| 00:15.11 | *** join/#asterisk `paul (~kutimoy@119.93.15.82) |
| 00:15.37 | `paul | can i config asterisk to save cdr to two separate mysql DB at the same time? |
| 00:19.21 | b14ck | Hey guys, I think I have relaxdtmf understood backwards. |
| 00:19.33 | b14ck | I thought that if relaxdtmf=yes, then asterisk would be more lenient about voices that are detected as dtmf. |
| 00:19.41 | b14ck | But I just read somewhere that it should be no for that. |
| 00:19.59 | b14ck | Basically, some users' voices are mistaken as dtmf, should I set relaxdtmf=yes or no? |
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| 01:10.30 | pabelanger | rustyclarkson: Hmm, I'm surprised to here that |
| 01:11.42 | *** part/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 01:30.46 | MrUnagi | i have never heard of voices being recognized as dtmf |
| 01:31.35 | tzanger | MrUnagi: it happens |
| 01:31.38 | MrUnagi | weird |
| 01:31.56 | tzanger | the old TE403/5 had the issue with the Oki-fabbed EC |
| 01:36.59 | *** join/#asterisk JamesVoss (j@unaffiliated/jamesvoss) |
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| 02:03.27 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 02:22.11 | jamko | Every time I start asterisk, I see this in the console on time for each family name: Realtime mapping for 'family_name_from_extconfig.conf' found to engine 'mysql', but the engine is not available |
| 02:22.42 | jamko | I can connect to mysql, view the DB etc, and all user names and passwords appear to be correct. |
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| 03:30.24 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 03:52.29 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 04:34.17 | jamko | constantly getting this message in console: WARNING[10081]: res_config_mysql.c:159 realtime_mysql: MySQL RealTime: Failed to query database. Check debug for more info. |
| 04:34.51 | jamko | but realtime is working and 'realtime mysql status' returns that it is connected. |
| 04:36.21 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
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| 04:47.03 | EmleyMoor | ChannelZ about? |
| 04:49.59 | ChannelZ | about what? |
| 04:50.08 | EmleyMoor | As in here |
| 04:50.19 | ChannelZ | oh |
| 04:50.24 | ChannelZ | well, answered then |
| 04:51.40 | EmleyMoor | SIP Debug for a successful call over WiFi from my N97 - http://asterisk.pastey.net/139698 - ntill failing over 3G as you saw yesterday |
| 04:51.45 | EmleyMoor | still, even |
| 04:51.49 | *** join/#asterisk ankur_6997 (~Dev_1@122.177.214.111) |
| 04:52.06 | EmleyMoor | hrhrhr may be interested too |
| 04:53.10 | ankur_6997 | hi i had setup callcentric with trixbox for IVRS system that's fi9 but i want my custum ivr that can launch AGI scripts how to do it ? please help |
| 04:53.43 | ChannelZ | The difference is that when it INVITEs the first time and gets rejected, it tries again with the proper authorization. Last night the debug seemed to show that it just gave up and kept re-registering after being rejected. |
| 04:53.50 | ChannelZ | Why that is, you'd have to ask the phone/app |
| 04:54.13 | ChannelZ | fi9? seriously? |
| 04:54.57 | ChannelZ | in any event I guess you'd have to ask Trixbox |
| 04:55.05 | ChannelZ | (the PBX for whores!) |
| 04:55.06 | EmleyMoor | ChannelZ: I am in consultation with Nokia about it - though nobody clueful has reached me yet |
| 04:55.41 | EmleyMoor | Not Asterisk's fault of course - just handy it could get the information |
| 04:55.58 | ankur_6997 | where to write custom ivrs scripts and map then to a particular incomming DID ? |
| 04:56.09 | ChannelZ | I don't know why it's responding differently via WiFi |
| 04:57.06 | ChannelZ | ankur_6997: ??? this is not a trixbox support channel |
| 04:57.24 | ankur_6997 | ok |
| 04:57.41 | ChannelZ | If you were doing vanilla asterisk you edit extensions.conf |
| 05:02.36 | jamko | constantly getting this message in console: WARNING[10081]: res_config_mysql.c:159 realtime_mysql: MySQL RealTime: Failed to query database. Check debug for more info. |
| 05:02.36 | jamko | <jamko> but realtime is working and 'realtime mysql status' returns that it is connected. |
| 05:02.57 | ChannelZ | EmleyMoor: Still don't know why setting the insecure options wouldn't bypass the auth, did you reload after making the changes? |
| 05:03.07 | [TK]D-Fender | jamko: Perhaps a query format failure due to invalid table design? |
| 05:03.15 | EmleyMoor | ChannelZ: Yes, I did |
| 05:04.14 | EmleyMoor | If I find the answer, especially if it is configurable around in Asterisk, I will keep you and the rest of the channel posted |
| 05:04.15 | jamko | TK: possibly, but where can I see more specifics on the error? Dumping debug shows nothing more. |
| 05:04.54 | Gugge | jamko: you could turn on the querylog on your mysql, and check what mysql receives from * |
| 05:05.29 | jamko | thanks.. will try that. |
| 05:06.02 | ChannelZ | EmleyMoor: I'm looking for a bug related to insecure possibly being broken in your version but not finding anything so far |
| 05:07.18 | ChannelZ | pastebin your sip.conf |
| 05:08.04 | *** join/#asterisk lyetz (~lyetz@me.lyetz.me) |
| 05:09.24 | drmessano | ChannelZ: You do know this all started after he did a firmware upgrade on his phone, correct? |
| 05:09.54 | ChannelZ | dunno |
| 05:10.27 | ChannelZ | either way the insecure option should be able to work around what is happening, which means either his particular version of asterisk is busted or he didn't config it right |
| 05:11.18 | drmessano | I would be more willing to blame the upgraded firmware on his Nokia.. |
| 05:11.34 | drmessano | Supposed nothing else changed |
| 05:11.39 | drmessano | Supposedly* |
| 05:12.47 | ChannelZ | Something is definately not right with the phone, but again at least from the SIP debugs I've seen asterisk is also misbehaving. |
| 05:13.05 | drmessano | He said it was working fine over 3G before the upgrade |
| 05:13.23 | ChannelZ | You're not listening |
| 05:13.39 | drmessano | lame |
| 05:13.52 | ChannelZ | It's refusing the initial INVITE because of no auth. If insecure is indeed turned on for the right peer, asterisk shouldn't be doing that. |
| 05:14.02 | drmessano | I could accuse you of ignoring the obvious, but that would be just as bad |
| 05:14.24 | *** join/#asterisk pinoyskull- (~pinoyskul@124.6.182.55) |
| 05:14.26 | ChannelZ | No I hear what you're saying and understand and AGREE that the phone is retarded |
| 05:14.51 | bougyman | phones can't be retarded |
| 05:15.01 | bougyman | they're either broken a lot, broken a little, or barely broken. |
| 05:15.06 | bougyman | none are break-free. |
| 05:15.08 | drmessano | You're not listening to the part about how ASTERISK and his PHONE were working fine before he upgraded the firmware on the PHONE.. then it stopped working. |
| 05:15.23 | *** part/#asterisk ankur_6997 (~Dev_1@122.177.214.111) |
| 05:15.23 | ChannelZ | No I understand that perfectly well. |
| 05:15.35 | drmessano | You're trying to fix something that wasn't broken |
| 05:16.56 | ChannelZ | Forget it I'm not going to explain in circles |
| 05:18.14 | *** join/#asterisk `Sauron (sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 05:20.24 | EmleyMoor | Anyway, FWIW, my sip.conf is at http://asterisk.pastey.net/139730 |
| 05:21.12 | EmleyMoor | I might try off-site WiFi a bit later - I expect it to work |
| 05:22.19 | *** join/#asterisk jdneat (~jdneat@c-98-197-56-14.hsd1.tx.comcast.net) |
| 05:22.19 | [TK]D-Fender | Why are we not seeing the CALL? |
| 05:22.59 | jdneat | Fender: I know i'm comming in late to this but Is it a video call? Otherwise I feel like it would be hard to see.... |
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| 05:58.28 | jamko | http://pastebin.com/F2QQK6JJ ---- mysql logs for this asterisk error: WARNING[10081]: res_config_mysql.c:159 realtime_mysql: MySQL RealTime: Failed to query database. Check debug for more info. |
| 05:59.54 | jamko | it looks to me like asterisk is querying for a blank host name, but not sure. |
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| 07:13.42 | DND | guys is there any windwos software that i can use to check for call status of extensions? |
| 07:14.01 | golikwid|mac | hud |
| 07:14.04 | DND | or maybe a web-based panel aside from fop |
| 07:14.12 | golikwid|mac | fop2 |
| 07:14.19 | DND | you mean hudlite? |
| 07:14.29 | DND | fop2 is paid |
| 07:14.30 | golikwid|mac | isymphony |
| 07:14.42 | golikwid|mac | so pay for it its 40 bucks |
| 07:15.11 | DND | it will only be used by one person. just need to exhaust opensource first before paying |
| 07:15.32 | golikwid|mac | hudlite than |
| 07:15.38 | DND | ok will try that.. thanks |
| 07:17.32 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 07:23.12 | *** join/#asterisk russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 07:23.26 | russ | did anyone else notice that teliax seems to have purchased a cidname database? |
| 07:24.04 | russ | I got my DID a few days ago and was very pleasantly surprised |
| 07:27.38 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:29.54 | ChannelZ | isn't the old fop free? |
| 07:30.56 | russ | ya |
| 07:31.17 | russ | fop vs fop2 |
| 07:31.40 | russ | so fop 0.30 is still free |
| 07:33.12 | ChannelZ | holy crap, hudlite runs an entire IRCd?! |
| 07:34.14 | ChannelZ | And it's written in perl. Is it just irony they put 'lite' in the name? |
| 07:34.48 | russ | anything that can compile to bytecode and or native code is actually really fast |
| 07:35.01 | Tim_Toady | developed by fonality, that should ring some bells :> |
| 07:35.21 | russ | amazed at how fast python can chew through a 16MB csv file, do processing, and produce a 64MB xml file |
| 07:36.29 | ChannelZ | I suppose the ircd is optional for their IM. |
| 07:36.34 | ChannelZ | Looks interesting but I think I'll pass |
| 07:37.33 | *** join/#asterisk lyetz (~lyetz@me.lyetz.me) |
| 07:37.33 | russ | xmpp is more buzzworthy anyway |
| 07:37.40 | russ | er, buzz-word-worthy |
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| 10:43.35 | Dovid | has anyone here set up Asterisk+SS7 ? |
| 10:53.54 | DND | sorry Dovid what is ss7? |
| 10:55.13 | DND | Dovid i think this is what you are looking for? http://www.voip-info.org/wiki/view/Asterisk+libss7 |
| 11:07.31 | Dovid | DND: I hae seen that but I need an expert on it, that we can pay when we have issues on the set up. |
| 11:10.04 | *** join/#asterisk lftsy (~lftsy@pul-lav-fw-so-01-x1.vtxnet.net) |
| 11:10.08 | Secret_Hamster | err Signalling System 7 its one of the primary telephony signalling protocols |
| 11:14.37 | golikwid|mac | hey Dovid what are you using ss7 for? |
| 11:15.01 | golikwid|mac | are you a CLEC? |
| 11:15.50 | Dovid | golikwid|mac: Carrier in Cyprus |
| 11:16.54 | *** join/#asterisk guilhermebr (~Guilherme@189.50.119.172) |
| 11:16.59 | golikwid|mac | the RO Cyprus? |
| 11:17.35 | Dovid | RO == ? |
| 11:17.43 | golikwid|mac | repubic of |
| 11:18.07 | Dovid | i assume thats what it's called. i manage part of it remotely. don't live there |
| 11:18.27 | golikwid|mac | oh ok |
| 11:18.29 | golikwid|mac | anyway |
| 11:18.55 | golikwid|mac | im looking in to ss7 as well and would be interested in adding an ss7 professional to my rolodex |
| 11:19.48 | *** join/#asterisk DelphiWorld (~Delphi@41.200.25.201) |
| 11:19.54 | Dovid | golikwid|mac: Sent you a PM |
| 11:19.57 | DelphiWorld | hi |
| 11:20.10 | DelphiWorld | Katty: hello |
| 11:24.29 | DelphiWorld | could someone help me configuring asterisk? |
| 11:24.51 | Dovid | DelphiWorld: What is your issue ? |
| 11:25.25 | DelphiWorld | Dovid: no issu but i am runing it into a router and don't know how do i configure it |
| 11:27.41 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 11:27.54 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 11:27.55 | Dovid | ok. it is behind NAT ? |
| 11:28.31 | DelphiWorld | Dovid: no, 100% unnated |
| 11:29.32 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
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| 11:31.02 | Dovid | So whats the issue ? What do you need to set up ? The router or Asterisk ? |
| 11:32.03 | DelphiWorld | Dovid: the issus is the way of configuration |
| 11:32.12 | DelphiWorld | Dovid: i can't edit files i think you know allready |
| 11:37.02 | SiNGLer | why you can't edit files? |
| 11:37.37 | DelphiWorld | SiNGLer: i am blind using text to speech from windows, using ssh to connect to linux box but screen reader's don't read shell based editors |
| 11:37.59 | SiNGLer | oh |
| 11:38.27 | *** join/#asterisk ccesario (~ccesario@187.75.139.188) |
| 11:38.29 | SiNGLer | what about download files, edit it locally and upload back? |
| 11:39.07 | tzafrir | DelphiWorld, considered using ex mode of vi? |
| 11:39.08 | DelphiWorld | SiNGLer: how to download it? |
| 11:39.27 | DelphiWorld | tzafrir: what is exmode? |
| 11:39.41 | tzafrir | the command-mode editing of vi |
| 11:39.52 | DelphiWorld | tzafrir: how to try it? |
| 11:40.02 | tzafrir | sorry. line-mode editing. Not full-screen mode |
| 11:40.29 | SiNGLer | DelphiWorld: you can use sftp protocol. You could use WinSCP, maybe some other program |
| 11:40.32 | DelphiWorld | tzafrir: how to try it? |
| 11:40.47 | DelphiWorld | SiNGLer: let try winSCP |
| 11:42.13 | *** join/#asterisk darkskiez (~dz@62-50-207-34.client.stsn.net) |
| 11:43.41 | DelphiWorld | SiNGLer: wow |
| 11:43.47 | DelphiWorld | SiNGLer: did you know that you rock? |
| 11:44.28 | SiNGLer | maybe :) |
| 11:44.43 | DelphiWorld | SiNGLer: winSCP 100% solved my problem |
| 11:44.51 | DelphiWorld | SiNGLer: no download required |
| 11:45.07 | SiNGLer | great to hear that it helped you |
| 11:45.47 | DelphiWorld | SiNGLer: :D |
| 11:52.56 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
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| 12:04.30 | nunners | Afternoon all... a couple of quick questions, mainly for discussion, rather than being able to answer straight away... I'm currently using asterisk 1.4.35, with DAHDI and asterisk_GUI (I know but it was easy when I first setup)... I'm now thinking are there any benefits of upgrading to 1.6 and whether there's another free GUI worth using that does things properly... unlike the gui muckiing... |
| 12:04.32 | nunners | ...the conf files up... |
| 12:05.41 | [TK]D-Fender | nunners: The point of the GUI's is to keep you out of the configs. |
| 12:06.16 | [TK]D-Fender | nunners: You GIVE UP control to them, so if you don't like the "fine print", then you should have realized it was actually the BIG PRINT :) |
| 12:07.04 | [TK]D-Fender | nunners: In the free dept the most decent would still be FreePBX (which is in the process of forking) |
| 12:07.24 | nunners | [TK]D-fender: fair point... and yes, I did think that the fine print wasn't worth worrying about when we setup, and now when I want ot do something slightly outside the most basic, asteriskGUI mucks it up! FOr isntance, setting up the ability to call a third party number out of hours seems beyond asteriskGui! |
| 12:08.13 | nunners | [TK]D-fender: whats happening with Freepbx? I did try it once, but couldn't get it working, so went for a straight install.. |
| 12:08.41 | [TK]D-Fender | nunners: Everyone else runs it just fine. We have nothing to comment on regarding your tpast troubles with it |
| 12:10.01 | nunners | [TK]D-fender: I know, it's so long ago I can't remember what the problem was - probably OS related, and I couldn't be bothered ot fix it! I'm actually about to do a new phonesystem for a friend, and they definitely want a nice Gui, and preferably one you don't have to pay for... so yes Freepbx will probably be the option.... |
| 12:10.23 | nunners | Also, what's the main benefits/issues between 1.4 and 1.6? |
| 12:11.02 | [TK]D-Fender | nunners: GUI people probably won't CARE about the difference. GUI's aren't "interesting" or take advantage of many of the features * gets |
| 12:11.59 | nunners | [TK]D-Fender: but if a non-technical user wants to edit things GUIs are useful, but then if a technical person wants to adjust something a bit cleverer, then the conf files need to be understandable! |
| 12:12.27 | drmessano | No they dont |
| 12:12.45 | drmessano | They're not meant to be read by a human.. they're auto generated configs |
| 12:13.50 | nunners | But it would help if things were put in the right conf files, rather than stuff everything into the "wrong" single file? |
| 12:14.24 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
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| 12:14.41 | [TK]D-Fender | nunners: GUI's don't work around you, they work INSTEAD of you can can't afford to have you screwing with them too much |
| 12:15.19 | [TK]D-Fender | nunners: FreePBX has many "hook spots" and little tricks to let you do bunch of stuff without too much difficulty, its the REST that will completely screw you over |
| 12:15.47 | [TK]D-Fender | nunners: So you have better get VERY precise about what "special" things you want to do if you want good advice as to what will or will not work. |
| 12:17.26 | nunners | Ok, fair point again... the technical thing I've just done is adding a BLF group, which you can't do from the GUI, and whilst it being actually quite simple, it was hampered slightly by the way the gui had put for instance the sip.conf stuff in the users.conf file... but then if I wasn't here tomorrow and another user wanted to change someone from one group to another, they'd use the gui....... |
| 12:17.28 | nunners | ...does that make sense? |
| 12:18.20 | [TK]D-Fender | nunners: if you are talking BLF for direct access via your phones you could do this in a separate INCLUDE file |
| 12:19.19 | *** part/#asterisk nunners (~chatzilla@host81-149-123-240.in-addr.btopenworld.com) |
| 12:19.47 | [TK]D-Fender | BAI BAI |
| 12:21.59 | jamko | Could someone check this out: http://pastebin.com/F2QQK6JJ - and speculate why asterisk is giving me: WARNING[10081]: res_config_mysql.c:159 realtime_mysql: MySQL RealTime: Failed to query database. Check debug for more info. |
| 12:22.43 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 12:22.50 | jamko | that is a mysql query log. Peers are registering with realtime, but that message is constant. |
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| 12:25.49 | [TK]D-Fender | jamko: Doesn't show anything funny there. max out *'d debug, and show some complete backup to match |
| 12:34.07 | jamko | TK: debug to 4? |
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| 12:43.31 | [TK]D-Fender | Jam20 |
| 12:43.38 | [TK]D-Fender | JamCORE debug, verbose, the works |
| 12:44.03 | jamko | ok.. going to post from 'full' |
| 12:44.11 | [TK]D-Fender | JamkCLI, not LOGS |
| 12:44.33 | [TK]D-Fender | is on auto-complete FAIL Day |
| 12:44.43 | jamko | ok |
| 12:45.57 | [TK]D-Fender | look at logs in parallel with CLI to effectively timestamp the points of failure |
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| 12:59.12 | jamko | TK: http://pastebin.com/FCDPVfG8 |
| 12:59.59 | henk | hi, i am setting up an asterisk behind a iptables firewall. registration to my provider via sip/tcp works fine, but it seems nothing else comes through to asterisk. i used tcpdump to verify there is traffic on port 5060 both on the firewall and the asterisk server. the traffic coming from and going to my sip provider. rasterisk -dddvvv doesn't show any sign of an incoming call though even with sip set debug on. did i overlook something? do i need other ... |
| 13:00.05 | henk | ... ports open in the firewall? how could i debug that further? |
| 13:00.25 | jamko | TK: There is not much going happening on this box. It's just a test box with 2 ua registered to it. That message seems to correspond to registration attempts, but I could be wrong. |
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| 13:01.16 | [TK]D-Fender | henk: Nothing in sip debug (if you didn't restrict it) = no packets. |
| 13:02.36 | [TK]D-Fender | jamko: Is there anything that actually fails? |
| 13:03.11 | *** join/#asterisk skyion (~bradc@siza.thusa.net) |
| 13:03.48 | jamko | TK: Not that I am aware of. The MySQL query log shows this attempt is on the SIP table, but realtime registrations are working. |
| 13:03.50 | skyion | is there a way to determine how long a sip channel has been on an active call for from the asterisk console? |
| 13:04.28 | skyion | I know it gets written to mysql after the call has completed, but trying to hunt down calls that aren't disconnecting properly. |
| 13:05.39 | henk | [TK]D-Fender: hm ok, thank you, going on with my search... |
| 13:07.41 | [TK]D-Fender | jamko: In your debug I see it appearing only after * sends an "options" packet. Could be you are missing a field for the qualify time, etc |
| 13:08.03 | [TK]D-Fender | jamko: but it doesn't appear to be triggered by a more important action |
| 13:09.46 | Dovid | TK: Every worked with libss7 ? |
| 13:10.11 | [TK]D-Fender | Dovid: nope |
| 13:11.11 | Dovid | know anyone that has ? |
| 13:14.47 | [TK]D-Fender | Dovid: Plenty of people have, I just never remembered the names... |
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| 13:19.58 | Dovid | lol |
| 13:20.03 | Dovid | Anyone from Digium here? |
| 13:20.07 | henk | [TK]D-Fender: http://pastie.org/private/7yi31tqnqc0s5mm0wkqw this is what comes in on my server. i can't really interpret that. is it a call or something else? |
| 13:20.43 | [TK]D-Fender | henk: no, that is a response to an OPTIONS packet (qualify) |
| 13:20.53 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 13:23.04 | henk | [TK]D-Fender: hm ok... i see a tcp connection to my provider (which is a cisco 2801 i have access to as well). are calls coming in via this connection? i'm asking because there might be some firewall rules missing... |
| 13:23.16 | jamko | TK: 4 |
| 13:23.21 | jamko | woops.. |
| 13:23.56 | [TK]D-Fender | henk: I see nothing to comment on, and a lack of a decent description. I think you'll have to do far better... |
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| 13:25.51 | henk | [TK]D-Fender: yeah, i'm probably a bit lost, sorry :( i'll try giving a thorough description of my setup. |
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| 13:28.07 | henk | we terminate an E1 line with several number ranges with some 'teles' device i do not really know anything about. they are fed to a cisco 2811. on this device i have a sip-via-tcp account. my asterisk is behind a linux firewall which allows port 5060 from and to my asterisk ip. asterisk registers to the cisco without problems. |
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| 13:29.50 | [TK]D-Fender | henk: place calls. If you don't see them come in, then your Cisco isn't getting the call or isn't configured right to pass ti on to *. |
| 13:30.47 | henk | now when i place a call to one of the number ranges, the cisco tries to route it to the asterisk. i can see several packets coming in on the firewall on port 5060 and responses from asterisk. i have not yet examined them closer in wireshark though. asterisk doesn't say anything about an incoming call though, even with sip set debug on. now i wonder if it's the firewall blocking some other connection that is being established. |
| 13:32.58 | henk | [TK]D-Fender: ok, i don't see them... the cisco seems to try to pass it on to *, but somehow fails afaict. is it only port 5060 to and fro for a sip-via-tcp connection or something else? |
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| 13:33.41 | jamko | henk: the only other ports are rtp, but that won't affect the sip invite attempts to your box. |
| 13:34.28 | hrhrhr | henk: 5060 udp and tcp? |
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| 13:35.22 | henk | hrhrhr: no, only tcp. sip _either_ works over udp or tcp, right? it doesn't need udp for tcp-sip connections, does it? |
| 13:35.32 | drmessano | udp |
| 13:36.02 | drmessano | TCP SIP use is minimal |
| 13:36.17 | drmessano | 99.9% of the world uses UDP on SIP |
| 13:37.14 | coppice | not true. 99% of client terminal use UDP may be true |
| 13:37.18 | henk | so i am one of 0.1%. sip via tcp still doesn't need any udp port, right? |
| 13:38.07 | hrhrhr | it will take you 20 secs to forward udp to find out :P |
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| 13:38.40 | henk | hrhrhr: assuming it works when i did this. if it doesn't i'm not any wiser than before... |
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| 13:40.23 | drmessano | If the cisco is indeed using TCP, Asterisk must also be configured to use TCP on SIP.. it's not on by default |
| 13:41.46 | tzanger | good morning... I am hearing rumour that oslec's capture device has been rolled into DAHDI (i.e. no extra patch needed) -- is this true? I'm having some trouble seeing it if it is |
| 13:42.28 | drmessano | henk: Namely tcpbindaddr= and tcpenable=yes in sip.conf [general] |
| 13:42.49 | henk | drmessano: it is using tcp, i can see the conf. asterisk also uses tcp for that connection and registers just fine. it says "Prim.Transp.: TCP" when i do 'sip show peer voicegatehsr' which is the peer i have problems with... |
| 13:42.54 | tzafrir | tzanger, I haven't heard it |
| 13:43.01 | drmessano | ok |
| 13:43.34 | drmessano | ~debug |
| 13:43.35 | infobot | ACTION DeBuggers $1 |
| 13:43.41 | drmessano | ~sipdebug |
| 13:43.44 | drmessano | gah |
| 13:43.51 | henk | but it seems i found one problem: access-list on the cisco... |
| 13:44.00 | henk | i guess i'll have to fix that before i go on... |
| 13:44.02 | coppice | tzanger: wouldn't that be bad for EC hardware sales? |
| 13:45.23 | drmessano | tzanger: All I do is add the needed kernel sources.. no other patching |
| 13:46.27 | tzanger | coppice: :-) |
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| 13:47.06 | tzanger | got a "customer" (friend of a friend) who's got nasty echo with MG2 on FXO even though he's in a city centre |
| 13:47.11 | tzanger | (i.e. not a long loop) |
| 13:47.29 | drmessano | tzanger: http://www.rowetel.com/blog/?page_id=454 |
| 13:47.32 | coppice | MG2 is pretty useless. its author agrees |
| 13:47.36 | tzanger | he's positive it's the EC and I'm positive it's becuase he appears to have about 12dB of loss on the line |
| 13:47.59 | drmessano | Simple download of the kernel sources, a few SED's and make |
| 13:48.03 | [TK]D-Fender | drmessano: * TCP IS working fine |
| 13:48.18 | [TK]D-Fender | drmessano: He jsut insn't gettnig calls from the Cisco due to misconfiguration on its side |
| 13:48.28 | tzanger | capturing the local milliwatt number normally shows about 4dB of loss in my experience, and the various specs on telco lines tells me that up to even 8.5dB loss is acceptable (that seems ridiculously high to me) |
| 13:48.52 | tzanger | but with gains of 0 and (he says he is) verified 0 gain/atten on the DAA, milliwatt shows up as -12dBm |
| 13:49.09 | coppice | in most countries the copper planning limit (or whatever they call it locally) can give you about 17dB loss |
| 13:49.36 | tzanger | 17dB?! how the hell are you expected to hear the other side? |
| 13:49.46 | m0t3jl | How many channels does GSM offer? |
| 13:50.03 | [TK]D-Fender | tzaSorry, can you repeat that? |
| 13:50.09 | [TK]D-Fender | tzanger: Sorry, can you repeat that? |
| 13:50.21 | [TK]D-Fender | m0t3jl: what "GSM"? |
| 13:50.23 | tzanger | [TK]D-Fender: exactly |
| 13:50.47 | coppice | tzanger: your 4dB loss is really 0dB line loss, as they put 4dB of loss in there on purpose |
| 13:50.59 | tzanger | except in this case it's tzanTZANGER: SorrySORRY CAN YOU... |
| 13:51.06 | tzafrir | tzanger, instead of maintaining 3 or 4 different patches, I maintain http://git.tzafrir.org.il/?p=dahdi-extra.git;a=summary |
| 13:51.17 | tzafrir | Which automates the generation of a patch |
| 13:51.20 | tzanger | tzafrir: oh? |
| 13:51.43 | tzanger | coppice: yeah I had understood that you get about 3-5dB just for the copper losses in most installs |
| 13:51.51 | m0t3jl | [TK]D-Fender, er, BRI has 2 channels per line, how many channels does GSM have per SIM card ;) |
| 13:51.59 | tzanger | which is why you tune your rxgain such that you "see" about 4dB of loss |
| 13:52.36 | tzafrir | m0t3jl, a single "channel" per sim card. That is: it allows a single voice call at a time |
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| 13:59.11 | m0t3jl | tzafrir, thanks ;) |
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| 14:01.41 | tzanger | tzafrir: that's a nice git repo you got there :-) |
| 14:03.23 | tzanger | coppice: do echo cans actually work with that much path loss? Do they work after teh user bumps up the rx gain so they can actually hear? :-) |
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| 14:13.37 | henk | ok, now i see tcp stuff coming in to port 5060 on the * box when i call. everything coming from the sip server is SYN and answered by an RST, ACK. any idea what that could be about? |
| 14:14.46 | henk | no iptables-rules or anything similar on the * host... |
| 14:14.48 | [TK]D-Fender | henk: the only thing you showed was a single OPTIONS packet |
| 14:15.06 | [TK]D-Fender | henk: You don't have traffic to debug. |
| 14:15.36 | coppice | tzanger: they work OK. put the gain back after the canceller, and not before. The degrades the EC performance, but putting the gain before might lead to howling |
| 14:15.38 | henk | [TK]D-Fender: exactly, because the connection isn't working correctly yet. i'm trying to fix that... |
| 14:15.53 | henk | [TK]D-Fender: do you want to see the tcpdump? |
| 14:16.51 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 14:17.59 | [TK]D-Fender | henk: Is there something MORE in there? |
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| 14:18.43 | henk | [TK]D-Fender: that's my problem: just syns coming from the sip server and RST,ACKs coming from my * :( |
| 14:19.21 | henk | i guess i'll just allow _any_ communication between the sip server and * in the firewall... perhaps that helps. |
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| 14:27.00 | Secret_Hamster | henk waddya doing in here, get back in #networking |
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| 14:28.04 | henk | Secret_Hamster: awww, come on, this channel is big enough for both of use, isn't it? can i stay? please? pretty please? |
| 14:28.10 | henk | s/use/us/ |
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| 14:35.51 | henk | okayyyy... it seems i misunderstood something completely :-/ i _need_ tcpenable=yes, even though i am not offering any sip service via tcp on my *, just to make incoming calls via a tcpsip connection work. is that intended? |
| 14:36.22 | henk | and even though asterisk registers fine to the provider without it... that's confusing imho :-/ |
| 14:37.14 | russellb | I don't understand "even though I am not offering any sip service via tcp" .... "to make incoming calls via tcp sip connection work" |
| 14:37.22 | russellb | so you _are_ offering SIP TCP, then |
| 14:37.26 | russellb | if you want incoming calls to work ... |
| 14:37.38 | russellb | tcpenable tells chan_sip to listen for incoming TCP connections |
| 14:38.15 | russellb | disappears for a bit |
| 14:39.12 | henk | russellb: uhm... * is just the 'client' in that case. i thought it would not be needed then, just for clients using * as the server. but i'm probably just not getting some concept of sip or something... |
| 14:40.11 | henk | * already has a tcp sip connection to the provider, so why do i have to listen anyway? can't it just use the existing connection for everything? |
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| 14:45.07 | drmessano | henk... tcpenable=yes must be set |
| 14:45.58 | henk | drmessano: yeah, i know, you told me :-/ i just don't understand why registering with a peer works without but seemingly nothing else does... |
| 14:46.29 | drmessano | I didn't realize it would work at all without it.. I needed it when I used TCP |
| 14:46.39 | hrhrhr | all registering with a peer does is tell the peer your ip address |
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| 14:47.11 | hrhrhr | did you get anywhere with the udp side of things? i know you are focusing on tcp as a solution but in terms of debugging etc... |
| 14:47.26 | drmessano | henk: Perhaps the registration is on a specific server with your provider? |
| 14:47.43 | [TK]D-Fender | henk: Just because you registered doesn't mean that device is configured to SEND you any calls |
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| 14:48.15 | henk | drmessano: we are the provider. kind of... it's just one machine. |
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| 14:48.27 | drmessano | Ok, so the registration is with the 2801? |
| 14:49.13 | henk | hrhrhr: i didn't really try with udp... and i'm getting somewhere with tcp, which i'd prefer. i'll go back to udp when tcp fails completely, but i have hope :) |
| 14:50.37 | henk | [TK]D-Fender: yeah, but it still uses tcp and it works... so setting tcpenable to yes isn't the first thing that comes to mind... especially when it says 'incoming tcp connections' while i considered the connection to my provider an outgoing connection which is used in both ways once it's established... |
| 14:50.42 | drmessano | henk: tcp should not be much mroe work than UDP here.. I used TCP with MS Exchange.. and it was pretty much those two settings I gave you for general, and I think I did a proto=tcp for the peer |
| 14:50.44 | henk | drmessano: yes, it is. 2811 it is. |
| 14:51.18 | henk | afaict it works kind of fine now, the 2811 access-list just denies rtp traffic afaict... |
| 14:51.45 | [TK]D-Fender | henk: And how are you "able to tell"? |
| 14:52.09 | henk | [TK]D-Fender: about rtp traffic? |
| 14:53.20 | [TK]D-Fender | henk: Yes. Not that this should eevn be a point of concern yet |
| 14:53.37 | henk | [TK]D-Fender: i'm not really telling, but guessing a bit. i thought you had noticed ;) but the log on the cisco denies udp packets with ports between 10000 and 20000 which matches exactly the udp port range used for rtp in asterisk. also asterisk sees and answers the calls, i just don't hear anything... |
| 14:56.08 | [TK]D-Fender | henk: Who says that is the RTP range IT uses? |
| 14:56.22 | [TK]D-Fender | henk: they don't have to be even REMOTELY similar |
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| 14:57.10 | henk | [TK]D-Fender: i don't know, but i don't care. i'll just see what happens if i change the access list to allow udp from and to my * host. then i'll tell the person mainly managing that voice gateway for us what i did and how to do it properly... |
| 14:57.21 | Naikrovek | anyone here know of any software that electronically tracks things that are policy based? like say I want to request that the system admin give a particular user (Dave) access to a shared folder or something. Dave's supervisor would go to some site that let him request that, and then the security "team" would get notified and then if they approve, it gets sent electronically to the IT "team" for implementation |
| 14:57.23 | henk | i guess he'll know what ranges the device uses. |
| 14:57.24 | [TK]D-Fender | henk: RTP isn't the problem. |
| 14:57.32 | [TK]D-Fender | henk: you have no call to need RTP FOR |
| 14:57.42 | [TK]D-Fender | henk: Stop fixing problems you don't have |
| 14:58.02 | henk | [TK]D-Fender: i don't? but i see the calls in asterisk now... |
| 14:58.19 | hrhrhr | any sort of change request software i guess? place i used to work at used a HP based one. can't think what it was called now... |
| 14:58.36 | [TK]D-Fender | henk: I don't see you showing us. let alone telling us that |
| 14:58.45 | Naikrovek | hrhrhr: at state farm we had a brilliant piece of software but it is proprietary |
| 14:59.11 | Naikrovek | and i need it for more reasons than that. I need to enforce some policy 'round here |
| 14:59.17 | Naikrovek | suppose i could just write it |
| 14:59.34 | drmessano | Naikrovek: Wouldn't a simple ticket system work? |
| 14:59.49 | henk | [TK]D-Fender: i don't think i need to show you _everything_... when i manage to go from "no call arrives at asterisk at all because it RSTs every connection coming in" to "asterisk sees the calls and connects to the proper extension but i have no audio", i don't think i have to keep the world up to date about my progress... |
| 15:00.00 | Secret_Hamster | Naikrovek, then implement it with a wooden club with a nail through it |
| 15:00.07 | henk | now do i? |
| 15:00.08 | drmessano | Naikrovek: Notifications configured properly to match internal processes |
| 15:00.10 | Naikrovek | Secret_Hamster: heh |
| 15:00.15 | drmessano | We do that with Sharepoint, of all things |
| 15:00.34 | Naikrovek | i'll save the stick for later. right now i need to use the carrot |
| 15:00.41 | Secret_Hamster | you can have all the process in the world, but it's getting the pig headed idiots to comply to it |
| 15:00.57 | Naikrovek | well i have CEO buy-in |
| 15:00.57 | [TK]D-Fender | henk: If you have problems and you fail to show us debug and expect our help .... do you seriously WANT our help? |
| 15:01.02 | [TK]D-Fender | henk: then SHOW us something we can fix |
| 15:01.20 | Naikrovek | Secret_Hamster: and i can force people to stick to the procedure and the process with CEO standing behidn me |
| 15:01.22 | hrhrhr | Naikrovek: i've been looking for a good open source one for a while. holla if you find one |
| 15:01.35 | Secret_Hamster | hum the CEOs are generally the worse for procedure compliance. It generally doesn't apply to them |
| 15:01.49 | Naikrovek | he follows it all |
| 15:01.52 | henk | [TK]D-Fender: i think we have a little misunderstanding here. but i need to get home now. would you care to talk about it in a query in an hour or so? |
| 15:02.00 | Naikrovek | it's his minions that he can't wrangle |
| 15:02.03 | Secret_Hamster | arm you CEO with the stick |
| 15:02.16 | Naikrovek | so he asked me to find an electronic way to wrangle them into line |
| 15:02.24 | Naikrovek | well i've done this before |
| 15:02.40 | Naikrovek | I have to drag the organization, kicking and screaming, into some freaking order |
| 15:02.46 | Secret_Hamster | ahh the electronic way |
| 15:02.48 | [TK]D-Fender | henk: I may help out whenever you're in a position to do something about it. |
| 15:02.59 | Secret_Hamster | 240v sure does give a tingle |
| 15:03.02 | Naikrovek | lol |
| 15:03.07 | Naikrovek | it does indeed |
| 15:03.09 | henk | [TK]D-Fender: ok, talk to you later then... |
| 15:06.03 | *** part/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 15:06.11 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
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| 15:20.17 | lukhas | hello |
| 15:20.25 | JamesHarrison | Thought I'd ask in here in case anyone knew: I'm looking for an SIP phone or some other solution that would permit usage of an SIP phone to feed a radio mixing desk, ie XLR in and out for speaker and microphone. |
| 15:23.31 | Naikrovek | XLR? |
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| 15:24.07 | Naikrovek | you want a voip setup for a radio station |
| 15:24.12 | Naikrovek | as opposed to analog, traditional style |
| 15:24.24 | [TK]D-Fender | JamesHarrison: SIP phone > headset jack > splitter > adapter. |
| 15:24.45 | Naikrovek | that would be my suggestion as well |
| 15:24.55 | JamesHarrison | That's what I was thinking [TK]D-Fender, yeah |
| 15:25.05 | Naikrovek | if you want to integrate the call pickup and hold time monitoring then it's going to require a bit more integration than that |
| 15:25.22 | Naikrovek | (into the DJ console, I mean) |
| 15:25.33 | JamesHarrison | Yeah, that's the only snag; hangup and pickup would have to be controllable easily enough. |
| 15:26.00 | Naikrovek | i guarantee there's some way-overpriced software that does this |
| 15:26.20 | JamesHarrison | Yep, but we're a student radio station; no money at all, hence doing SIP instead of PSTN ;) |
| 15:26.35 | Naikrovek | if you have a programmer friend i'm sure something can be whipped up |
| 15:26.42 | Naikrovek | there are lots of SIP libraries out there |
| 15:26.45 | Naikrovek | some open source |
| 15:26.55 | JamesHarrison | My other plan was to use the existing PSTN hybrid we own, and getting a PSTN->SIP box of some description |
| 15:27.00 | Naikrovek | you could just incorporate it as a plugin into your existing system |
| 15:27.03 | JamesHarrison | which would also work, in theory |
| 15:27.05 | Naikrovek | those are called ATA adapters |
| 15:27.07 | Naikrovek | and they're cheap |
| 15:27.13 | Naikrovek | and they might also be perfect for you |
| 15:27.31 | p3nguin | ~ata |
| 15:27.32 | infobot | somebody said ata was Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 15:27.33 | Naikrovek | analog telephone adapter |
| 15:27.39 | Naikrovek | s/telephone/terminal/ |
| 15:27.50 | Naikrovek | those start at like $30 |
| 15:27.52 | JamesHarrison | The only thing we were thinking about was doing multiple lines, which afaik would require multiple ATAs? |
| 15:28.04 | p3nguin | Unless you have a multi-line ATA. |
| 15:28.07 | JamesHarrison | and doing conferencing between them would be impossible? |
| 15:28.20 | Naikrovek | not impossible |
| 15:28.21 | [TK]D-Fender | JamesHarrison: Between what? |
| 15:28.39 | JamesHarrison | Between multiple incoming callers |
| 15:28.50 | [TK]D-Fender | JamesHarrison: Calls go to *. * can conference |
| 15:29.26 | JamesHarrison | Okay. Well, plenty to think about and plenty of options, then; will go do some more research. Cheers all! |
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| 15:53.54 | hrhrhr | what's the general feeling of using friend= instead of the peer/user pair |
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| 15:55.20 | p3nguin | If you use type=peer, that should be good enough. |
| 15:55.57 | p3nguin | You don't need to add another entry just to make it a user if you already have one that's a peer. |
| 15:56.00 | Micc | orderlyq pricing is outragous, 150$ a month per seat. Do they really have people other than clearwire and digium that pay that much? It seems like something I could write in a weekend. |
| 15:56.10 | raden | hrhrhr, http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+type |
| 15:58.44 | p3nguin | That page isn't very clear and should probably be avoided. |
| 15:59.27 | puzzled | Micc: check out OpenACD. not sure to what extent they compare but it might be interesting and afaik is free |
| 15:59.55 | puzzled | Micc: http://github.com/Vagabond/OpenACD |
| 16:01.24 | [TK]D-Fender | Micc: Can you show us where you got that pricing info. That looks horribly wrong |
| 16:02.01 | Micc | From an email I got from them, 120 euros/month per seat. I'll copy and paste the exact text from them. |
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| 16:04.54 | raden | anyone use skype w/ asterisk ? |
| 16:05.07 | [TK]D-Fender | heads to lunch |
| 16:05.08 | puzzled | Micc: found this: http://www.orderlyq.com/orderlystatsse/pricing.html |
| 16:05.57 | p3nguin | I wonder if the high price of the skype channel driver is to deter those who want to use it for novelty purposes. |
| 16:06.07 | Micc | puzzled, thats orderlystats pricing |
| 16:06.10 | Micc | not the queue part |
| 16:06.11 | Micc | Our OrderlyQ product runs on Asterisk and is a managed service and costs 120 Euros per seat per month. |
| 16:06.29 | drmessano | p3nguin: No, I am sure there is some development cost recovery involved |
| 16:06.40 | puzzled | guesses that's too high for most small call centers |
| 16:08.22 | drmessano | Micc: the pricing on the website is less |
| 16:09.12 | drmessano | 80 per seat is the highest price on the site.. then you save with more seats |
| 16:09.55 | Micc | drmessano, did you find orderlyq pricing on the website or orderlystats? |
| 16:10.14 | p3nguin | "NEWSFLASH: ORDERLYQ is now available FREE OF CHARGE for all UK call centres!" |
| 16:10.27 | p3nguin | That's kinda neat. |
| 16:10.51 | drmessano | Oh NM |
| 16:10.52 | ChannelZ | bleh this website is a horror |
| 16:10.52 | Micc | mother f**kers |
| 16:11.23 | Micc | p3nguin, I saw that somewhere too. They give it away there and rape us, thats fucked up. |
| 16:11.29 | drmessano | My call centres are all in india.. |
| 16:11.52 | drmessano | I pay like 25 dollars a year for my agents.. |
| 16:12.15 | drmessano | They not only answer calls, but I can call them and make them do things.. like lick their own feet |
| 16:12.20 | drmessano | Not bad "owning" someone |
| 16:12.25 | Micc | OpenACD says its for freeswitch |
| 16:13.19 | Micc | drmessano, do you ever feel bad about taking advantage of someone like that? |
| 16:13.49 | drmessano | Micc: I do, but then they sing nursey rhymes to me, on demand, and I feel calm |
| 16:14.22 | Micc | is that an average wage for someone over there? 25$ a year? |
| 16:14.35 | Micc | or is that just for the software seat? |
| 16:15.05 | drmessano | Micc: I think you could pay them as much as $30 a year, but they don't sing well enough for that |
| 16:15.34 | Micc | I feel like your pulling my leg. |
| 16:15.39 | p3nguin | As a customer, can I get your agents to sing to me? |
| 16:16.17 | drmessano | p3nguin: If they sing to customers, they disappear |
| 16:16.28 | drmessano | p3nguin: It's a tough market |
| 16:17.06 | p3nguin | Damn, I was hoping I could get some tunes on your dime. |
| 16:17.13 | drmessano | p3nguin: I don't have time for my agents to play games.. except this tic-tac-toe I force them to play with me everyday at lunch |
| 16:17.56 | drmessano | p3nguin: It was "tic-tac-paycheck", but apparently withholding wages is illegal |
| 16:19.01 | p3nguin | You could probably let them play one-dimensional Tetris. |
| 16:19.06 | p3nguin | http://www.tetris1d.org/ |
| 16:19.17 | coppice | all joking aside, I see many factories in India where the sign at the entrance says unskilled labour Rs100 per day. Skilled labour Rs110 per day. That's around US$2. How can people live like that. the professional people in India are buying nice cars and homes, and live quite well, but factory workers are in a whole other category |
| 16:21.13 | drmessano | coppice: The same way we employ illegal aliens here.. give them $5 a day to do 10x that in work, and hopefully today wont be the day they get stiffed out of the $5 again |
| 16:21.22 | drmessano | ... and who are they gonna call... the cops? |
| 16:21.50 | hrhrhr | ghostbusters |
| 16:22.36 | drmessano | Of course, it's not legal here, but then again, we don't want them here, so it's not like anyone is gonna stand up for it, because they're the enemy stealing american jobs |
| 16:22.48 | drmessano | ...for $5 a day.. that they may not get paid for.. |
| 16:23.09 | p3nguin | They are willing to risk it, though. |
| 16:23.10 | Micc | It seems like it would be pretty simple to make the asterisk queue allow for saving a spot and calling back the caller when they are next in line. |
| 16:24.04 | drmessano | p3nguin: Yeah, my issue is the morality of how we condone it via our hatred of illegals.. as if anyone "deserves" it. |
| 16:24.46 | p3nguin | Some people deserve it, but not likely because they are here illegally. That's a completely separate matter. |
| 16:25.21 | p3nguin | I do believe that an illegal worker should receive a lower wage, though. |
| 16:25.23 | drmessano | p3nguin: You step on my lawn, and I tie you to a tree, throw lawn darts at you for days.. and I don't go to jail for torturing you because you were trespassing in the first place |
| 16:25.38 | drmessano | Love the way the legal system turns a blind eye |
| 16:25.53 | raden | why when i call a G729 phone from G729 phone on same network is asterisk using a license ? |
| 16:26.15 | Naikrovek | because asterisk is still in the call path somehow. are you recording the call? |
| 16:26.23 | raden | not at all :( |
| 16:27.01 | p3nguin | What I like is that if I break into your house and fall and break my leg while I'm robbing your safe, I can sue you for damages. And I'd probably win. |
| 16:27.11 | *** join/#asterisk defswork (~andy@mx2.3gcomms.co.uk) |
| 16:27.29 | drmessano | p3nguin: Not if I shoot you first |
| 16:27.32 | drmessano | :) |
| 16:27.57 | Micc | orderlyq can kiss my ass. http://www.voip-info.org/wiki/view/Asterisk+Queue+Callback |
| 16:28.03 | p3nguin | Assume you're having to deal with one of your Indian agents in person. |
| 16:28.04 | *** join/#asterisk Zarnick (~Zarnick@unaffiliated/zarnick) |
| 16:28.20 | drmessano | p3nguin: Moral of the story, always keep a handgun nearby.. Even if you're not in fear for your life, it's cheaper to shoot the burgular than to deal with him injuring himself on the job |
| 16:28.32 | p3nguin | yep |
| 16:28.35 | Zarnick | Hello, anyone has some experience with the Sipura SPA 3102 device? I'm having a hard time with routing calls trough the PSTN... |
| 16:29.04 | drmessano | Zarnick: http://www.2l2o.com/how-to/spa-3102 |
| 16:29.37 | [TK]D-Fender | Micc: that pricing page said the license was for 5 years, not per month |
| 16:29.44 | p3nguin | Even though I live in an apartment complex in a nice part of town and my chances of a break-in are virtually zero, I still have a 9mm next to my bed. |
| 16:29.56 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 16:30.46 | Zarnick | drmessano, what would be the DIDNUMBER? |
| 16:32.14 | *** join/#asterisk d00gster (~dt@77.31.90.141) |
| 16:32.17 | drmessano | Zarnick: See the next line |
| 16:32.34 | drmessano | Force the DID for CID or pass the CID from the PSTN.. your choice |
| 16:33.41 | Zarnick | Also, what I want to make is: if the user press 0 on the Line 1 box, the number would get routed to the PSTN line, so far I've reached this dial plan "(<0:>x.<:@gw0>|x.)" , but this always givex me a busy tone |
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| 16:34.15 | hrhrhr | requirecalltoken=no <--- for anyone scratching their head about iax from 1.4 to 1.6 | this needs to go under peer settings not general iax |
| 16:34.33 | drmessano | Zarnick: Use Asterisk to route the call |
| 16:34.48 | drmessano | Zarnick: FXS to FXO is wonky on the 3102, in my experience |
| 16:34.58 | Zarnick | the ideia is to bypass Asterisk on the regional offices |
| 16:35.05 | Zarnick | so they can make local calls |
| 16:35.35 | drmessano | Bypassing Asterisk and them making local calls are two different things |
| 16:35.48 | drmessano | Asterisk can still handle the call and send it back out the FXO |
| 16:35.58 | p3nguin | That's how I use my 3102. Local calls go out the FXO. |
| 16:36.41 | p3nguin | The @gw0 is what tells it to send the call right out the FXO. |
| 16:36.42 | drmessano | p3nguin: Do you have the dialplan in 3102 working where local calls on the FXS go direct to the FXO, or does Asterisk route the call? |
| 16:36.57 | p3nguin | direct |
| 16:37.03 | drmessano | Interesting |
| 16:37.22 | Zarnick | p3nguin....please....how can I make this??? (been strugling with this the whole morning...) |
| 16:37.22 | *** join/#asterisk garymc (~chatzilla@host81-148-79-26.in-addr.btopenworld.com) |
| 16:37.23 | p3nguin | Just specify the gw0 in the dialplan on the device. |
| 16:37.58 | Zarnick | the dialpan I've made on the Line 1 configuration, is it wrong? |
| 16:37.59 | p3nguin | The device will want to send all calls to Asterisk unless you write your dialplan to use gw0 on specific number/dialing patterns. |
| 16:38.11 | p3nguin | Looks wrong to me. |
| 16:38.27 | Zarnick | hum....how could I fix it? |
| 16:38.30 | p3nguin | Specify a better pattern for which numbers should go out gw0. |
| 16:39.10 | Naikrovek | portal 2 release date announced |
| 16:39.10 | Zarnick | let's say I want only 1234567 to go out via pstn |
| 16:39.20 | Naikrovek | sorry i know this isn't #games but i'm stoked |
| 16:39.28 | Zarnick | (1234567<:@gw0>|x.) |
| 16:39.50 | Zarnick | this would be correct? |
| 16:44.39 | *** join/#asterisk ayrjola (~ayrjola@a85-156-216-139.elisa-laajakaista.fi) |
| 16:44.52 | ChannelZ | Naikrovek: there better be cake this time. |
| 16:45.04 | MrUnagi | the cake is a lie |
| 16:46.17 | Zarnick | Sorry p3nguin, I just can't find the correct dialplan, could you provide me some example that should work? |
| 16:47.22 | Naikrovek | ChannelZ: mmm cake. pie is better tho. you gotta force cake on people, people come for pie |
| 16:49.37 | *** join/#asterisk ayrjola (~ayrjola@a85-156-216-139.elisa-laajakaista.fi) |
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| 16:50.56 | Zarnick | p3nguin, If I put (1234567@gw0) I can see it going trough Asterisk, which is exactly the oposite I want to make |
| 16:51.47 | p3nguin | zarnick: What you wrote above should work, I expect. I would probably try something like this to capture all valid 7-digit numbers for the local phone co. and send the rest to Asterisk: ([2-9]xxxxxx<:@gw0>|xx.) |
| 16:52.41 | paulc | Naikrovek: Out of interest, when's Portal 2 out? |
| 16:52.51 | Naikrovek | feb 9 2011 |
| 16:52.58 | paulc | Day before my birthday! nice :) |
| 16:53.18 | paulc | Still need to finish the last level or 2 on the original.. just a case of time.. |
| 16:53.35 | paulc | Back to the 3102 issue - what p3nguin wrote looks right to me |
| 16:53.46 | paulc | we've put some remote 3102s for testing DISA and 900 numbers in certain areas |
| 16:55.54 | Zarnick | arrr...it's not working...but if I disconnect the PSTN line and use the (1234567@gw0) it doesn't work, if I connect the PSTN line it works...can I assume it's going trough the correct port? |
| 16:56.37 | p3nguin | 1234567@gw0 is wrong. |
| 16:56.48 | p3nguin | 1234567<:@gw0> is right. |
| 16:57.06 | p3nguin | But you have to dial 1234567 for it to match. |
| 16:58.18 | p3nguin | [2-9]xxxxxx<:@gw0> will match any 7-digit number starting with 2, 3, 4, 5, 6, 7, 8, or 9. |
| 16:58.42 | p3nguin | If you start dialing with a 1, it won't match. |
| 16:59.12 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:00.10 | Zarnick | and now...it's not registering...lord |
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| 17:16.28 | Zarnick | p3nguin, can you send me the dialplan again? (everything fell apart here..hehe) |
| 17:16.55 | *** join/#asterisk mroe (~anon__@unaffiliated/roe) |
| 17:20.07 | mroe | I am working on getting the polycom pickup groups working with asterisk. *8 functions correctly, so I am reasonably certain that my callgroups and pickup groups are configured correctly in asterisk. Does anyone have experience setting this feature up with polycom phones? |
| 17:20.08 | *** join/#asterisk Lantizia (~Lantizia@erebus.seaquake.net) |
| 17:20.19 | Naikrovek | mroe: yes |
| 17:20.43 | Naikrovek | you can have up to 60 i believe, numbered 1-60 |
| 17:20.54 | mroe | right, the group I am using is 60 |
| 17:21.10 | mroe | and *8 works as expected with members of the group |
| 17:21.26 | Naikrovek | okay |
| 17:21.47 | Naikrovek | so what haven't you accomplished that you wish to accomplish |
| 17:22.00 | mroe | the soft button on the polycom phone that says 'pickup' |
| 17:22.22 | Naikrovek | hmm, that i've not done. |
| 17:23.09 | Naikrovek | man i'm getting spread thin |
| 17:23.18 | Naikrovek | "like butter over too much bread" |
| 17:23.32 | mroe | good movie |
| 17:23.47 | Naikrovek | i want to devote a lot of my time to learning asterisk and truly making this a system with everythign else we have here |
| 17:24.06 | Naikrovek | i want to get back to java programming on a daily basis |
| 17:24.37 | Naikrovek | i want to remove the kinks in our domain configuration |
| 17:24.41 | Naikrovek | i want to smooth out the networking |
| 17:24.43 | Zarnick | If I would like to make, for instance, 911 to go trough the PSTN: (911S0<:@gw0>) right? |
| 17:24.44 | Naikrovek | not enough time |
| 17:26.15 | p3nguin | [369]11<:@gw0> this is the match I have for that. |
| 17:26.37 | Naikrovek | i want to learn to play piano well |
| 17:26.45 | Naikrovek | plus i have a family |
| 17:26.46 | Zarnick | even like this...it doesn't work...still give me a busy tone |
| 17:26.53 | Naikrovek | jeepers i gotta prioritize better |
| 17:28.27 | *** join/#asterisk calmh (~jb@acro.nym.se) |
| 17:29.06 | Zarnick | Could the problem be somewere else? Some configuration on the PSTN Line tab maybe? |
| 17:29.42 | *** join/#asterisk _structz (~structz@gandalf.ai.unesp.br) |
| 17:30.08 | _structz | Hello all, little help, my asterisk frozed and got this message on FULL LOG - asterisk.c: Accept returned -1: Too many open files |
| 17:30.21 | p3nguin | PSTN Line is for communication between your FXO port and Asterisk... similar to how Line 1 is for communication between the FXS and Asterisk. |
| 17:30.23 | _structz | At ver = 1.6.2.9 - DEBIAN 5 x64 |
| 17:31.02 | Zarnick | p3nguin, then I have no ideia what could be wrong |
| 17:31.31 | Zarnick | I now that if I call from the Asterisk to the PSTN Line on that ATA, I can get the tone from the PSTN and call normally |
| 17:31.45 | Zarnick | but using the <:@gw0> to route some number there...no chance |
| 17:31.46 | p3nguin | Make sure you have VoIP-to-PSTN Gateway enabled on PSTN Line tab. |
| 17:32.01 | Zarnick | it is |
| 17:32.15 | Zarnick | and alsto PSTN-to-VoIP |
| 17:32.46 | p3nguin | and PSTN-To-VoIP Gateway too. |
| 17:32.51 | p3nguin | Oh. |
| 17:33.57 | p3nguin | You've enabled the line at the top of that page, as well? |
| 17:34.04 | p3nguin | Line Enable: yes|no |
| 17:34.40 | Zarnick | Yes |
| 17:35.22 | p3nguin | I'm not sure what else to do short of checking it out myself. |
| 17:35.44 | Zarnick | gah...getting desperate here |
| 17:36.09 | p3nguin | Mine works with the settings we've discussed. Something else must be different about your configuration. |
| 17:37.05 | p3nguin | There aren't a lot of settings that need changed to get it going. |
| 17:37.31 | Zarnick | That's what's odd about it... |
| 17:37.34 | p3nguin | from the factory, I mean. |
| 17:39.12 | p3nguin | I can only think of one problem I ever had with my device, and that was having the VoIP-to-PSTN Gateway disabled. I wasn't trying to connect my Asterisk outbound to the PSTN, so I felt like I didn't need that enabled. I was wrong. Without that enabled, the phone on Line 1 can't call out to the PSTN. |
| 17:39.22 | _structz | Hello all, little help, my asterisk frozed and got this message on FULL LOG - asterisk.c: Accept returned -1: Too many open files (Ast Version 1.6.2.6 - Debian 5 x64) |
| 17:39.37 | ChannelZ | Anyone know what a typical phone number in the Philipines looks like? |
| 17:40.03 | Zarnick | darn...think I'll just reset it and start over... |
| 17:40.08 | *** join/#asterisk talntid (~talntid@c-76-104-157-191.hsd1.wa.comcast.net) |
| 17:40.10 | *** join/#asterisk deonv (~adium@mail.flexpro-tech.com) |
| 17:40.12 | p3nguin | I was ->| |<- this those running a factory reset on the device because I thought I screwed up something somewhere and couldn't figure it out. I enabled that setting again and it was solved. |
| 17:40.30 | talntid | [Aug 18 10:39:00] WARNING[30808]: app_exec.c:222 execif_exec: Deprecated syntax found. Please upgrade to using ExecIf(<expr>?Congestion((null))) AND [Aug 18 10:39:07] WARNING[30831]: app_exec.c:222 execif_exec: Deprecated syntax found. Please upgrade to using ExecIf(<expr>?Busy((null))) |
| 17:40.39 | p3nguin | s/those/close/ |
| 17:40.57 | talntid | Are those because of.... exten => s,n,ExecIf($["${DIALSTATUS}" = "BUSY"],Busy) AND exten => s,n,ExecIf($["${DIALSTATUS}" = "NOANSWER"],Congestion) likely? |
| 17:41.04 | p3nguin | Fuck it, I'm not correcting it again. |
| 17:42.08 | Zarnick | hehehe... |
| 17:42.23 | Zarnick | yeah...but I'll start over anyway....let's see how it turns out... |
| 17:42.37 | Zarnick | actually...let me review just one more time... |
| 17:45.22 | Zarnick | darn...it should be all correct...the time for reseting it has come...hehe |
| 17:45.47 | p3nguin | I bet it doesn't fix it. |
| 17:46.14 | _structz | Hello all, little help, my asterisk frozed and got this message on FULL LOG - asterisk.c: Accept returned -1: Too many open files (Ast Version 1.6.2.6 - Debian 5 x64) |
| 17:46.17 | Zarnick | so do I |
| 17:46.26 | Zarnick | but I have nowhere else to look..hehe |
| 17:46.49 | p3nguin | Debian 5 x64 ... that would be Debian 320. |
| 17:46.54 | *** join/#asterisk bmint (~bmint@h174.92.190.173.static.ip.windstream.net) |
| 17:47.14 | _structz | Debian GNU/Linux 5.0 \n \l |
| 17:47.25 | _structz | <PROTECTED> |
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| 17:49.25 | tzafrir | _structz, how many active channels do you have? |
| 17:49.56 | tzafrir | something in the order of magnitude of 500? |
| 17:50.58 | _structz | tzafrir, no |
| 17:51.03 | _structz | tzafrir, might no get even to 30 |
| 17:51.09 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 17:51.19 | tzafrir | so it's some sort of leak. |
| 17:52.54 | *** join/#asterisk SiNGLer (~singler@78-60-54-125.static.zebra.lt) |
| 17:53.03 | _structz | that's my guess |
| 17:53.13 | _structz | if i update to 1.6.2.11 might resolve?: |
| 17:54.39 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 17:55.45 | p3nguin | I would always recommend updating to the most current release in your branch before asking for help fixing whatever is busted. If you ask Digium for support, that's what they would insist you do. |
| 17:57.23 | *** join/#asterisk hemantvoip (~hemant.vo@122.163.9.44) |
| 17:58.34 | russellb | it's also what we'll say if you report something to the issue tracker |
| 17:58.35 | p3nguin | http://news.softpedia.com/news/Critical-Vulnerability-Silently-Patched-in-Linux-Kernel-152678.shtml |
| 17:59.10 | WIMPy | And then you upgrade and get a lot more problems than you had before. |
| 17:59.16 | _structz | p3nguin, that's what i'll do. ;) just asked anyway |
| 17:59.39 | *** join/#asterisk hemantvoip (~hemant.vo@122.163.9.44) |
| 18:01.51 | p3nguin | Sometimes an upgrade does bring about other problems, but at least you have a better chance of solving the current ones as opposed to the old one that no one important cares about anymore. |
| 18:02.34 | WIMPy | Maybe, but I already broke LOTS of stuff by upgrading. |
| 18:04.21 | _structz | p3nguin, yeah.. that is something that worries me. |
| 18:05.07 | p3nguin | Back up your configs and modueles/libs. If the upgrade pukes, roll back to your current problems. |
| 18:07.03 | _structz | ok tkz ;) |
| 18:07.40 | Zarnick | IT WORKED!!! |
| 18:07.41 | Slugs_ | is there a wiki on allowing international dialing? im trying to allow calls to Korea |
| 18:08.41 | Zarnick | God damn it...it bloody worked!! |
| 18:08.56 | Qwell | zamba: sorry to hear that. |
| 18:09.00 | Qwell | Zarnick too |
| 18:09.20 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
| 18:09.41 | [TK]D-Fender | Slugs_: How are you NOT allowing calls to Korea? |
| 18:11.40 | Zarnick | now to find what the hell was wrong |
| 18:13.31 | *** join/#asterisk deonv (~adium@mail.flexpro-tech.com) |
| 18:18.18 | Zarnick | Well...couldn't find what was wrong...now that's weird |
| 18:18.27 | p3nguin | Accept success. |
| 18:19.11 | p3nguin | Study your settings. You'll want to remember how you made it work later when you start making changes and it breaks again. |
| 18:26.52 | Slugs_ | [TK]D-Fender: well one issue is that it's setup for 10 digit dialing |
| 18:26.58 | Slugs_ | [Aug 18 11:13:13] NOTICE[5557] chan_sip.c: Call from 'desk-4638' to extension '8210999604' rejected because extension not found. |
| 18:27.35 | Slugs_ | where can i read to fix this? |
| 18:28.23 | Slugs_ | 82 109 996 0454 <--- full number |
| 18:30.06 | p3nguin | extension not found <--- clue |
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| 18:33.08 | [TK]D-Fender | Slugs_: this is YOUR dialplan. Why are YOU limiting it to 10 digits? |
| 18:33.41 | [TK]D-Fender | Slugs_: "it's setup" <- YOU |
| 18:33.57 | [TK]D-Fender | Slugs_: Go make it NOT be limited to 10 digits. |
| 18:34.36 | p3nguin | Personally, I would want to have another extension specifically for matching international numbers, and they would be preceeded by 00 or 011. But that's just me. |
| 18:35.04 | p3nguin | My ITSP requires I send one of those on the front of the number, too. |
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| 18:37.04 | *** join/#asterisk Preytell (~jerry.win@65.114.21.3) |
| 18:38.26 | Preytell | I am working with routeflow for DID numbers and have a question, how do you stop random people from attempting to register extensions on your phone system from the internet but allow port 5060 for normal operations? |
| 18:39.00 | [TK]D-Fender | Preytell: google fail2ban |
| 18:39.32 | Preytell | thanks |
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| 18:54.53 | raden | when were talking codec bandwith G.711 87.2 Kbps would that be total for both streams or for each stream ? 87.2 UP & 87.2 DOWN ? or 87.2 TOTAL ? |
| 18:58.35 | timholum | I have Monitor(wav,filename,m) the output is the first half is incoming and the second half is outgoing, is there a way to make it so it is a murged file, so I can listen to them the way the conversation went? |
| 18:59.01 | p3nguin | You can use MixMonitor() instead of Monitor(). |
| 18:59.15 | [TK]D-Fender | raden: per direction |
| 18:59.27 | p3nguin | But Monitor() can combine the files after the recording has completed if you insist on using Monitor(). |
| 18:59.30 | timholum | p3nguin, Thanks Ill give that a try |
| 19:00.19 | raden | [TK]D-Fender, thanks |
| 19:00.59 | p3nguin | Note that the syntax of the application data differs between Monitor and MixMonitor. |
| 19:02.59 | timholum | I noticed that, how do I tell it the format? |
| 19:03.31 | p3nguin | "core show application MixMonitor" should help. |
| 19:07.10 | timholum | I think I found a different way to do it, I can set MONITOR_EXEC variable to soxmix that should work( hopfully :) ) |
| 19:07.33 | p3nguin | Or you could just yse MixMonitor and forget about trying to rig things. |
| 19:08.08 | p3nguin | But suit yourself. |
| 19:08.40 | timholum | I can not find any documentation on how to set the format of mixmonitor |
| 19:09.20 | p3nguin | (1403.30) <p3nguin> "core show application MixMonitor" should help. |
| 19:09.24 | p3nguin | clue ^^^ |
| 19:12.14 | ruben23 | hi guys i have asterisk used for voip trunk and even i had one caller i kee getting bad encho on my line when i call, any idea on how to solve any issue like this.. |
| 19:13.10 | Chainsaw | ruben23: You need echo cancellation then. |
| 19:13.44 | timholum | p3nguin: I am rilly not trying to be difficault, I did core show application mixmonitor, and read it. it shows the same thing as voip-info.org's mix monitor thing, there is no way to set the format. I am going to give it a shot though and hope it sets it to wave. |
| 19:14.05 | [TK]D-Fender | timholum: pastebin the instructions |
| 19:14.12 | p3nguin | [Description] MixMonitor(<file>.<ext>[|<options>[|<command>]]) |
| 19:14.32 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 19:14.48 | p3nguin | The format of the recorded file is determined by the file extension provided in the application data. |
| 19:15.08 | timholum | ahh :) ok :) |
| 19:15.34 | ruben23 | Chainsaw: how do i implement that, i heared there is no echo on voip trunk only for analog. |
| 19:15.48 | p3nguin | MixMonitor(${STRFTIME(${EPOCH},,%Y%m%d)}-${CALLERID(num)}.gsm) would create a file of gsm encoding with the name of 20100818-3149691077.gsm. |
| 19:16.59 | timholum | p3nguin: thanks, from all my other experiance with asterisk you have to specify the format so that is why I was confused |
| 19:17.20 | p3nguin | You're specifying the format, just not as an explicit option. |
| 19:19.08 | timholum | ya, I see that now |
| 19:19.14 | [TK]D-Fender | it is pretty explicit though. the name is the format jsut like every file that * reads |
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| 19:24.43 | ruben23 | :) |
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| 19:38.35 | ruben23 | Chainsaw:..? |
| 19:47.58 | Naikrovek | ruben23: people that call you hear echo? |
| 19:53.08 | [TK]D-Fender | ruben23: You can't Your provider is at fault if they can't fix this then replace them |
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| 19:57.06 | ruben23 | Naikrovek: when i call the client hear the echo on my end its clear..i have test 4 carriers but still having the same issue |
| 19:57.28 | [TK]D-Fender | ruben23: And what are YOU listening with? |
| 19:57.36 | Naikrovek | then the echo is the responsibility of the client or their provider |
| 19:58.21 | ruben23 | <PROTECTED> |
| 19:58.37 | Naikrovek | ruben23: you are hard to understand :) |
| 19:58.45 | [TK]D-Fender | ruben23: What DEVICE are you using during this call that you hear echo using? |
| 19:59.14 | Naikrovek | you are calling FROM different devices, all to the same number, or you are calling different numbers all from the same device? |
| 19:59.51 | ruben23 | <PROTECTED> |
| 20:00.14 | [TK]D-Fender | runAll experience this echo across multiple ITSPs? |
| 20:00.19 | ruben23 | Naikrovek: calling different numbers uisng my asterisk device.. |
| 20:00.21 | [TK]D-Fender | ruben23: All experience this echo across multiple ITSPs? |
| 20:01.09 | ruben23 | [TK]D-Fender: i ahve tried 4 voip carriers and still its present, some time it fine but most of the time, its present. |
| 20:01.24 | ruben23 | i mean the echo most of the time present. |
| 20:01.32 | [TK]D-Fender | ruben23: And all of your different phones get this problem as well? |
| 20:02.00 | ruben23 | [TK]D-Fender:yes and they get it randomly.. |
| 20:02.51 | [TK]D-Fender | ruben23: What ITSPs? |
| 20:03.03 | [TK]D-Fender | ruben23: And what phones? |
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| 20:04.35 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 20:04.41 | ruben23 | i got ITSPs,- vitality, rapidvox, and phones linksys SPA. |
| 20:04.54 | ruben23 | softphones- eyebeam |
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| 20:05.50 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
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| 20:18.03 | jly2680 | need help |
| 20:18.41 | russellb | don't we all |
| 20:19.11 | jly2680 | i need bootfile and app file for aastra 4422 |
| 20:19.27 | jly2680 | i cant boot the device |
| 20:19.53 | bmoraca_work | aastra.com might be a good place to go |
| 20:20.18 | jly2680 | check all their sites but nothing |
| 20:20.56 | jly2680 | does someone have aastra 4422 work with their asterisk box |
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| 20:22.58 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 20:23.01 | cusco | hi |
| 20:24.22 | *** join/#asterisk b14ck (~rdegges@24.24.128.47) |
| 20:26.14 | cusco | at the end of a outbound, I want to get the vars ${CDR(duration)} and ${CDR(billsec)} |
| 20:26.30 | cusco | but since the call hungup I cannot get them anymore? |
| 20:26.41 | cusco | what do you folks use to calculate these values? |
| 20:27.30 | bmoraca_work | are those variables not accessible in the h extension? |
| 20:27.42 | cusco | they return 0 |
| 20:27.44 | cusco | in h |
| 20:27.47 | bmoraca_work | if they're not, you might be able to get them via a query of your CDR source |
| 20:28.22 | cusco | in h I have: NoOp(Finished outbound. Total duration: ${CDR(duration)} billsec: ${CDR(billsec)}); |
| 20:28.36 | cusco | and log shows: [Aug 18 21:25:09] VERBOSE[2362] pbx.c: -- Executing [h@agents:1] NoOp("SIP/150-0000101b", "Finished outbound. Total duration: 0 billsec: 0") in new stack |
| 20:28.52 | bmoraca_work | you might have to query them from your CDR log |
| 20:29.16 | bmoraca_work | or, maybe they don't get written until after h ends |
| 20:29.17 | cusco | I don't wish to keep a cdr log... once I can get these values I will inster them in mysql database |
| 20:29.42 | cusco | bmoraca_work: you think? but if I do core show channel bla while it is going I see them set!! |
| 20:31.09 | cusco | returns the same on the end of h |
| 20:32.07 | bmoraca_work | why can't you just use a CDR log and query that? |
| 20:32.34 | cusco | my boss does not agree in maitaining a table with loads of useless info... |
| 20:32.53 | bmoraca_work | obviously, it's not useless info, because he needs it for some reason. |
| 20:33.16 | cusco | well these two variables aren't but there is LOADS of other stuff that is |
| 20:33.18 | cusco | I've seen it |
| 20:33.40 | bmoraca_work | not really. cdr logs are pretty useful for a lot of reasons. |
| 20:34.09 | cusco | we keep a log... with DIAL CONNECT COMPLETEAGENT etc |
| 20:34.43 | cusco | queue_log |
| 20:34.53 | cusco | and we insert a few more values into it |
| 20:35.08 | cusco | thus not needing cdr... |
| 20:35.10 | cusco | until now :( |
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| 20:36.47 | cusco | isn't there any other way of taking them values other than cdr? |
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| 20:38.26 | bmoraca_work | this is an aweful lot of hoops to jump through for no reason |
| 20:38.58 | cusco | hoops? |
| 20:38.58 | bmoraca_work | you could, i guess, potentially query the CDR database and then immediately delete the record... |
| 20:39.12 | bmoraca_work | hoops...as in, a lot of bullshit for no reason. |
| 20:39.18 | bmoraca_work | CDR logs are not that much |
| 20:39.18 | cusco | :( |
| 20:39.32 | bmoraca_work | i have a table with more than a quarter million CDR logs that's less than 50mb |
| 20:39.57 | cusco | well... cdr for 2 days 129.6 MiB |
| 20:40.14 | bmoraca_work | then you're not storing them properly |
| 20:40.20 | cusco | hmm |
| 20:40.50 | cusco | it logs everything, every dial.. whenever a call coems in queue it gets dialed maybe 10 times before answered, ringall strategy |
| 20:41.09 | bmoraca_work | you could set a variable to the epoch at the beginning of a call and then compare it to the epoch at the end of the call...that'd get you call duration, but not billsecs |
| 20:41.10 | cusco | I don't need to know who it dialed to, only who took the call |
| 20:41.25 | cusco | yes I do that already |
| 20:41.32 | cusco | I did that... |
| 20:41.38 | bmoraca_work | you're not talking about CDRs, then. you're looking at the asterisk log. i'm not talking about the asterisk log. |
| 20:41.43 | cusco | my boss is not satisfied, he wants to be able to tell the billsec |
| 20:41.54 | bmoraca_work | then tell him to look at the damn CDRs |
| 20:41.59 | cusco | well that was when I set up cdr_mysql.conf |
| 20:42.31 | cusco | also it shows every extension in dialplan.. like s,t etc |
| 20:42.40 | bmoraca_work | you can also use NoCDR and ForkCDR to more accurately record CDR entries |
| 20:42.54 | cusco | hmm.. |
| 20:43.04 | bmoraca_work | you're not looking at the right thing, then, or you have majorly screwed up your config |
| 20:43.44 | *** part/#asterisk thansen (~thansen@S0106001c1092cd20.cg.shawcable.net) |
| 20:44.18 | cusco | err... I can paste it |
| 20:45.15 | cusco | http://paste.debian.net/84208/ |
| 20:46.04 | bmoraca_work | cusco: a CDR record is one record per call. it's not irrelevant information, and it's the easiest way to get what you're looking for. |
| 20:46.21 | cusco | I understand... |
| 20:46.56 | cusco | still, can't I get those records any other way? |
| 20:47.09 | cusco | I mean values |
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| 21:39.41 | seanjohn | is there anything wrong with this call file? http://pastebin.com/dVJFSx4H |
| 21:40.03 | seanjohn | the permissions are correct and asterisk is refusing to note anything about it |
| 21:40.20 | seanjohn | auto loading for modules is on |
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| 23:30.50 | SirLouen | i have a problem with a tdm410p, i have connected a fxs module, and the molex conector to the card |
| 23:31.02 | SirLouen | also installed the dahdi linux complete package |
| 23:31.27 | SirLouen | despite of this i get this kernel error during the bootup -- DID YOU REMEMBER TO PLUG IN THE HD POWER CABLE TO THE TDM CARD?? |
| 23:32.02 | SirLouen | any ideas on what may be happening? |
| 23:32.35 | SirLouen | more info about the problem: wctdm24xxp 0000:05:08.0: ProSLIC on module 0 failed to powerup within 510 ms (0 mV only) |
| 23:32.36 | tzanger | SirLouen: you're sure that the hdd power cable is actually supplying power? |
| 23:33.07 | SirLouen | tzanger this molex was connected to a pci-x card before so i got the molex from that point |
| 23:33.27 | SirLouen | and switched the video card to another molex just to be care that the molex was ok and supplied power |
| 23:33.42 | tzanger | if you can, plug it into a hdd and make sure the hdd spins up |
| 23:34.01 | SirLouen | uhm... ide hdd difficult to be found |
| 23:34.03 | tzanger | I've seen those pins "come open" a little and they look like they're making a good connection but in reality it's not |
| 23:34.10 | tzanger | do you have a multimeter around? |
| 23:34.20 | SirLouen | polymeter maybe? |
| 23:34.38 | tzanger | I'm talking a regular old v-o-m meter the kind electornics hobbyiests have |
| 23:34.54 | SirLouen | i might have some in the garage |
| 23:35.13 | tzanger | you just want to make sure that you get 12V on the yellow-black leads |
| 23:35.17 | tzanger | the 5V isn't used on the card IIRC |
| 23:35.31 | SirLouen | 5V red? |
| 23:36.15 | SirLouen | well not sure i'm 100% sure that the molex works |
| 23:36.23 | SirLouen | since it works into the graphics card |
| 23:36.45 | SirLouen | i believe the videocard have a 12V connection |
| 23:37.01 | SirLouen | but maybe as you say is not properly fittin into the card |
| 23:37.06 | tzanger | um |
| 23:37.16 | tzanger | your video card won't SOURCE 12V, it will TAKE 12V |
| 23:37.23 | SirLouen | i've tried connection pullin into and out like 7 times |
| 23:38.06 | SirLouen | in 4 different molex |
| 23:38.23 | SirLouen | switching the system on and off all that 7 times |
| 23:38.52 | SirLouen | the powersource has like 10 molex you believe not all are ok for creating the connection? |
| 23:39.04 | SirLouen | this power source is like 4 years old |
| 23:39.35 | TJNII | Just check it with a voltmeter. |
| 23:39.48 | TJNII | That will settle the connector factor. |
| 23:40.04 | SirLouen | what shall i look for? |
| 23:40.08 | SirLouen | -12V? |
| 23:40.13 | TJNII | +12 and _5 |
| 23:40.16 | TJNII | *+5 |
| 23:40.24 | tzanger | - lead on black, + lead on yellow, you should see 12V |
| 23:40.28 | tzanger | if you reverse the leads, you'll see -12 |
| 23:40.29 | TJNII | Yellow is +12, red is +5, black is ground |
| 23:40.37 | SirLouen | ok |
| 23:40.41 | SirLouen | i must try this |
| 23:40.47 | SirLouen | if i see this |
| 23:40.57 | SirLouen | then what? |
| 23:41.28 | TJNII | I would also check black to chassis / a good ground. If you see voltage (more than a couple millivolts) you have an open ground. |
| 23:42.10 | SirLouen | i've i had an open ground |
| 23:42.41 | SirLouen | i would be constantly electrostatic shoking |
| 23:43.41 | SirLouen | one point is that the power source doesnt have enought W |
| 23:43.55 | SirLouen | only has 350W... but not so sure that the tdm needs much energy |
| 23:45.33 | SirLouen | anyway i'm going to check this 12V thing just to discard possibilities |
| 23:46.51 | TJNII | Remeber, if you disconnect the cable the test is invalid |
| 23:46.56 | TJNII | Check on the PCB. |
| 23:47.17 | tzanger | it only needs a few mA of 12v, nothing serious |
| 23:47.20 | tzanger | but it does need it |
| 23:47.41 | *** join/#asterisk russ (~russ@206.29.188.191) |
| 23:48.57 | TJNII | And 1000mA at 12V is only 12W. Unless yuor supply was already teetering on the edge of overload, this isn't going to push it over. |
| 23:49.15 | *** join/#asterisk lyetz (~lyetz@me.lyetz.me) |
| 23:54.09 | *** join/#asterisk proute (AnthonyCBR@s114-194.resnet.ucla.edu) |
| 23:54.56 | *** join/#asterisk b14ck (~rdegges@cpe-24-24-128-47.socal.res.rr.com) |
| 23:57.55 | proute | Hello |
| 23:58.58 | proute | I use asterisk 1.4.32 with B410P card. Is it possible to use dahdi module with euroisdn? On Internet somes forum give that it's possible and others give that not possible. |
| 23:59.19 | proute | So, Can I use B410P with dahdi and euroisdn with asterisk 1.4 ? |
| 23:59.20 | proute | thanks |
| 23:59.40 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |