's an older windows comp that I installed Linux on |
| 05:58.00 | dijib | i need more DDR2.. anybody? |
| 05:58.06 | dijib | tradeski? |
| 05:58.10 | Kobaz | p3nguin: Playback() and friends will Answer() for you eventually, and you'll miss a few seconds sometimes |
| 05:58.20 | p3nguin | kobaz: That would be a negative. Priority 2 is a Playback(). Playback() has an implicit Answer() unless you are using the noanswer option. |
| 05:58.41 | dijib | there has to be an automated way to correct dialplans |
| 05:58.45 | Kobaz | yes it has an implicit answer, but not all cases will you hear the entire audio |
| 05:58.47 | p3nguin | There's no need for an Answer immediately before a Playback. It does no good.` |
| 05:58.55 | linuxgeek | [TK]D-Fender, Kobaz http://pastebin.com/QdWhRydK |
| 05:58.55 | dijib | like parser/replacer |
| 05:59.02 | Kobaz | p3nguin: i always do Answer() Wait(2) |
| 05:59.03 | goathouse774_ | this computer couldn't handle anything more than xubuntu, honestly |
| 05:59.08 | p3nguin | If you used Answer(1000), then maybe it would matter. |
| 05:59.12 | goathouse774_ | in fact, I may downgrade to lubuntu |
| 05:59.33 | p3nguin | If you used Answer followed by a Wait, maybe that would also matter. |
| 05:59.35 | Kobaz | p3nguin: so many systems have the problem of not answering soon enough, and starting to play tracks too soon |
| 05:59.58 | p3nguin | Or if you started out with Playback(silence/1&...), that would be fine. |
| 05:59.59 | Kobaz | call 800foobar, and head the cut off tracks saying "..for calling xyz enterprises" |
| 06:00.31 | Kobaz | yeah actually lately i have been using playback(silence/x) some sip endpoints don't realize the audio is going during the Wait |
| 06:00.54 | Kobaz | when the track should have been "thank you for calling" |
| 06:00.58 | p3nguin | If I need time after an answer but before the next command, then I'll use an answer and some form of wait time, typically Answer(2000) if I need to answer and wait 2 seconds. |
| 06:00.58 | Kobaz | two whole words get cut off |
| 06:01.04 | Kobaz | if you don't answer and wait beforehand |
| 06:01.31 | Kobaz | but it depends on the system |
| 06:01.35 | Kobaz | and what's in between |
| 06:01.37 | p3nguin | But anyway, that was the secondary problem anyway. |
| 06:01.42 | Kobaz | yeah |
| 06:02.08 | p3nguin | The primary problem is that his call did not match his peer AND that he didn't define a good general context value. |
| 06:02.13 | Kobaz | goathouse774_: if you throw on a more server-centric distro, like say debian... you can use a 486 for all i care, it'll work fine |
| 06:02.27 | dijib | goathouse774_: your going the wrong direction |
| 06:02.35 | [TK]D-Fender | linuxgeek: <--- SIP read from UDP:172.21.42.32:5060 ---> <-------------------- |
| 06:02.40 | [TK]D-Fender | linuxgeek: Different IP |
| 06:03.07 | Kobaz | eye pee |
| 06:03.22 | dijib | goathouse774_: install CentOS with groupinstall "Gnome Desktop" |
| 06:03.26 | linuxgeek | btw - hello from afghanistan :-) |
| 06:03.28 | dijib | eyeyeyeyeye |
| 06:03.36 | dijib | hell afghanistan |
| 06:03.43 | dijib | american? |
| 06:03.55 | linuxgeek | yes |
| 06:04.02 | linuxgeek | so how do i define a complete subnet |
| 06:04.07 | p3nguin | Why would any server ever need gnome desktop? |
| 06:04.23 | linuxgeek | this is a switch from Nokia Siemens , the issue is i dont have a complete list of their IPs |
| 06:04.25 | p3nguin | You shouldn't give deliberate bad advice. |
| 06:04.58 | Kobaz | linuxgeek: you can define allow/deny by subnet |
| 06:05.14 | p3nguin | That isn't going to help on peer matching. |
| 06:05.15 | Kobaz | linuxgeek: but not a matching range |
| 06:05.18 | Kobaz | i know |
| 06:05.43 | goathouse774_ | ok |
| 06:05.51 | linuxgeek | Kobaz, exactly , so i create a peer for all IP address? - this thing has 42 signaling units every unit has 5 ips |
| 06:06.15 | Kobaz | oh sweet |
| 06:06.22 | p3nguin | It seems like you need to match on user name instead of host. |
| 06:06.23 | Kobaz | wow this is so cool |
| 06:06.41 | Kobaz | sip general config match_auth_username is the best thing ever |
| 06:06.54 | *** join/#asterisk samul (~user@58.60.63.195) |
| 06:07.00 | Kobaz | BEST THING EVAR!!! |
| 06:08.36 | Kobaz | linuxgeek: you need a [93700805114] type=user |
| 06:08.37 | Kobaz | i guess |
| 06:08.42 | Kobaz | or is that the callerid of the caller |
| 06:08.46 | Kobaz | (like p3nguin's problem) |
| 06:08.54 | Kobaz | this is why itsps need to send a goddamn username |
| 06:09.12 | Kobaz | and asterisk needs multi host matching for peers |
| 06:09.37 | p3nguin | That looks like a phone number to me, not a username. |
| 06:09.40 | Kobaz | there's nothing to match on in that sip invite |
| 06:09.43 | Kobaz | yeah it does |
| 06:09.43 | linuxgeek | that's the caller id |
| 06:09.48 | p3nguin | weeeeeeeee |
| 06:10.19 | Kobaz | linuxgeek: looks like you're adding peer entries for those 10 ip addresses |
| 06:10.35 | p3nguin | 42 units, each unit has 5 addresses |
| 06:10.35 | Kobaz | the only way to go |
| 06:10.39 | Kobaz | haha |
| 06:10.44 | Kobaz | seriously? |
| 06:10.46 | linuxgeek | lol , yes |
| 06:10.53 | linuxgeek | and that's for one R4 |
| 06:10.57 | linuxgeek | i have to connect to 5 |
| 06:10.59 | Kobaz | write a script to add them all |
| 06:11.13 | Kobaz | you can't turn on some sort of username in the header? |
| 06:11.20 | Kobaz | can't use digest auth? |
| 06:11.25 | p3nguin | There's no way I would do that. I'd figure out something else. |
| 06:11.43 | p3nguin | adding all those hosts, that is. |
| 06:11.48 | Kobaz | heh |
| 06:11.58 | p3nguin | No. Way. |
| 06:12.05 | linuxgeek | nope it requires a feature called NVS [ nokia voice service ] which they want to charge 1 million $ for , the operator wouldn't pay for that |
| 06:13.09 | butchevans2001 | http://pastebin.com/ZUhJH6Kx Still looking for help with asterisk/chan_sccp-b/mysql |
| 06:13.18 | [TK]D-Fender | LinuJust check the context in [general] and let it fall in un-authed and check the IP in the dialplan |
| 06:13.19 | butchevans2001 | Not sure if there is further documentation, but I'm using this: http://chan-sccp-b.sourceforge.net/doc/setup_sccp.html#sccp_config_rt |
| 06:13.35 | butchevans2001 | looks to me like either: 1.) the sccp documentation assumes some other config OR 2.) this feature (realtime) isn't working in either version I've tried OR 3.) I've simply missed some important step |
| 06:15.16 | Kobaz | [TK]D-Fender: heh, hah... yeah |
| 06:15.50 | p3nguin | That actually wouldn't be too hard. |
| 06:15.53 | Kobaz | linuxgeek: might as well have bought a broadsoft switch |
| 06:16.39 | linuxgeek | :) |
| 06:16.44 | linuxgeek | not my decision |
| 06:17.40 | Kobaz | if you're gonna spend a million bucks on a soft switch, might as well get a decent one |
| 06:18.09 | p3nguin | For the low low price of only .5 million dollars, I'll set up all 210 of the host entries for you. |
| 06:18.16 | Kobaz | heh |
| 06:18.25 | p3nguin | tonight! |
| 06:19.11 | linuxgeek | lol |
| 06:19.17 | p3nguin | You think PayPal will process that payment for me? |
| 06:19.23 | Kobaz | butchevans2001: i played with sccp-b for about two days, never did anything with mysql |
| 06:19.23 | linuxgeek | :) this is a R4 soft switch for the gsm network |
| 06:19.39 | linuxgeek | hahaha , p3nguin , i can APO cash ;) |
| 06:19.48 | linuxgeek | freshly printed US$ from afghanistan |
| 06:19.52 | p3nguin | I currently use chan_sccp, but I don't use realtime. |
| 06:19.55 | butchevans2001 | Kobaz: Meaning you never did anything with it, or the software never did |
| 06:20.13 | Kobaz | butchevans2001: i never configured it to use mysql |
| 06:20.13 | butchevans2001 | ahh |
| 06:20.15 | Kobaz | i just got some phones calling each other and crashing asterisk |
| 06:20.17 | Kobaz | that was about it |
| 06:20.30 | p3nguin | It's pretty good at doing that from time to time. |
| 06:20.35 | Kobaz | yeap |
| 06:20.43 | Kobaz | skinny and sccp and sccp-b scare me |
| 06:20.59 | butchevans2001 | sccp isn't my first choice...I'm doing a free favor for a friend and he insists on sccp because of ONE stupid cisco failure with sip |
| 06:21.10 | Kobaz | with skiiny you can't reload any configs without pretty much restarting asterisk |
| 06:21.18 | Kobaz | the old sccp-b is the same |
| 06:21.20 | butchevans2001 | failure is a strong word...lack of feature. |
| 06:21.22 | [TK]D-Fender | And then you compound it by insisting on realtime.... |
| 06:21.22 | p3nguin | I do it all the time... |
| 06:21.24 | Kobaz | and it doesn't really work |
| 06:21.34 | Kobaz | and then the new sccp-b has all kinds of nice features |
| 06:21.36 | p3nguin | but after a while, asterisk always seems to crash. |
| 06:21.38 | butchevans2001 | sccp-b still requires restarting sccp, which kills all phones with sccp |
| 06:21.39 | Kobaz | but then you put someone on hold and crash |
| 06:21.40 | [TK]D-Fender | Throwing bad money after worse.... |
| 06:21.49 | Kobaz | the new new sccp-b you can hot reload |
| 06:21.53 | p3nguin | I just do module unload chan_sccp, then module load chan_sccp. |
| 06:22.02 | Kobaz | yeah, which will nuke all calls |
| 06:22.14 | Kobaz | or not let you unload, i forget which |
| 06:22.18 | p3nguin | It's better than taking down all the calls using other techs, too. |
| 06:22.27 | Kobaz | yeah |
| 06:22.49 | Kobaz | cisco phones flashed with sip aren't much good either |
| 06:23.03 | Kobaz | you get a dumb endpoint, all the cisco phone features are gone |
| 06:23.04 | butchevans2001 | given that I'm stuck with sccp, is there a better choice than sccp-b? |
| 06:23.10 | Kobaz | no, sccp-b |
| 06:23.19 | Kobaz | that's your option |
| 06:23.23 | Kobaz | all hail the crashing |
| 06:24.24 | [TK]D-Fender | Checkout time, later all |
| 06:24.29 | Kobaz | people are like "i heard asterisk can support cisco.. let's deploy it for 100 phones" |
| 06:24.34 | butchevans2001 | hrrm...so...I've looked at realtime configuration documentation and I have all of that configured. Is there anything else I can look at to see why it isn't even trying mysql? |
| 06:24.42 | Kobaz | pukes |
| 06:25.18 | Kobaz | butchevans2001: gdb |
| 06:25.25 | butchevans2001 | I've offered to send him about 1/4 of the phones he needs (Polycom) if he'll buy the remainder |
| 06:25.40 | butchevans2001 | gdb? please elaborate a bit |
| 06:25.49 | Kobaz | the gnu debugger |
| 06:26.15 | butchevans2001 | that may be a bit over my head to try. I'm not even sure where to start with that |
| 06:26.19 | Kobaz | ie: step through the code and fix it yourself... is probably the best option |
| 06:26.48 | butchevans2001 | yeah...that's not likely to happen. I'm not a dev by any stretch of the imagination |
| 06:27.10 | Kobaz | do you need to use mysql? |
| 06:27.21 | Kobaz | can you export your stuff to config files upon change |
| 06:27.27 | Kobaz | i think mysql supports triggers finally |
| 06:27.42 | butchevans2001 | I'm thinking that is my option |
| 06:28.10 | butchevans2001 | but I would need to figure out how to cause perl or some other language to restart/reload the sccp module in radius |
| 06:28.15 | butchevans2001 | errr..asterisk |
| 06:28.54 | butchevans2001 | I may step through the php code in freepbx and figure out how to do it from there |
| 06:29.17 | butchevans2001 | as I said, I'm not nearly a strong enough coder to do very much in that regard |
| 06:29.49 | butchevans2001 | how about this approach |
| 06:30.12 | butchevans2001 | 1. php page to allow configuration/management of phones |
| 06:30.27 | butchevans2001 | 2. write the above configs to the sccp.conf (or whatever file) |
| 06:30.40 | butchevans2001 | 3. on change, tell asterisk to reload the sccp module |
| 06:30.48 | butchevans2001 | I can do 1 and 2 easily enough |
| 06:30.54 | butchevans2001 | it's 3 that I don't know how to do |
| 06:31.17 | butchevans2001 | having my friend muck around in the config file would be a disaster |
| 06:32.35 | kaldemar | butchevans2001: CLI commands via AMI or "asterisk -rx ..." as a system command. |
| 06:34.20 | butchevans2001 | kaldemar: can you show me the documentation for the CLI (AMI)? I'd be happy to read that. I don't need any handholding here. This is why I offered to help for free, because it is new to me and I love to learn |
| 06:35.15 | butchevans2001 | of course, it was SIP that I thought I was doing, which was at least a LITTLE familiar to me |
| 06:36.14 | Kobaz | yeah |
| 06:36.52 | kaldemar | CLI as in the command line interface which to attach to with "asterisk -r" from shell. those commands (module unload, module load) can be issued through AMI aswell with "Command" (manager show command Command). |
| 06:37.34 | kaldemar | for AMI, start with this: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-AMI.html |
| 06:39.30 | Kobaz | i should get to bed and read my book on whale hunting |
| 06:41.12 | butchevans2001 | hmm. Looks like there is no SCCPReload... |
| 06:41.43 | butchevans2001 | lots of SCCPShow stuff and SCCPList stuff, but no way to reload sccp, which is what I need |
| 06:41.45 | butchevans2001 | sigh |
| 06:41.49 | Kobaz | yeah |
| 06:41.53 | Kobaz | that's the problem with sccp |
| 06:42.01 | Kobaz | are you on the svn trunk? |
| 06:42.07 | Kobaz | or the one that builds for 1.4 |
| 06:42.09 | butchevans2001 | yes...v4 |
| 06:42.19 | Kobaz | hmm i kinda remember a reload in the new new version |
| 06:42.35 | butchevans2001 | there is, but not showing up via AMI |
| 06:42.53 | butchevans2001 | asterisk CLI would be "sccp reload" |
| 06:43.28 | butchevans2001 | WAIT. I see what kaldemar meant |
| 06:43.53 | butchevans2001 | that may work for me |
| 06:44.08 | butchevans2001 | If I can find just one example of how to do it, I should be golden! |
| 06:44.57 | butchevans2001 | You have given me (I think) everything I need to at least take a shot at this. THANKS! |
| 06:45.40 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 06:45.42 | Kobaz | well |
| 06:45.51 | Kobaz | from the ami you can run every single asterisk cli command |
| 06:46.02 | butchevans2001 | yup. that's what I just saw |
| 06:46.44 | butchevans2001 | I was looking at the SCCP* commands...I completely missed/overlooked the "Command" command. Even though that is exactly what kaldemar said. |
| 06:46.47 | Kobaz | Command |
| 06:46.50 | Kobaz | from ami |
| 06:46.54 | Kobaz | yeah |
| 06:46.57 | *** join/#asterisk irieKEN (~ken@cpe-72-130-175-87.san.res.rr.com) |
| 06:48.09 | butchevans2001 | I am pretty certain that I have a good way to get out of this mess. I really was in WAY over my head once he told me sccp was his only option. Thanks, again, guys! |
| 06:51.02 | irieKEN | Can anyone direct me to info on how to register a toll-free number? Google isn't getting me to what I need. |
| 06:51.20 | ChannelZ | Register? As in just get one? |
| 06:51.57 | irieKEN | Kind of. I'm trying to figure out how I can find the one I want, how I'm supposed to track down the respOrg, etc. |
| 06:52.00 | butchevans2001 | ChannelZ: Personally, I use Flowroute for my upstream ITSP |
| 06:52.04 | ChannelZ | Most ITSPs usually provide toll-free DIDs, you just buy through them. |
| 06:52.25 | irieKEN | ChannelZ: But don't they own the number, and not you? |
| 06:52.50 | butchevans2001 | sorry, ChannelZ, I meant irieKEN |
| 06:53.17 | ChannelZ | Well you don't own it in any more of a way than you own your home phone number or whatnot... but it works the same way |
| 06:53.29 | butchevans2001 | You can port the number, irieKEN. That way, if you move to a different ITSP |
| 06:53.48 | butchevans2001 | you can keep the number |
| 06:54.08 | irieKEN | butchevans2001: Flowroute is one of my ITSPs, but they don't have so many toll-free numbers available. |
| 06:54.32 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 06:54.48 | butchevans2001 | they can get you any one you want, if it is available. |
| 06:54.53 | butchevans2001 | or ALMOST any one you want |
| 06:55.20 | butchevans2001 | there is a fee for that if it isn't in their current inventory |
| 06:55.31 | irieKEN | butchevans2001: Ok... |
| 06:59.55 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 07:01.07 | linuxgeek | <PROTECTED> |
| 07:01.30 | butchevans2001 | irieKEN: I just looked through my email from when I asked FR about a specific TF DID. It was $50 + their regular activation fee for a TF number |
| 07:01.39 | butchevans2001 | That was March 2011 |
| 07:04.05 | Kobaz | heh breakfast |
| 07:04.09 | Kobaz | time for sleep |
| 07:04.18 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 07:04.23 | v0lZy | hello |
| 07:05.17 | Kobaz | sleep |
| 07:06.07 | *** join/#asterisk hrolf (7514166e@gateway/web/freenode/ip.117.20.22.110) |
| 07:07.06 | hrolf | Asterisk doesn't execute the next statement after the first on transfer from SIP. The dialplan is Set() and then AGI() the Set executes but AGI doesn't and it drops the call |
| 07:07.43 | ChannelZ | pastebin the verbose console output of the call |
| 07:08.14 | ChannelZ | Probably your priorities are messed up |
| 07:08.22 | ChannelZ | check that first |
| 07:08.55 | *** join/#asterisk jofry (~procube@port-212-202-54-41.dynamic.qsc.de) |
| 07:10.19 | irieKEN | butchevans2001: thanks. I think that I may have done that for our last TF number. |
| 07:11.01 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 07:19.21 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 07:23.24 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 07:29.39 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 07:32.32 | hrolf | ChannelZ: I increased the output and find something which says "Didn't get a frame from SIP/100-0000000b" and after wards it says "Bride stops channel SIP/.... and SIP/...." |
| 07:32.38 | hrolf | ChannelZ: and hangups |
| 07:32.48 | hrolf | ChannelZ: Shall I paste the log? |
| 07:40.13 | v0lZy | Hey guys, one quick VoIP/SIP RTP question |
| 07:41.52 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 07:42.59 | hrolf | ChannelZ: http://pastie.org/3443251 |
| 07:43.40 | *** join/#asterisk jzaw (~jzaw@macbook.dzki.co.uk) |
| 07:45.36 | hrolf | I'm having issue that asterisk drops call |
| 07:45.58 | v0lZy | Am I correct in presuming, that when I place a call, first, SIP "rings" the other side and negotiates the codecs and the ports for the call. The process of this negotiation depends on the side that is being called. the receiving end lets the side that initiates the call know which ports to use. |
| 07:46.10 | v0lZy | ? |
| 07:46.16 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:46.21 | hrolf | In console, the error shows that "Didn't get a frame from SIP/100-000000b" (which is the caller) |
| 07:46.25 | hrolf | http://pastie.org/3443251 |
| 07:47.10 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:47.12 | WIMPy | Moin v0lZy: Each side tells the other where it listens for RTP. |
| 07:47.23 | v0lZy | hey WIMPy |
| 07:47.31 | v0lZy | excellent |
| 07:47.33 | v0lZy | so |
| 07:47.44 | WIMPy | hrolf: That looks rather like the call (audio) not being set up in the first place. |
| 07:47.46 | v0lZy | If I want 10 concurrent calls established |
| 07:48.00 | v0lZy | i need 20 ports for RTP, right? (each call taking 2 ports?) |
| 07:48.17 | WIMPy | Usually, yes. |
| 07:48.52 | v0lZy | but do i really need 20 open on the firewall? there will at any time only be 10 incoming calls. |
| 07:48.55 | *** join/#asterisk LemensTS (~matthew@70.238.150.222) |
| 07:49.03 | LemensTS | anyone awake? |
| 07:49.42 | v0lZy | my firewall allows outgoing connections normally... so i can connect to whichever port they provide me with |
| 07:50.45 | hrolf | WIMPy: What does it mean not setup? |
| 07:51.07 | hrolf | WIMPy: I can perfectly listen to the queue hold music at the caller side. |
| 07:51.46 | WIMPy | v0lZy: You culd be ok with 10, it might become even more than 20 depending on configuration (e.g. video). |
| 07:51.58 | WIMPy | hrolf: And can you speak? |
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| 08:02.09 | hrolf | WIMPy: Yes. Just tested it. |
| 08:03.25 | hrolf | WIMPy: Both parties can hear each other perfectly |
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| 08:06.46 | ChannelZ | hrolf: are you on to a different problem now? I thought it was your dialplan wasn't executing something |
| 08:07.16 | hrolf | ChannelZ: No, it seems that it isn't executing because the call is getting dropped |
| 08:08.11 | hrolf | ChannelZ: and the call is getting dropped because of the error "Didn't get a frame from SIP......" and after wards it shows another message "Bride stops bridging channel SIP.... and SIP ... " |
| 08:08.20 | hrolf | which are my caller and receiver SIP channels |
| 08:09.07 | ChannelZ | the debug is a mess to look at. Turn that off and do a 'core set verbose 3' and show things that way so we can see what the dialplan is even doing |
| 08:09.11 | hrolf | The scenario is that, call is received by AGI, it is put in queue, agent receives the call and then transfers to an extension. So in dialplan I'm again transferring to AGI but that part doesn't execute. |
| 08:10.30 | hrolf | I do blind transfer and it puts the message "soft hanging up channel" and then it says "didn't get a frame" is it because of the soft hanging up? |
| 08:15.32 | ChannelZ | re: the debug output says some things that look alarming but probably aren't, and leaves out a lot of things that make it easy to see what is going on |
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| 08:17.51 | mirelab | hello, has anyone used HASH function? |
| 08:18.10 | mirelab | I can't seem to irretate through it |
| 08:18.36 | mirelab | i always get first set of values |
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| 08:22.22 | *** join/#asterisk bitwize (~bitwize@h87-96-213-2.dynamic.se.alltele.net) |
| 08:27.51 | bitwize | I have a question: I have a system where agents connects to a conference by dialing an inbound number and authenticates with DTMF. Some of these agents do not hear anything after a little while, normally theese agets are located on the same network. |
| 08:28.50 | bitwize | Anybody with experience about this problem? Everything goes well but after a while some agents dont hear anything.. |
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| 08:46.01 | kaldemar | bitwize: sounds like a router that drops connections. are the problematic agents behind a NAT? |
| 08:48.35 | bitwize | kaldemar: OK, in some cases they are behind a NAT and in some cases they call in with a mobile phone. |
| 08:49.17 | bitwize | But I only have reports about this problem from customers behid "home"-based netowks so I also thought this is related to bad hardware |
| 08:50.23 | kaldemar | then that is one considerable option, yes. |
| 08:50.46 | bitwize | kaldemar: Do you have any tips regarding configuration of smaller networks with routers like d-link etc.? |
| 08:51.01 | bitwize | Protocol prio etc.? |
| 08:54.04 | bitwize | When the problem occures the call is still established but no audio beeing received, is it possible for the call to stay established when the router dropps the connection? |
| 08:55.00 | bitwize | (Everybody who experience the problem are using X-Lite) |
| 08:56.23 | kaldemar | sure, the SIP session can stay up even if there is no RTP end-to-end. |
| 08:57.10 | kaldemar | if the router really drops connections, i don't know if there's much you can do configuration wise. |
| 08:57.33 | bitwize | ok, is there anyway for me to detect this in Asterisk and re-establish the RTP? |
| 09:02.52 | *** join/#asterisk jofry (~procube@port-212-202-54-41.dynamic.qsc.de) |
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| 09:13.01 | mirelab | does anyone know how to get the next hash value using HASH function? |
| 09:13.54 | mirelab | i seem to get only first set of values that odbc function returns, and in mysql it returns 3 rows |
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| 09:25.14 | puR0 | hello all |
| 09:27.21 | mirelab | I need help with HASH function :) |
| 09:29.22 | Faustov | what could be the reason for keypresses being completely ignored without any error in the CLI? I've got dtmf set to auto, sip debug for a test call here: http://pastebin.com/WMa3Lzce |
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| 09:33.23 | *** join/#asterisk enoch (~enoch@unaffiliated/enoch) |
| 09:33.26 | enoch | hi all |
| 09:33.42 | enoch | i need a good solution for fax-to-mail and mail-to-fax |
| 09:34.01 | enoch | any suggestion? |
| 09:35.55 | puR0 | hi enoch, you should take a look to this: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#asterisk-Fax |
| 09:37.01 | puR0 | but, depending on what you need, you can find thousand recipes on the internet |
| 09:37.57 | enoch | http://ictfax.org/ something like that? |
| 09:38.05 | enoch | it seems to be a good project |
| 09:39.48 | enoch | but to install it i have to override a lot of freepbx generated confs |
| 09:40.24 | enoch | overwrite* |
| 09:41.32 | puR0 | well this is far away from the simple solution I provided |
| 09:42.09 | puR0 | btw thank you for the link, I'm quite new here and I've never heard of this project |
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| 09:44.14 | enoch | i don't want to break my pbx :P |
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| 10:03.56 | atekinalp | hi how can i see active calls uniqueid |
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| 10:07.45 | v0lZy | huh |
| 10:07.48 | v0lZy | i cant call out |
| 10:07.48 | *** join/#asterisk Maliuta_ (~nobusines@kiev.lusan.id.au) |
| 10:07.55 | v0lZy | whatever i dial i get circuit busy |
| 10:07.57 | v0lZy | any ideas? |
| 10:08.50 | stephendowed | hei people ssup......... |
| 10:15.01 | v0lZy | Hey guys, i have a problem dialing out |
| 10:15.21 | v0lZy | my asterisk first says auntetication failed and then something about congestion and circuit busy |
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| 10:22.07 | atekinalp | any suggestion? |
| 10:24.59 | hrolf | ChannelZ: Here is the log without debug (with verbose set to 3) |
| 10:25.02 | hrolf | ChannelZ: http://pastie.org/3444118 |
| 10:26.15 | hrolf | ChannelZ: You can see that when we execute -- Executing [A2I-1002-testtreatment_Welcome@call_manager:1] line, the next line in dialplan doesn't execute because the call has been dropped. |
| 10:27.10 | hrolf | ChannelZ: And if I set debug to on, than it will show the same message I previously mentioned "Didn't get a frame from SIP/100 .... " |
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| 11:04.54 | v0lZy | Helpz plz! |
| 11:06.43 | mirelab | Does anyone know how to reference to a specific row in a HASH function, cause I can only see first row that sql query returned? |
| 11:08.28 | davidduffy | what is your dialplan? |
| 11:08.52 | mirelab | Set(HASH(Polasci)=${ODBC_MetodName(${Day}, ${From}, ${To})}); |
| 11:09.16 | mirelab | I call odbc function that returns 3 rows from MySQL base |
| 11:10.08 | mirelab | and in CLI i see : HASH(Polasci)=1,1,2,Lasta <- which is the first row |
| 11:10.17 | *** join/#asterisk mattwj2002 (~matt@wikisource/pdpc.active.mattwj2002) |
| 11:10.22 | mattwj2002 | hi guys |
| 11:10.24 | mattwj2002 | :) |
| 11:10.33 | mattwj2002 | how is everyone this morning? |
| 11:10.46 | mirelab | good thx :) |
| 11:11.05 | *** join/#asterisk swente (~swente@unaffiliated/swente) |
| 11:11.14 | mattwj2002 | hey I see digium is getting into the phone business (I mean desk phone) |
| 11:11.28 | mattwj2002 | what do you guys think? |
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| 11:20.27 | jacc0 | hi all! |
| 11:20.36 | mattwj2002 | hey jacc0 |
| 11:20.36 | mattwj2002 | :D |
| 11:20.52 | swente | hello |
| 11:24.31 | *** part/#asterisk mattwj2002 (~matt@wikisource/pdpc.active.mattwj2002) |
| 11:28.52 | snadge | i think its a good idea |
| 11:29.12 | snadge | i wouldn't mind an ip phone myself.. and a digium one sounds like it would work nicely with asterisk |
| 11:29.22 | snadge | and probably standards compliant as well |
| 11:29.45 | swente | phew. the entire voip/asterisk/isdn/etc. - topic seems to be pretty wide. it is probably pointless to ask for recommendations for e.g. isdn-hardware? :) |
| 11:31.55 | jacc0 | I like the Siemens Gigaset S450IP; it supports sip and good old analoge lines simulaniously\ |
| 11:32.47 | swente | we have 2x S0 point-to-multipoint here. externally, at least for now isdn should be used (no external voip -- is hw error cancellation "required" in that case?). looking at digium te420bf and junghanns quadBRI currently. can someone comment those cards? |
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| 12:44.04 | kchehab | i enable the recording to my extension and have no problem with that but the problem is that after a while the (.wav)files been deleted from /var/spool/asterisk/monitor/ in other hand the records still in the mysql and sure when you click download it will reply by no files found ,how to fix it ? |
| 12:44.21 | kchehab | btw i am using elastix |
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| 13:40.05 | florianh | Hi, I've got a question about conferences... |
| 13:40.46 | florianh | Maybe anybody can help. I need a very special conference feature and don't know how to achieve this feature. |
| 13:41.40 | jacc0 | ~ask |
| 13:41.40 | 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. |
| 13:42.54 | florianh | Specially i need a conference, where the master user can hear all other users, but they can't hear each other. And if the master want's to speak to somebody, he will only speak to the specified user. |
| 13:43.51 | florianh | Thats not really a conference anymore i think... |
| 13:47.14 | akrohn | that sounds like a cool application. i don't really know, but it sounds like there will not be a trivial answer. i don't see why it wouldn't be possible but will take a fair amount of fiddling |
| 13:48.24 | florianh | Its an intercom for communication for example for ships |
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| 13:52.26 | florianh | Any hints for me how this could be done? I've already adapted the alsa channel driver to my needs so i'm a little bit experiences with asterisk programming. |
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| 13:55.54 | florianh | I've thought about writing a special conference app, but when looking at the existing conference application code it seems to be harder than i thought. |
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| 14:07.58 | leifmadsen | florianh: well you can certainly use app_meetme (or app_confbridge in asterisk 10) to do part of that. You have have a single speaker and multiple listeners who can't talk to each other. |
| 14:08.28 | leifmadsen | florianh: what I'm thinking on top of that, is you could use chanspy() somehow to "whisper" to the participant of the conference.... but that seems like it might not be as smooth as you want |
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| 14:10.02 | leifmadsen | florianh: what I'm thinking on top of that, is you could use chanspy() somehow to "whisper" to the participant of the conference.... but that seems like it might not be as smooth as you want |
| 14:10.08 | leifmadsen | oops, dupe :) |
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| 14:11.10 | florianh | leifmadsen: Yes, MeetMe sounds like fitting good... But the "other" users should not hear each other. Only the "master" should hear them. I see a problem in reaction time when using chanspy. In the application you have Buttons in a GUI. When you want to talk to a user in the conference you press the button... Creating a ChanSpy on fly via AMI interface might be too slow |
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| 14:12.49 | leifmadsen | florianh: so basically one person can hear everyone and can talk to everyone, but everyone else should only be able to hear the master |
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| 14:15.20 | florianh | leifmadsen: No sorry, its hard to explain in english for me :-) The "master" can hear all users. The "users" can't hear each other, also not the master. But the Master ( The intercom ) can decide by a button press to talk to one of the users |
| 14:17.09 | florianh | leifmadsen: In this application example the "users" will be often radios which an operator only listen to. Often they have many radios which they listen to. And when something happens on a radio the operator speaks to this special radio. |
| 14:17.10 | autofsckk | florianh: like in a classroom maybe? |
| 14:19.24 | florianh | autofsckk: Hmm, I can't follow you. You mean the teacher speaks to all? But thats not the same here... |
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| 14:20.26 | leifmadsen | florianh: ah gotcha, so master hears everyone, users hear everyone, but master only talks to a specific person |
| 14:21.00 | WIMPy | swente: That's the wrong card. S0 is BRI, not PRI. |
| 14:22.20 | florianh | leifmadsen: Nearly... The "users" in the conference should NOT hear each other. Only one User should hear the master IFF he whish to speak to him (PTT button pressed for user xy) |
| 14:22.22 | Nivex | I'm using the Debian packages from https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages but I'm not seeing current updates. |
| 14:22.53 | swente | WIMPy: more likely i mixed up wording, sorry. but, i actually wonder if i wouldn't have an easier life with an external media gateway [patton smartnode]? |
| 14:24.21 | WIMPy | swente: Easier maybe, but the conversion is very lossy and may have serious side effects. |
| 14:24.31 | swente | WIMPy: (wordings -- it is two ordinary isdn-lines, not multiplexed, but sharing phone-numbers) |
| 14:24.43 | swente | WIMPy: hmmm |
| 14:24.45 | WIMPy | So what kind of line is it you really have (or want)? |
| 14:24.57 | WIMPy | So then my comment was true. |
| 14:25.11 | WIMPy | Wrong type. |
| 14:25.13 | swente | WIMPy: we have an "anlagenanschluss" ;-) |
| 14:25.28 | WIMPy | They come in both versions. |
| 14:26.01 | swente | as in "ordinary and multiplexed"? (mpx having up to iirc 40 lines?) |
| 14:26.11 | WIMPy | And with that term you may like to join #asterisk-de |
| 14:26.35 | swente | WIMPy: danke :) |
| 14:26.38 | WIMPy | BRI=2B+d PRI 30B+D |
| 14:26.47 | *** part/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 14:26.56 | swente | ok, cu in "-de" |
| 14:30.46 | hackeron | hey, I'm trying to accept incoming calls from a sip provider, I put 'exten => 4348739,1,Answer()' in my default section in extensions.conf - but I'm still seeing: [Feb 24 14:31:14] NOTICE[7000] chan_sip.c: Call from '4348739' to extension '4348739' rejected because extension not found. -- any ideas? |
| 14:31.47 | florianh | leifmadsen: did you get me? Any ideas how i can get this functionality? |
| 14:31.52 | WIMPy | Using the default context is not a goood idea. |
| 14:34.09 | leifmadsen | florianh: sorry, not sure... I got distracted by work... I understand the concept. I think you're going to have to have to try and hack something together unfortunately.... conference is where I'd start and controlling the talkers and listeners, but I think you'll need to keep them in separate conferences and control them perhaps through AMI |
| 14:34.45 | hackeron | WIMPy: oh, never mind, figured it out, I had to set context to default in my sip.conf too - I'll create separate context for everything now, thanks :) |
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| 14:51.28 | WIMPy | ~versions |
| 14:51.47 | WIMPy | ~lts |
| 14:51.48 | infobot | somebody said lts was Leaning Toothpick Syndrome. Long Term Support: a version that is supported for a longer time period. |
| 14:52.04 | WIMPy | Hmm. Where is the Link hidden? |
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| 15:10.36 | pigpen | A quick question for those in the "know": Asterisk is 64bit right? (answer: yes, if compiled so) Then why are their codec and fax license tools 32bit? |
| 15:10.59 | pigpen | Hardened systems do not allow 32bit calls. |
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| 15:12.29 | pigpen | specifically referring to their "register" tool for the res_fax_digium |
| 15:12.34 | *** part/#asterisk mirelab (~mirko@212.200.146.253) |
| 15:13.17 | pigpen | if anyone from digium is listening?see case 00269924 |
| 15:14.18 | coppice | use a free FAX option and don't worry |
| 15:16.28 | pigpen | heh?still have to install the free license. |
| 15:16.34 | pigpen | with the same tool. |
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| 15:17.10 | mentax | I sometimes have loooong delay before the first ring on outbound calls… Is anyone know how to fix it? Or debug it? |
| 15:17.17 | coppice | don't use digium FAX. use something genuinely free |
| 15:17.52 | pigpen | coppice, yeah, we did too for some time?but iaxmodem and hylafax are starting to be problematic. |
| 15:17.59 | pigpen | and not developed. |
| 15:18.02 | [TK]D-Fender | mentax, go look at the call at * CLI |
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| 15:19.09 | coppice | pgpen: use spandsp. its a direct substitute for the digium FAX, with better files handling (not my words, but Digium's :-) ) |
| 15:19.29 | coppice | pigpen: what makes you think iaxmodem + hylafax are no longer developed? |
| 15:19.58 | pigpen | coppice, sorry, hylafax mostly. |
| 15:20.04 | pigpen | they haven't had an update is some time. |
| 15:20.34 | coppice | yes they have. both hylafax and hylafax+ have regular updates, although I'd go with hylafax+ |
| 15:20.34 | pigpen | in fact, with fax quality issues, the commercial support side of them is recommending customers to use digium's res_fax |
| 15:21.06 | coppice | when digium's fax doesn't work the "support" seems to be "try spandsp" |
| 15:23.09 | mentax | [TK]D-Fender: http://pastebin.com/6N8s3iZr |
| 15:23.43 | [TK]D-Fender | mentax, So at what point in all of this does the dealy occur? |
| 15:24.00 | mentax | [TK]D-Fender: I don't see anything wrong =( |
| 15:24.26 | [TK]D-Fender | mentax, Never said you would see anything specific. I asked at what point in this flow does the delay occur.... |
| 15:24.34 | mentax | [TK]D-Fender: delay from 20 sec to 1 min |
| 15:24.56 | [TK]D-Fender | mentax, ---> WHEN <--- |
| 15:25.33 | mentax | [TK]D-Fender: when I dial number |
| 15:26.05 | mentax | [TK]D-Fender: it should call immediately, but it wait for 20 - 60 sec, then start call |
| 15:26.15 | [TK]D-Fender | mentax, Last time. Please try to follow along. You rpvided debug. and what point IN THAT DEBUG does the delay occur? |
| 15:26.27 | [TK]D-Fender | provided* |
| 15:36.04 | akrohn | i need to record outbound calls. our outbound context does some includes which pattern match and ship off to outgoing macros. my question is this. if i start recording, then include the pattern match contexts, when can I do StopMixMonitor() ? here's the basic gist of the dialplan: http://pastebin.com/gFQAua0w |
| 15:39.13 | [TK]D-Fender | akrohn, I don't see [record-outbound] having any chance of ever being called |
| 15:39.25 | [TK]D-Fender | akrohn, And it will stop itself when the channel ends anyway |
| 15:40.22 | akrohn | that's just a sample. assume you will enter record outbound. ... so i don't have to worry about it? It will end the mixing when the call is terminated? excellent! |
| 15:41.26 | akrohn | will the 's' extension be executed before it searches for a match in the includes? |
| 15:42.02 | *** part/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 15:42.19 | leifmadsen | akrohn: no |
| 15:42.28 | leifmadsen | 's' is not a catch-all |
| 15:42.52 | akrohn | hmm that's unfortunate |
| 15:42.54 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v007-159.mobile.uci.edu) |
| 15:43.32 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v007-159.mobile.uci.edu) |
| 15:46.34 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v007-159.mobile.uci.edu) |
| 15:47.30 | akrohn | what about something like this? http://pastebin.com/tMPpbVLW ... can you use Goto on a context that has nothing but 'include =>' statements? |
| 15:48.18 | ollii | yes |
| 15:48.25 | ollii | sry, wrong channel |
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| 15:58.12 | hackeron | hey, I have exten => 4348739,n,MixMonitor(${CALLFILENAME}.wav,b) in my dialplan, that records the audio just fine, but I want to email the audio at the end of the call, any advise how to do that? |
| 15:58.44 | *** join/#asterisk oldhack (~oldhack@cpe-24-28-23-78.austin.res.rr.com) |
| 15:59.54 | [TK]D-Fender | hackeron, use Monitor instead. |
| 16:00.07 | [TK]D-Fender | hackeron, it offers an option to call a script and mix |
| 16:00.24 | *** join/#asterisk Nasga (~Nasga@82.113.117.78.rev.sfr.net) |
| 16:01.04 | hackeron | [TK]D-Fender: thanks, I'll have a look at Monitor - it calls the script at the end of the call? |
| 16:01.44 | [TK]D-Fender | when monitoring stops |
| 16:02.14 | hackeron | [TK]D-Fender: ok, great, having a look |
| 16:02.50 | *** part/#asterisk swente (~swente@unaffiliated/swente) |
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| 16:09.16 | leifmadsen | has anyone ever implemented a "parked calls list" feature in dialplan? |
| 16:09.31 | leifmadsen | trying to think of which functions or applications I would need to get the list |
| 16:11.29 | Kobaz | do de do |
| 16:11.31 | Kobaz | hey leif |
| 16:11.42 | Kobaz | in the docs in the sip.conf file |
| 16:12.21 | Kobaz | i guess the stuff listed under [general] should be noted that it's only available in general |
| 16:12.46 | Kobaz | several times i've been playing with options trying to get them to work, and then realized when i looked at the source code that it's a global-only option and not per-device |
| 16:12.58 | leifmadsen | Kobaz: huh? I don't see what you're talking about. The only reference to park in sip.conf.sample is the parkinglot option |
| 16:13.10 | Kobaz | park? |
| 16:13.15 | leifmadsen | does not exist |
| 16:13.24 | Kobaz | not talking about park.... like an option like match_auth_username |
| 16:13.27 | Kobaz | is only global |
| 16:13.27 | leifmadsen | Kobaz: I'm looking for the ability to play back a list of parked called |
| 16:13.30 | leifmadsen | calls* |
| 16:13.39 | leifmadsen | Kobaz: I have no idea what you're on about |
| 16:13.42 | Kobaz | while an option like allowtransfer= |
| 16:13.49 | Kobaz | can be set per-peer |
| 16:13.58 | Kobaz | leifmadsen: this is new... nothing to do with your issue |
| 16:14.07 | leifmadsen | Kobaz: oh then I'm not paying attention |
| 16:14.12 | leifmadsen | I have no idea what you're talking about |
| 16:14.21 | Kobaz | okay lemme start over |
| 16:14.26 | Kobaz | it's not obvious in the sip.conf example file |
| 16:14.32 | Kobaz | which options are [general] only |
| 16:14.37 | Kobaz | and which can be set per-user per-peer |
| 16:14.54 | _Corey_ | leifmadsen: I did something a long time ago with an AGI that enumerated parked calls and did something... i can't recall what though. may be an approach worth exploring |
| 16:14.56 | leifmadsen | Kobaz: I agree -- feel free to make the documentation change |
| 16:15.13 | leifmadsen | _Corey_: ya I was hoping to not have to go out and connect to the AMI to get a list, but I might have to |
| 16:15.23 | Kobaz | leifmadsen: that was my only point |
| 16:15.43 | leifmadsen | Kobaz: yes I agree that it could be made more obvious |
| 16:15.43 | _Corey_ | Yeah, there may be another solution... the only one that's coming to mind at the moment would be that though |
| 16:16.07 | leifmadsen | _Corey_: ya, nothing built in dialplan wise it seems. So either need to implement it and add it in out-of-band in this version, or hack something together |
| 16:16.11 | leifmadsen | _Corey_: thaks |
| 16:16.14 | leifmadsen | thanks too |
| 16:16.24 | _Corey_ | good luck :) |
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| 16:31.56 | SuPrSluG | looking at our call reports I saw an odd entry SIP/184.10... unknown "unknown" <unknown>. Each call was 3 seconds. Anyone seen something like this before? |
| 16:32.03 | [TK]D-Fender | leifmadsen, Seems easy enough with DIAL_EXISTS in a loop. |
| 16:33.02 | SuPrSluG | 3 seconds apart. Robo dialer or something more sinister? |
| 16:34.09 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 16:35.14 | ChannelZ | Did they actually dial out to anywhere? I assume SIP/184.10... means it was a whole IP, and not one of your own |
| 16:36.17 | ChannelZ | If so to both then you might reevaluate having allowguest turned on if you don't need it, and if you do, take a hard look at the context and dialplan and make sure random people can't be doing things you don't want them to be able to. |
| 16:36.18 | leifmadsen | [TK]D-Fender: that function doesn't exist |
| 16:36.32 | *** join/#asterisk TimeRider (~steve@204.93.201.17) |
| 16:36.49 | [TK]D-Fender | leifmadsen, Was something like that "dialplan_VALID or somesuch.. its the one one that replaced using chanisavail with Local |
| 16:37.11 | [TK]D-Fender | loops, DIALPLAN_EXISTS |
| 16:37.15 | [TK]D-Fender | opps even |
| 16:37.19 | leifmadsen | [TK]D-Fender: I don't think that'll do it |
| 16:37.26 | [TK]D-Fender | anyway... some obvious name like that |
| 16:37.28 | leifmadsen | that just tells me if a destination in the dialplna exist |
| 16:37.43 | [TK]D-Fender | leifmadsen, How so? You wanted to read them off, right? |
| 16:37.48 | SuPrSluG | No they didn't dial out |
| 16:37.51 | leifmadsen | DIALPLAN_EXISTS just checks for dialplan destination sexisting |
| 16:37.56 | [TK]D-Fender | then you'd just loop though the extens in the parking context and there you go |
| 16:38.04 | leifmadsen | [TK]D-Fender: parked calls, not valid parking spaces |
| 16:38.09 | [TK]D-Fender | as they are created on the fly |
| 16:38.10 | leifmadsen | actively parked calls |
| 16:38.28 | [TK]D-Fender | No, the exten only exists fonce the ARE parked |
| 16:38.35 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 16:38.53 | SuPrSluG | ChannelZ: allow guests is off |
| 16:39.00 | [TK]D-Fender | exists = someone parked there |
| 16:39.01 | leifmadsen | [TK]D-Fender: I see... I'll look into that thanks |
| 16:39.14 | [TK]D-Fender | it doesn't "reserve" them in any way |
| 16:39.21 | leifmadsen | yes i get it thanks |
| 16:39.24 | [TK]D-Fender | alrighty :) |
| 16:40.55 | bindi | uh uh, anyone have a guide on asterisk and instant messaging (SIMPLE?) between the users? |
| 16:41.08 | leifmadsen | I don't think asterisk does SIMPLE msg'ing |
| 16:43.13 | bindi | i've googled around and i somehow got the picture it does |
| 16:43.56 | [TK]D-Fender | "somehow" is the magical world where crazy stuff happens... |
| 16:43.58 | [TK]D-Fender | :) |
| 16:45.09 | bindi | well I need it anyways, if not asterisk, can you recommend me something? |
| 16:47.12 | drmessano | Asterisk 10 does SIMPLE. I've yet to make it work |
| 16:47.34 | Qwell | The SIMPLE messaging...ain't so SIMPLE. |
| 16:49.28 | drmessano | It looks as though the implementation is very basic at this point. If you want something more mature, there's other projects out there that been supporting it much longer |
| 16:50.37 | drmessano | Kamailio comes to mind |
| 16:50.39 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 16:50.43 | drmessano | But there are others |
| 16:52.46 | bindi | drmessano: kamalio looks good |
| 16:52.59 | drmessano | I need to see if it supports Google Voice |
| 16:53.56 | drmessano | Maybe not.. SIP/SIMPLE/XMPP and a bunch of other tricks.. no plugins I can see for GV |
| 16:53.59 | drmessano | Boo |
| 16:55.04 | bindi | what do you need google voice for? |
| 16:55.15 | drmessano | Phone calls :) |
| 16:57.47 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 16:58.06 | drmessano | Tried to take a call at home the other day and found out Asterisks GV support is broken, again. :( |
| 17:10.36 | bindi | drmessano: happen to have some guide for kamailio? how to add users D: |
| 17:13.44 | leifmadsen | drmessano: pretty sure it's the way Google deploys stuff and changes servers back and forth because the "fix" always appears to be to revert or un-revert the same change |
| 17:14.21 | Qwell | yes :( |
| 17:14.25 | Qwell | and the servers aren't consistent |
| 17:14.33 | leifmadsen | right |
| 17:14.35 | Qwell | it could work on one, but not another. depends which patches they've had applied |
| 17:14.41 | leifmadsen | that ^^ |
| 17:14.54 | leifmadsen | it would be good if asterisk could be aware of what it was getting and use method 1 or method 2 |
| 17:15.08 | leifmadsen | depending on how the communications were happening |
| 17:15.21 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 17:16.29 | [TK]D-Fender | bindi, go ask in #kamailio ... and stop ordering Whoppers at McDonalds.. :p |
| 17:16.37 | bindi | thought he'd know :p |
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| 17:46.35 | p3nguin | Mmmmm... flame-broiled goodness... |
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| 18:00.56 | chuckf | is now thinking of a Five Guys burger and fries for lunch |
| 18:01.52 | Qwell | Free pallet with every large order of fries. |
| 18:08.56 | Qwell | thehar: guess what |
| 18:09.38 | thehar | what |
| 18:09.44 | Qwell | thehar: FOCUS! |
| 18:09.47 | thehar | lol |
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| 18:18.15 | jmls | evening all |
| 18:18.23 | jmls | got a strange question |
| 18:18.39 | jmls | can I put someone into a queue *without* answering the call ? |
| 18:18.59 | Kobaz | anyone have any experience with avaya sip phones and asterisk |
| 18:19.03 | jmls | or does the call need answering to play sounds / read digits etc ? |
| 18:19.03 | Kobaz | trying to get mwi to work |
| 18:19.36 | Qwell | jmls: in some limited circumstances it's possible to play audio without answering |
| 18:19.46 | jmls | is interested |
| 18:19.57 | Kobaz | [2012-02-24 13:19:48] -- Got SIP response 400 "Bad Request (Unknown Subscription State)" back from 172.16.100.145:5060 |
| 18:20.02 | Kobaz | is what i get from the avaya phone |
| 18:20.06 | jmls | what are the circumstances ? |
| 18:20.11 | Qwell | jmls: I just work here. |
| 18:20.14 | jmls | heh |
| 18:20.17 | Qwell | jmls: I have no idea. :p |
| 18:20.24 | jmls | lol |
| 18:20.34 | Qwell | Kobaz: sounds like it's sending you a bogus subscription packet |
| 18:20.39 | Qwell | wait, no |
| 18:20.43 | Qwell | I totally misread that. |
| 18:20.50 | Kobaz | the sip notify for mwi went out |
| 18:20.51 | Qwell | your phone is dumb. |
| 18:20.52 | Kobaz | it looks perfect |
| 18:20.54 | *** part/#asterisk OzBob (~Adium@38.105.162.11) |
| 18:20.58 | Kobaz | yeah it looks like a stupid phone |
| 18:21.27 | Qwell | Kobaz: Cisco used to get all pissy about the format for new/old messages. Same type of deal, I bet. |
| 18:21.34 | Qwell | It's rejecting something perfectly valid. |
| 18:21.38 | Kobaz | yeap |
| 18:21.55 | Qwell | hard to tell what it's complaining about, unless you've got something it accepts to compare against |
| 18:22.23 | Qwell | Kobaz: try setting buggymwi=yes |
| 18:22.31 | jmls | Qwell: misreading somethign often leads to problems ... https://www.facebook.com/photo.php?fbid=305713752824512&set=a.239843812744840.58862.239461312783090&type=1&theater |
| 18:23.14 | Qwell | Kobaz: I do mean that literally too. That's the setting for dumb Ciscos. |
| 18:23.27 | jmls | can I make a channel wait for something to happen, or do I have to loop ? |
| 18:23.30 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 18:23.44 | jmls | wait ... jabber read ... |
| 18:24.22 | jmls | _or_ wait for a stomp message |
| 18:24.28 | jmls | hmmm. choices, choices |
| 18:26.00 | Kobaz | Qwell: looks like mwi is not allowed when connected to a non-avaya sip system |
| 18:26.08 | Qwell | seriously? |
| 18:26.10 | Kobaz | yes |
| 18:26.21 | Qwell | wow |
| 18:26.21 | Kobaz | it has survivability-mode |
| 18:26.29 | Kobaz | which is what mode it goes into when it's not connected to an avaya |
| 18:26.36 | Kobaz | and it turns off half the phone features |
| 18:26.50 | Kobaz | like it assumes it's a generic backup system |
| 18:26.59 | Kobaz | i wonder if you can fake out the useragent |
| 18:27.05 | Qwell | heh, maybe |
| 18:27.08 | Kobaz | of asterisk, and then it'll think it's avaya |
| 18:30.35 | leifmadsen | that'd be fun :) |
| 18:36.30 | [TK]D-Fender | jmls, Originate a new channel and send the caller into a ringing loop. Have the originated Local channel join the queue and the other end of it loop a "press 1 to pick up the caller" upon ACK have it Bridge() the originating channel in. |
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| 18:41.41 | Kobaz | <PROTECTED> |
| 18:41.53 | Kobaz | Qwell: that's the sip useragent from an avaya ip office |
| 18:42.12 | jmls | [TK]D-Fender: yeah, so the loop is a simple wait() and a goto ? |
| 18:48.02 | *** join/#asterisk Ionic (ionic@ionic.de) |
| 18:48.34 | [TK]D-Fender | jmls, Pretty much. |
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| 19:03.54 | goathouse774_ | what is a good program for burning isos? |
| 19:05.49 | pabelanger | dd? |
| 19:05.50 | *** join/#asterisk Devon_ (~chatzilla@63.214.236.169) |
| 19:06.29 | Qwell | cdrecord.. |
| 19:07.46 | [TK]D-Fender | keb |
| 19:07.49 | [TK]D-Fender | brassero |
| 19:07.55 | [TK]D-Fender | Bic |
| 19:08.01 | [TK]D-Fender | Zippo |
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| 19:13.44 | goathouse774_ | which is the most like imgburn? |
| 19:13.52 | goathouse774_ | for linux, btw |
| 19:14.31 | Qwell | just use cdrecord. It's trivial. |
| 19:14.38 | Qwell | cdrecord /path/to/file.iso |
| 19:14.38 | goathouse774_ | k, thanks |
| 19:15.06 | goathouse774_ | oh, is that a terminal command? |
| 19:15.12 | Qwell | ...yes |
| 19:15.29 | goathouse774_ | k, thanks |
| 19:19.02 | *** join/#asterisk DarthExpeditor (~IceChat9@96-42-133-130.static.trcy.mi.charter.com) |
| 19:20.23 | DarthExpeditor | I am having trouble transfering a inbound sip call from one Aastra 480i to another Aastra 480i. Log here: http://pastebin.com/WZTAXnPv |
| 19:20.44 | *** part/#asterisk mog (~mog@fsf/member/mog) |
| 19:21.18 | bindi | drmessano: why do you use google voice btw? i'm interested in what it does so good :p |
| 19:21.44 | p3nguin | It does totally free calls inbound and outbound. |
| 19:21.56 | bindi | i guess that's US only? |
| 19:22.01 | Qwell | and Canada |
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| 19:25.12 | DarthExpeditor | I should clarify. Asterisk 1.8.9.3 |
| 19:25.25 | DarthExpeditor | I had the problem with 1.8.7.2 as well. |
| 19:25.41 | Qwell | DarthExpeditor: There's a lot of cruft to wade through in that log. |
| 19:25.58 | DarthExpeditor | Do you want me to redo it without the SIP debug? |
| 19:26.28 | Qwell | no, there's just a lot of extra stuff in there that isn't important or related to that call |
| 19:26.45 | DarthExpeditor | hmmmm.... |
| 19:27.28 | *** join/#asterisk neurosys_ (~neurosys@69.198.141.161) |
| 19:27.41 | stephendowed | ummm aside twinkle ekiga please what other softphone can i use for my linux box |
| 19:30.33 | [TK]D-Fender | http://voipguides.blogspot.com/2009/07/best-sip-softphones-for-voip-on-linux.html |
| 19:30.54 | [TK]D-Fender | stephendowed, When in doubt (and there shouldn't ever really be any) .... JFGI :) |
| 19:33.44 | DarthExpeditor | http://pastebin.com/h3tAwhDx |
| 19:33.52 | DarthExpeditor | There is a simplier version of the log |
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| 19:45.01 | akrohn | Had an interesting bug happen to me yesterday. I configured a phone's 'display line' as "5202 <5202>"... turns out whatever setting this is gets inserted into the SIP From header. Well that header already has <fooo> brackets in it. So when asterisk tries parsing the header, the connected call drops after a few seconds. |
| 19:45.36 | Qwell | akrohn: upgrade |
| 19:45.39 | akrohn | obv not an asterisk bug. just... quirk in general |
| 19:46.03 | akrohn | oh, is that something that might be fixed? i assumed i'm the dumb one for breaking SIP headers ! |
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| 21:51.15 | oglynn | On Asterisk 10.1.2 while logged on as root if i issue "service asterisk start" all works fine. however if the server reboots and asterisk auto starts i get "[Feb 24 16:51:05] WARNING[5473] db.c: Unable to open Asterisk database '/var/lib/asterisk/astdb.sqlite3': unable to open database file" |
| 21:52.18 | *** join/#asterisk _jdccdevel (~chatzilla@69.196.87.218) |
| 21:53.06 | oglynn | anyone got any ideas why it would behave differently? |
| 21:55.28 | _jdccdevel | Hi! Has anyone had experience faxing through asterisk using the Cisco SAP8000 as the ATA? (asterisk 1.6.22 with the t38 gateway patch) ? |
| 21:57.03 | _jdccdevel | Oops. Sorry, that should be SPA8000 |
| 22:00.02 | bindi | where can i get something like asterisk management portal? |
| 22:00.22 | bindi | so openfire's asterisk-im can use it |
| 22:00.32 | p3nguin | Something like it? What's wrong with ... it? |
| 22:00.44 | bindi | well, it seems like it's dead |
| 22:00.48 | p3nguin | AMI? |
| 22:00.51 | p3nguin | Dead? |
| 22:01.06 | bindi | well i was looking at AMP |
| 22:01.15 | Qwell | AMP was renamed like...4+ years ago |
| 22:01.18 | p3nguin | Oh. I don't even know what that is. |
| 22:01.30 | Qwell | p3nguin: take a wild guess |
| 22:01.34 | Qwell | You'll be right. |
| 22:01.38 | bindi | 'python-asterisk'? |
| 22:01.46 | bindi | ah no nevermind lol |
| 22:01.56 | p3nguin | I just don't know it by its old name. |
| 22:02.03 | p3nguin | And I don't use it by its new name. |
| 22:02.08 | bindi | does it happen to be in the repos? |
| 22:02.14 | bindi | using debian squeezy |
| 22:02.14 | Qwell | "the repos"? |
| 22:02.52 | bindi | apparently not. yeah, debian repos or w/e you call them :p |
| 22:03.11 | p3nguin | Isn't it just a matter of making sure you enable it? |
| 22:03.13 | bindi | wait, is it built in? |
| 22:03.13 | bindi | yeah.. |
| 22:04.26 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:06.56 | bindi | now, how do i do phone mapping? |
| 22:07.42 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-lchnadajynjqqijf) |
| 22:08.03 | bindi | or do i even need that. gonna try |
| 22:09.16 | p3nguin | What does it mean? |
| 22:10.31 | bindi | irrelevant to this channel i guess, i'm trying to get instant messaging, installed openfire and asterisk-im |
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| 22:10.54 | drmessano | Openfire, UGH |
| 22:21.35 | bindi | how on earth does asterisk-IM even work |
| 22:25.44 | *** part/#asterisk sustav (~alfa@nat/digium/x-wewwslifyvnebhjg) |
| 22:28.41 | drmessano | You need the Spark client for it |
| 22:29.09 | drmessano | You can initiate calls from Spark, get call notifications on Spark, and when your extension is busy, you can see the presence in Spark |
| 22:29.16 | drmessano | Thats pretty much it |
| 22:30.13 | drmessano | Youre basically setting up an entirely different messaging platform that has some small hooks into Asterisk for call origination and presence |
| 22:30.19 | drmessano | Not worth it, IMO |
| 22:30.48 | bindi | yeah uh, ... *kills self* |
| 22:31.09 | bindi | I read that newer versions of asterisk have SIMPLE support, is this true? I have 1.6.4 from the debian repos |
| 22:31.24 | drmessano | You asked that earlier, and I answered Yes |
| 22:31.33 | bindi | i don't remember asking that but okay |
| 22:33.03 | drmessano | Actually, you asked if someone had a guide for setting that up. Leif commented that he didn't believe Asterisk had SIMPLE support, and I replied that I did. |
| 22:33.11 | drmessano | So technically, you are correct |
| 22:34.31 | drmessano | But anyway.. It does.. It's not a complete SIMPLE implementation, but it is supposed to support presence and messaging through the dialplan. Documentation appears rather limited at this point, since the messaging was new for Asterisk 10 |
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| 22:36.54 | bindi | http://pastebin.com/repQfEN4 |
| 22:36.56 | bindi | uh uh... |
| 22:37.31 | Qwell | bindi: it tells you exactly what to do. |
| 22:37.42 | bindi | oh lol |
| 22:37.46 | bindi | slaps self |
| 22:39.07 | bindi | how can make it use mysql instead of sqlite? |
| 22:39.41 | bindi | guess i can't |
| 22:40.42 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:42.05 | Qwell | bindi: 1.6.4? No such thing. |
| 22:42.45 | bindi | huh? |
| 22:43.29 | bindi | drmessano: do i need to select simple support in the menuselect? |
| 22:53.14 | bindi | right. how do i have to do my extensions.conf now? |
| 22:54.03 | [TK]D-Fender | ~book |
| 22:54.03 | 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://www.asteriskdocs.org/ or see ~buybook |
| 22:54.04 | [TK]D-Fender | ^^^^^^^ |
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| 23:13.15 | bindi | right, that's a lot of reading and a lot of not understanding |
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| 23:25.31 | [TK]D-Fender | A lot of reading ... in the 18 minutes since I just linked you to the book? |
| 23:25.41 | [TK]D-Fender | Nice to see you're dedicated |
| 23:32.38 | bindi | so does that book have anything about the new SMS? |
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| 23:36.25 | [TK]D-Fender | probably not |
| 23:36.32 | [TK]D-Fender | I'd check the Asterisk WIKI |
| 23:37.59 | drmessano | SMS is not new in Asterisk |
| 23:38.09 | bindi | well SIMPLE or w/e <.< |
| 23:38.34 | drmessano | SMS and SIMPLE are two completely different applications |
| 23:39.48 | bindi | yeah yeah |
| 23:40.10 | drmessano | O.o |
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| 23:40.29 | bindi | i am aware, was just confuzzled |
| 23:40.57 | drmessano | Are you? |
| 23:40.59 | bindi | http://pastebin.com/1FpKckfT I get this when I try to send messages, what kind of configuring do I need to do |
| 23:41.28 | drmessano | Are you running Asterisk 10? |
| 23:41.31 | bindi | yes |
| 23:41.54 | drmessano | Have you read the sip.conf examples on sending messages thru the dialplan? |
| 23:42.51 | bindi | i have not |
| 23:43.10 | bindi | trying to find it, which keyword can i use to find it? |
| 23:43.29 | drmessano | Sendtext I do believe |
| 23:43.39 | bindi | "sendtext" not found |
| 23:44.45 | drmessano | accept_outofcall_message |
| 23:44.48 | drmessano | and the next 2 |
| 23:46.10 | bindi | i'm not sure what ""outofcall_message_context" means |
| 23:51.15 | bindi | [Feb 24 23:48:03] NOTICE[25221]: chan_sip.c:26952 sip_request_call: Asked to get a channel of unsupported format (nothing) while capability is (gsm|ulaw|alaw|h263|testlaw) |
| 23:51.22 | bindi | :/ |
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